Re: [asterisk-users] problem to socket programming in AGI
I don't get it, if it is a web service, why do you use sockets? Isn't it just a matter of calling the web service using curl,and then wait for the response? what am I missing?Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] problem to socket programming in AGI
From: Justin Killen jkil...@allamericanasphalt.com
Date: Mon, February 04, 2013 12:05 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
You are correct, this is not an asterisk question. What I would suggest would be to run your script outside of asterisk and debug the connection. Looking at the php doc page for fsockopen (http://php.net/manual/en/function.fsockopen.php), I see this example: ?php $fp=fsockopen("www.example.com",80,$errno,$errstr,30); if(!$fp){ echo"$errstr($errno)br/\n"; }else{ $out="GET/HTTP/1.1\r\n"; $out.="Host:www.example.com\r\n"; $out.="Connection:Close\r\n\r\n"; fwrite($fp,$out); while(!feof($fp)){ echofgets($fp,128); } fclose($fp); } ? I would first try running that (put in your host and port) and see what the error string coming back is.-Justin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad Sent: Monday, February 04, 2013 5:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] problem to socket programming in AGIHi, I know maybe this question is not related to asterisk, but I want to make XML RPC web service to other http server. I have elastix system. it is https and problem is source not destination server. In xml rpc we have fsockopen connection to connect destination server(xml rpc server). It return me connect error(0). what is the problem. is this related to elastix(asterisk) server? --
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Re: [asterisk-users] problem to socket programming in AGI
I would just type in the web service url manually in a browser, and if the browser displays the response, then there it is, the connection to the host server is open.Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] problem to socket programming in AGI
From: Justin Killen jkil...@allamericanasphalt.com
Date: Mon, February 04, 2013 12:25 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Yes, I think curl would probably be a better option than trying to use sockets directly, but if the socket won’t connect it doesn’t really matter what higher level method is used.-JustinFrom: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Monday, February 04, 2013 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problem to socket programming in AGI I don't get it, if it is a web service, why do you use sockets? Isn't it just a matter of calling the web service using curl,and then wait for the response? what am I missing? Christian Savinovich VoIP Telephony Consultant 646-982-3572 Original Message Subject: Re: [asterisk-users] problem to socket programming in AGI From: Justin Killen jkil...@allamericanasphalt.com Date: Mon, February 04, 2013 12:05 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com You are correct, this is not an asterisk question. What I would suggest would be to run your script outside of asterisk and debug the connection. Looking at the php doc page for fsockopen (http://php.net/manual/en/function.fsockopen.php), I see this example: ?php $fp=fsockopen("www.example.com",80,$errno,$errstr,30); if(!$fp){ echo"$errstr($errno)br/\n"; }else{ $out="GET/HTTP/1.1\r\n"; $out.="Host:www.example.com\r\n"; $out.="Connection:Close\r\n\r\n"; fwrite($fp,$out); while(!feof($fp)){ echofgets($fp,128); } fclose($fp); } ?I would first try running that (put in your host and port) and see what the error string coming back is.-Justin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad Sent: Monday, February 04, 2013 5:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] problem to socket programming in AGIHi, I know maybe this question is not related to asterisk, but I want to make XML RPC web service to other http server. I have elastix system. it is https and problem is source not destination server. In xml rpc we have fsockopen connection to connect destination server(xml rpc server). It return me connect error(0). what is the problem. is this related to elastix(asterisk) server? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --
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Re: [asterisk-users] DIDForSale spam
Although people complaining of spam may be valid, the one part of complaining about spam that bothers me, is that some people should look at themselves in the mirror and ask the question aloud "Why does it bothers me to see another Asterisk professional compete with me for jobs?", "Why do I get angry at seeing someone else's solicitation for services I too provide?"In reality, anyone who solicits customers in this list does a folly thing because this list has more asterisk consultants than customers. This list is the equivalent of a Home Depot parking lot full of construction workers looking for a job.Unfortunately, there is a fine line between being a forum where people can exchange ideas, and being a forum where people can find asterisk consultants, and both don't seem to co-exist well together.Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] DIDForSale spam From: Carlos Alvarez car...@televolve.com Date: Thu, January 10, 2013 3:44 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hopefully it's not, "What is the best DID provider for Asterisk..."On Thu, Jan 10, 2013 at 1:37 PM, Steve Totaro stot...@totarotechnologies.com wrote: So what asterisk issue do you have? Let's fix it. On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler rwhee...@artifact-software.com wrote: That does not solve any asterisk issue that I have. On 10/01/2013 1:32 PM, Carlos Alvarez wrote: On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote: It really didn't bother me as much as reading all the posts but that's just me...now back to Asterisk issues :)Sorry to add another, but for me, the main point is that this activity speaks to the character, ethics, and trustworthiness of the company doing it. We all have spam filters. I just also add the company to my do not buy/do not recommend list.-- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos AlvarezTelEvolve602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
Isn't this precisely the raison d'être for [asterisk-biz]? Oh my goodness!, the asteriz-biz? nooo, they will kill you if you try to post anything offering your services!... that list ceased to provide any value and died a long time ago precisely because its members ran each other away from it. A while back, I wrote a nice click-to-call service and I dared put a post indicating that I was offering it for a fee, and in no time they called me "spammer". There is really no incentive to reward someone else's achievements, unless you tell them that you are given them your code for free, then they want it (totally contradicting the meaning of the word "business").Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] DIDForSale spam From: Chris Bagnall aster...@lists.minotaur.cc Date: Thu, January 10, 2013 5:17 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote: Unfortunately, there is a fine line between being a forum where people can exchange ideas, and being a forum where people can find asterisk consultants, and both don't seem to co-exist well together. Isn't this precisely the raison d'être for [asterisk-biz]? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR platform for a mobile operator
What in the world "Asterisk to a mobile operator" means? you mean you are are using a gsm gateway? what interface are you using?... not that I intent to answer your question, but you should be clear and specific if you expect someone to give you a pointer.Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] IVR platform for a mobile operator From: "Danny Nicholas" da...@debsinc.com Date: Wed, January 09, 2013 10:07 am To: "'luke devon'" luke_de...@yahoo.com, "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devonSent: Wednesday, January 09, 2013 9:06 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] IVR platform for a mobile operatorHi Friends ,I want to setup a IVR platform using asterisk to a mobile operator.Can somebody give me some guides withrecommendedhardwaretypes ?Thank youLuke.IMO you will be happiest with a SIP trunk handling this as there can be horrible latency in DAHDI/Mobile connections.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR platform for a mobile operator
I think that the mobile operator as any other company receives calls by pots lines as T1 E1... ina ny way if he will receive calls through a gsm gateway the gateway itself must connect to pbx in a standard way probabilly voip the hardware will be a server an the interface .. Thanks Adriano, but if that is the case, then it is just an IVR like any other IVR. It doesn't make any difference for us to know the business goal. I was just trying to figure out all ulterior motives the poster might have to use the term "mobile operator".Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] IVR platform for a mobile operator From: adriano adriano.ghe...@gmail.com Date: Wed, January 09, 2013 3:52 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I think that the mobile operator as any other company receives calls by pots lines as T1 E1... ina ny way if he will receive calls through a gsm gateway the gateway itself must connect to pbx in a standard way probabilly voip the hardware will be a server an the interface .. hth Adriano Il 09/01/2013 18:38, C. Savinovich ha scritto: What in the world "Asterisk to a mobile operator" means? you mean you are are using a gsm gateway? what interface are you using?... not that I intent to answer your question, but you should be clear and specific if you expect someone to give you a pointer.Christian Savinovich VoIP Telephony Consultant 646-982-3572 Original Message Subject: Re: [asterisk-users] IVR platform for a mobile operator From: "Danny Nicholas" da...@debsinc.com Date: Wed, January 09, 2013 10:07 am To: "'luke devon'" luke_de...@yahoo.com, "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comFrom: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon Sent: Wednesday, January 09, 2013 9:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IVR platform for a mobile operatorHi Friends , I want to setup a IVR platform using asterisk to a mobile operator. Can somebody give me some guides withrecommendedhardwaretypes ? Thank you Luke. IMO you will be happiest with a SIP trunk handling this as there can be horrible latency in DAHDI/Mobile connections.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Impromptu conferencing
I use the ChannelRedirect function to redirect the desired channel to the meetme roomChristian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Impromptu conferencing From: James Sharp ja...@fivecats.org Date: Wed, November 07, 2012 2:55 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On 11/7/2012 2:01 PM, martin f krafft wrote: Dear list, we would really like to be able to "invite a third and fourth party" to our current one-on-one call. At the moment, we have to agree to dial into MeetMe 10 minutes later, then make calls to the third parties, and hope it all works out. I have found a couple of examples on the Internet for converting channels into conferences, but I could not get any of them working. Does anyone have a working example they would be willing to share? Why not blind transfer people to a conference extension and then dial into it yourself? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tips for installing and configuring Digum cards
Usually, channels 1-15 and 17-31 are B-channels and 16 is the D-channel;We don't use E1s here in the USA.I just finished installing a PRI line, and being a complete novice at it myself, this is what I wish someone had told me:- the dahdi program dahdi_genconf creates 2 files 1) /etc/dahdi/system.conf and 2) /etc/asterisk/dahdi-channels.conf. The latter is the one that maintains the configuration for asterisk. File chan_dahdi.conf is where the configuration is maintain, you will need to have an#insert dahdi-channels.conf at the start of chan_dahdi.conf, otherwise it will never be read. Else you can type the configuration yourselft in chan_dahdi.conf- Not written anywhere, but the way chan_dahdi.conf is read, there are no [] separators, so any parameters in the file apply to the first "channel" statement it finds.Hope it helps-Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Tips for installing and configuring Digum
cards
From: A J Stiles asterisk_l...@earthshod.co.uk
Date: Wed, October 10, 2012 1:17 pm
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
On Wednesday 10 October 2012, Mitch Claborn wrote:
I am a complete novice at T1's, etc. What else besides framing and
coding do I need to ask about?
Ask how numbers come through on incoming calls: in "international" format
(with IDD and STD codes, but without the leading double-zero and zero
respectively); in "full national" format (with STD code including the
leading zero); in "national" format with the STD code *but* with the leading
0 stripped; or just the local number.
Also ask how they want you to send numbers on outgoing calls (in particular,
do you need to insert the STD code even for a local call?)
Then, don't believe a word they say; and use something like
exten = s,1,NoOp(Call to ${EXTEN})
to see how they are _actually_ sending numbers to you. Start out by sending
your numbers in the same format, but don't expect consistency.
Usually, channels 1-15 and 17-31 are B-channels and 16 is the D-channel; but
again, check this with the telco. If your box can't find a D-channel, it won't
work at all.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Tips for installing and configuring Digum cards
(I'm sure its written somewhere, I just can't be bothered to look right now.) If you come across it, please let me know because other than getting a hint somewhere after 2 hours of googling, I would not have known.And aren't the characteristics 'cumulative' ? Can't tell you. You seem to hint that they are cumulative, although as far I tested, they weren't. characteristic-a = foo characteristic-b = bar channel = 1 characteristic-b = bar channel = 2channel 1 has foo and bar. Channel 2 has bar only. ...(?)In any case, my approach was not to take any chances and make sure all characteristics I use are cleared of any previous values in each channel declarationChristian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Tips for installing and configuring Digum cards From: Steve Edwards asterisk@sedwards.com Date: Wed, October 10, 2012 2:27 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com On Wed, 10 Oct 2012, C. Savinovich wrote: - Not written anywhere, but the way chan_dahdi.conf is read, there are no [] separators, so any parameters in the file apply to the first "channel" statement it finds. (I'm sure its written somewhere, I just can't be bothered to look right now.) Wouldn't it be more accurate to say you specify the characteristics and then specify the channels the characteristics apply to? And aren't the characteristics 'cumulative' such that characteristic-a = foo characteristic-b = bar channel = 1 characteristic-b = baz channel = 2 results in channel 1 having foo and bar while channel 2 has foo and baz? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trigger Asterisk after data inserted in mysql
Hello David,If you're going to top-post (which is against the rules on this list),I apologize if mistaken, but out of curiosity, can you please refer me to where in the rules it says that we can not top-post in this list?ThanksChristian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Trigger Asterisk after data inserted in mysql From: Chad Wallace cwall...@lodgingcompany.com Date: Wed, September 19, 2012 4:30 pm To: asterisk-users@lists.digium.com On Wed, 19 Sep 2012 14:04:33 -0400 David Cook dbc_aster...@advan.ca wrote: It looks like the answer is yes. I thought the answer was 42. If you're going to top-post (which is against the rules on this list), the least you can do is phrase your answers in a way that illustrates the question. Thanks! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trigger Asterisk after data inserted in mysql
If you're going to top-post (which is against the rules on this list), I agree with you. Just really wanted to locate where it was defined.Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Trigger Asterisk after data inserted in mysql From: Chad Wallace cwall...@lodgingcompany.com Date: Wed, September 19, 2012 5:14 pm To: asterisk-users@lists.digium.com On Wed, 19 Sep 2012 13:44:47 -0700 "C. Savinovich" c.savinov...@itntelecom.com wrote: Hello David, If you're going to top-post (which is against the rules on this list), I apologize if mistaken, but out of curiosity, can you please refer me to where in the rules it says that we can not top-post in this list? It's understandable that many are unaware of the rules, since it took some hunting for me to find them again. They aren't mentioned at all on any of the mailman pages at lists.digium.com. The rules can be found here: http://www.asterisk.org/community/rules Anyway, there are many who like to top-post, and I'm not interested in a holy war. I just wanted to point out that if you want to top-post, it may be worthwhile to consider ways to minimize the impact--like phrasing the answer so that the question is immediately apparent, and you don't force your readers to scroll down, possibly dodging signatures, disclaimers, pasted logs, and dragons, just to get to the question being answered. Original Message Subject: Re: [asterisk-users] Trigger Asterisk after data inserted in mysql From: Chad Wallace cwall...@lodgingcompany.com Date: Wed, September 19, 2012 4:30 pm To: asterisk-users@lists.digium.com On Wed, 19 Sep 2012 14:04:33 -0400 David Cook dbc_aster...@advan.ca wrote: It looks like the answer is yes. I thought the answer was 42. If you're going to top-post (which is against the rules on this list), the least you can do is phrase your answers in a way that illustrates the question. Thanks! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Not to bash on the developer who did this I get that we don't always think out side the box all the timeYou can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?. All you have to do, (that's what I did, and took me like 4 hours) is write a program that parses through your dialplan code and translates the n's into actual numbers, including the translation of the gotos into line numbers. Sucks? yes. Is the realtime limitation going to stop me from doing what I want? no way.Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority From: Leandro Dardini ldard...@gmail.com Date: Fri, August 03, 2012 10:11 am To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I am kissing every inch of land where each one of the asterisk's developer is putting his feet. In the last 10 years I have worked thanks to the availability of the asterisk code. Most of my income was possible just thanks to asterisk, so I am pretty biased when trying to evaluate if the asterisk code is good or not. You can understand if I "love" the way asterisk has been coded. Nevertheless things can be better and they can be better thanks to you. Asterisk is open source and Mark is a very kind person when you submit patches, so put your ideas in new code and send to him. If you don't know how to code, hire some developer and have him to code your view of a better RT code. If it will be accepted by the core developer, all us will be happy. if it will not accepted, you'll be happy with you own personal branch. I run for a small period of time my personal asterisk tree because the italian telephony system is flawed and clients want services not suitable for the general asterisk audience, so there is nothing to worry to have your personal asterisk code. LeandroPSI think your idea of extension RT can be accomplished with some triggers and replacing the extension table with a view on your own n-enabled extension table 2012/8/3 Bryant Zimmerman brya...@zktech.com Leandro I have to disagree reasonable designers would have done a better job with this one. But we developers are not always so reasonable. The issue is manydevelopers when pushingto put featuresin they don't put on their designers hat and think out side the box first.Heavenknows I have been guilty of this oneover the years and had togo back and refactor. It is not so reasonable to think that this limitation has to existdevelopers have been putting order by fields in dbdriven systemsfor years. What of the guy who want's to use n(target) or 4(target) (I know this may have not been an option when RT was first done now it is) sothey can addspecialized jumping code. If I had been designing the Realtime (today) I would have added a field for the priority and made it a full alpha / numeric and added an order by field. As it sits now how do you do n, i, h or tags ect It kinda sucks and limits the Realtime. Not to bash on the developer who did this I get that we don't always think out side the box all the time nor was some of this abilityavailable when the RT was written. but know it does so what do we do. Unfortunately I am not a ansi C guy or I could probably fix it . Thanks BryantZimmerman (ZK Tech Inc.) From: "Leandro Dardini" ldard...@gmail.com Sent: Friday, August 03, 2012 2:18 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, "virendra bhati" virbh...@gmail.com ha scritto: Hi Team, I want to used 'n' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:http://www.asterisk.org/hello asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
AJ, You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle?Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority From: A J Stiles asterisk_l...@earthshod.co.uk Date: Fri, August 03, 2012 11:45 am To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com On Friday 03 August 2012, C. Savinovich wrote: Not to bash on the developer who did this I get that we don't always think out side the box all the time You can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?. All you have to do, (that's what I did, and took me like 4 hours) is write a program that parses through your dialplan code and translates the n's into actual numbers, including the translation of the gotos into line numbers. Sucks? yes. Is the realtime limitation going to stop me from doing what I want? no way. That is the sort of thing that might actually be worth submitting upstream. There must be loads of dialplans out there that use "same", "n" and labels all over the place. The only reason mine don't, is because I've been using Asterisk since before these features were introduced and I got used to the old ways. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Basic?... no man, I am kid!Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority From: "Raj Mathur (राज माथुर)" r...@linux-delhi.org Date: Fri, August 03, 2012 2:21 pm To: asterisk-users@lists.digium.com On Friday 03 Aug 2012, C. Savinovich wrote: You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? If you ever used BASIC you'd remember the trick is to increment line numbers (priorities) by 10. I presume a dialplan would work fine even if the priorities aren't sequential, as long as they're increasing monotonically. Could someone confirm? Having said that, I use n with abandon. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question for the group
Thanks for this - but I am looking really for a software type solution.I would venture say that he means he wants it for free.Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Question for the group From: James Wystead szilvertho...@gmail.com Date: Fri, February 10, 2012 11:57 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Yes, I like the look of that.Researching it too - the commercial one looks nice too, but I don't know if there is a budget.GOn Fri, Feb 10, 2012 at 11:52, Terry Brummell te...@brummell.net wrote: I assume that solution was A2Billing? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group - Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestionsWe are going to be setting up a prepaid PBX service with the following features:• Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticatedI see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: "Thanks for this - but I am looking really for a software type solution". The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version that support Database Configuration
If you go to a web site called google.com, and enter "asterisk version that support the ability to have the configuration in the database", and read the first search result, you will get your answer.Christian Savinovich Original Message Subject: [asterisk-users] Asterisk version that support Database Configuration From: bilal ghayyad bilmar...@yahoo.com Date: Thu, February 02, 2012 3:51 pm To: asterisk-users@lists.digium.com Hi All; Which asterisk version that support the ability to have the configuration in the database? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Bull Service Providers
In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersChristian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF games with Asterisk
You have plenty of ways to do this. You can use the room number + user number to get the conference number. You can use the channel ids to keep a table of conference members and their statuses. C. Savinovich On September 7, 2011 at 9:15 AM Danny Nicholas da...@debsinc.com wrote: It seems to me that you are overworking AMI to do what could be done with AGI. You could use an AGI to poll Konference and return a dialplan variable with the file to use in Playback/Background or even MOH. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, September 07, 2011 6:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; onewaytoconnect Subject: Re: [asterisk-users] DTMF games with Asterisk Hi Amit, My scenario is that, If 3 conference is running in Asterisk then I will play a sound file with the help of Asterisk AMI then I will get DTMF from all the users. the same things will be done any all the Konference and all conference will be play different files. If you have any alternate suggestion the please help me On Wed, Sep 7, 2011 at 5:00 PM, amit anand onewaytoconn...@gmail.com [mailto:onewaytoconn...@gmail.com] wrote: Hi This can happen you can create more than 1 AMI connection. if you need better on access control then you can create new user in manager.conf with set of privileges that you can offer to each of them On Wed, Sep 7, 2011 at 15:59, virendra bhati virbh...@gmail.com [mailto:virbh...@gmail.com] wrote: Hi list, I want to know that will it be possible that more then 1 AMI is connected from single Linux machine with different name ? As we know that default 1st AMI connection will come with 127.0.0.1 and root information. My requirement is that I want to handling events for more then one Konference. So I required more then 1 AMI connection might be 1 connection for 1 konference. Because I will play some IVR files to get DTMF and on this DTMF i will check the correct DTMF. So that I will get the right user with correct input. So please guide me. -- - Thanks and regards Virendra Bhati +91-9172341457 [tel:%2B91-9172341457] Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Amit Anand +91 9818559898 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
Does this ConfBridge requires a hardware timing source? Will I be able to use this on any virtual server without having the need special changes to the VM setup? Thanks C. Savinovich On April 25, 2011 at 10:27 AM David Backeberg dbackeb...@gmail.com wrote: On Mon, Apr 25, 2011 at 9:38 AM, David Vossel dvos...@digium.com wrote: I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revisionrevision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely revamped, highly optimized, and feature rich conferencing application capable of mixing sample rates from 8khz all the way up to 192khz! Exciting right?! So way back when the 'old' ConfBridge was announced, my understanding was it was originally an internal Digium tool for exercising the Bridge() code and it was decided to release it to the public in the event the code might be useful to others. The old ConfBridge was missing stuff that was in MeetMe(), and wasn't that compelling for my particular usage. This 'new' ConfBridge looks to be much more full-featured. So can anybody explain the motivation for this? Is this a replacement for MeetMe() where at a certain point we envision dropping MeetMe() from the codebase? Does ConfBridge() scale to many users as nicely as MeetMe? I'm assuming the MeetMe ability to use a hardware source for timing will still be superior with large user counts in rooms? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unresponsive
Quick question out of curiosity: Did you googled your problem, and read through all the results, and made an exhaustive research on line of the error message before you opted to post your question here? CS On April 18, 2011 at 10:16 AM Jonas Kellens jonas.kell...@telenet.be wrote: On 04/18/2011 03:58 PM, Terry Brummell wrote: http://lmgtfy.com/?q=audiohook.c%3A+Failed+to+get+160+samples+from+read+factory This should tell me that there are others who experience this same problem in some kind of form and that there is no real answer to it ? Hence why I seek for an answer here. Kind regards, Jonas. From: Sent:Mon 4/18/2011 9:46 AM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject:[asterisk-users] Asterisk unresponsive Hello list, I've got a whole lot of these in my debug log : [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples Asterisk freezed and only a reboot of the whole server fixed this. Any command on the Asterisk CLI was not executed because Asterisk was too busy processing all of these messages that you see in the debug log. What is the origin of these messages ? Kind regards, Jonas. Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing
Re: [asterisk-users] Notify me when the call is answered
You want both phones to ring? then why don't you just create a group so your
mobile also rings at the same time as the other extensions just don't answer
your mobile.
CS
On March 17, 2011 at 8:52 AM Eric Smith e...@fruitcom.com wrote:
Hi
I want to have some signal when a call is answered.
I can watch the asterisk debug or logs and see when a call is answered
of course but I want a sound notification.
I tried this:
[macro-notifymobile]
exten = s,1,Dial(SIP/foobar,10)
exten = _0031.,n,Dial(SIP/foobar2/${EXTEN},60,wM(notifymobile))
But this results in:
[Mar 17 13:41:46] -- IAX2/4506-102 answered IAX2/4506-35
[Mar 17 13:41:46] -- Executing [s@macro-notifymobile:1]
Dial(IAX2/4506-102, SIP/foobar|10) in new stack
[Mar 17 13:41:46] -- Called foobar
[Mar 17 13:41:46] -- IAX2/4506-102 requested special control 20, passing
it to SIP/foobar-b760dd78
[Mar 17 13:41:46] -- SIP/foobar-b760dd78 is ringing
[Mar 17 13:41:46] -- IAX2/4506-102 requested special control 20, passing
it to SIP/foobar-b760dd78
I think it is passing the call to the extension SIP/foobar (my wifi mobile
device) which rings.
I want the call to stay connected to the original extension.
How would I achieve a notification this way or another way?
And is it possible to Dial() and only connect to your extension when someone
answers the call?
--
- Eric Smith
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Re: [asterisk-users] Free calls to the US provider recommendation
You should try paying for the call, and then you will be able to get good service CS On February 21, 2011 at 2:07 PM Christian christia...@runbox.com wrote: Hi all, Sorry for being a little off topic, but I just need some tips on some good provider that offers free calls to the US. I have tried out one called Whistlephone, but I am not able to receive calls with it and when I use the follow me feature it still rings here. So any other I should try? Many thanks for your help! Christian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London
45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: 1.Candidate must have good knowledge of setting up SIP and IAX Trunks. 2.Must have experience in installing and configuring SIP Express Router or OPEN SER. 3.Installation and trouble shooting of Asterisk Servers using Centos. 4.Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. 5.Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. 6.Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. 7.Scripting /Bash scripting would be useful. 8.Expert knowledge in Configuring, Maintaining andqueryingMySQL. 9.Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net [mailto:ja...@langleyjames.co.uk] Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London
Can you point out to me the places in London that sell food at American prices? Perhaps I get SeamlessWeb to deliver every morning from Brooklyn to London. On December 22, 2010 at 1:24 PM Don Kelly d...@donkelly.biz wrote: 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfC. Savinovich Sent:Wednesday, December 22, 2010 10:23 AM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net [mailto:ja...@langleyjames.co.uk] Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London
Wait, is 70k US for an experienced engineer supposed to be adequate? Thank you, not only that , but also note that it would be 70K at the US dollar exchange rate. However, because it is 45K Euros/Pounds earned and spent in UK, for all practical purposes it is just the same as if it was 45K US Dollars earned in the USA. On December 22, 2010 at 3:49 PM Watkins, Bradley bradley.watk...@compuware.com wrote: Wait, is 70k US for an experienced engineer supposed to be adequate? From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfDanny Nicholas Sent:Wednesday, December 22, 2010 2:27 PM To:'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer45KSouth London Wouldn't that be 70K USD? Or should we REALLY be worried about the British economy? From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfDon Kelly Sent:Wednesday, December 22, 2010 12:24 PM To:'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It's about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf OfC. Savinovich Sent:Wednesday, December 22, 2010 10:23 AM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. Required Skills Qualifications: Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. Kind Regards Jess 08451249555 Jess Hart __ Langley James IT Recruitment 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net [mailto:ja...@langleyjames.co.uk] Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com--
Re: [asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.
Without reading too much into your description, I can tell you that being an inband sound, and as long as the dtmf tone is heard by everybody during the conference, and being the ivr gateway one of the parties of the conference, I don't see a reason why the ivr gateway wouldn't act upon hearing the dtmf tone. It wouldn't know who pressed it, although if that matters, can be arranged by writing a patch to the meetme application where you can identify the channel that pressed the dtmf tone. Best Chris Savinovich On December 20, 2010 at 6:56 AM Asterisk Man theasterisk...@gmail.com wrote: Will someone help/direct me find a way to implement this? Or you can suggest some other method. On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man theasterisk...@gmail.com [mailto:theasterisk...@gmail.com] wrote: Hi friends, I want to implement following scenario using Asterisk. Please suggest me whether it is possible or not. This is bit off Asterisk and more on SIP side. An Asterisk box with one Station(SIP channel) and PRI. Agent dials a PSTN number of customer from station through Asterisk PRI. Agent gets connected with customer. Agent puts customer on hold. Agent dials another PSTN number which is of IVR gateway. Agent now makes conference(Station facility) with customer and IVR gateway. Gateway plays an IVR asking customer to enter his customer id number. My question is, will DTMF get forwarded to IVR gateway? I am asked to implement this and not having PRI for the moment in my Asterisk box. Thanking you in advance. -AsteriskMan Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
How do you know a good soccer player when you see one? If you are a good scout, just by his body language. Just by seeing him how he walks and positions himself on a field. By the time he touches the ball, he is either eliminated from my list of prospects or he is marked as good to be considered. How do you know a good technical person when you see one? Because irrespective of wherever he/she is from, regardless of his language, social status, and even upbringing, he KNOWS WHAT GOOGLE IS. The desire to investigate, research, and READ must be born with him the same day he is born. There is a saying that goes you can't tech a man anything, you can only help him find himself. In this case, this man has been attentively helped, and his questions have been duly answered more than once in this forum. He has been told in no uncertain terms to SEARCH IN GOOGLE... That is the answer, based in the fact that he will find plenty of solutions to HIS particular question within 1 minute. He doesn't listen. And listening is not a cultural constrain (perhaps if we call him little grasshopper will he be more attentive?). To me, he doesn't have it. I have friends who ask me all the time to teach them programming and I ignore them. Inside I know that they don't have it because a real programmer never asks another to teach him to program, therefore I would save myself the time. But maybe I am wrong, maybe this guy will learn from what we are trying to tell him... I hope. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, November 13, 2009 8:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: hi Dan On 13 Nov 2009, at 16:02, Cary Fitch wrote: Sorry, I can't resist. Evidently How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. You just joined. How about leaving member chastisement to the sponsor of the list? I'd happily do so if it happened. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Well, most of the people on the list managed to do this without harassing people.. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Yes, and they shouldn't. Its called etiquette. Be charitable and kind. Have a nice day. I always am. Here, have a free hug, *hug* I find that helps when I'm feeling grumpy. S ___ -- 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 ___ -- 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
Re: [asterisk-users] how to configure softphones in asterisk server
He wrote me too. I would have helped him, but the name on the email address threw me off. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 09, 2009 9:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] how to configure softphones in asterisk server That's what yahoo.answers.com is for! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, November 09, 2009 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to configure softphones in asterisk server You just don't get it, do you? Your indolent methods of getting what you want are not at your disposal here. This is not a homework help forum. -- Sent from mobile device On Nov 9, 2009, at 12:11 PM, aster...@opensourcesolution.in wrote: hi all, i have installed asterisk on Centos 5.3, plz i had installed one asterisk machine and two windows machine. now i want to install softphone in both windows machine. and both softphone should communicate with each other. any support and guidance will be highly appreciated. thx ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] how to check version of asterisk
Mr. aster...@opensourcesolution, if you had googled for how to know the asterisk version you would have found the solution right away. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, November 08, 2009 11:04 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check version of asterisk On Sun, Nov 08, 2009 at 06:20:46AM +, aster...@opensourcesolution.in wrote: hi all, i had installed asterisk under /etc. now i want to know by command which version of asterisk i had installed. how to know the version plz tell me. asterisk -V -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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 ___ -- 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
Re: [asterisk-users] Calls disconnects after short time
Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP/213.165.32.100-b7d21018' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7d21018, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7d21018, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7d21018, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7d21018, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7d21018, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7d21018, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7d21018, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7d21018' elastix*CLI ___ -- 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
Re: [asterisk-users] Calls disconnects after short time
The only informative part are the 2 paragraphs of the sip debug, but can't tell much since you only show a very small portion of the sip log. There is a 487 Request terminated there screaming at you but can't tell if meaning that provider is not handling the ACKs. That section of the [macro-hangupcall] context is useless as it is caused by the hangup, and not an effect. The usage of a public IP is not indicative of the existence of a firewall which can be blocking any necessary ports for tcp and/or udp. You should always cover your real IP numbers when showing samples of your logs CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Sunday, November 01, 2009 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time My server use public ip, so no nat issues, here is the out of sip debug: - --- (10 headers 0 lines) --- Sending to 213.165.32.100 : 5060 (no NAT) --- Reliably Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 elastix*CLI --- Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- Hungup 'IAX2/9-4490' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966599740196, 4) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7c10ad8, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7c10ad8, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7c10ad8, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7c10ad8, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' elastix*CLI thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP
[asterisk-users] [SPAM] RE: dCAP Exam
What about if I use the browser from my cellular phone? CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Wednesday, September 16, 2009 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dCAP Exam I believe the administrator can see what is on your screen with screen with those screen sharing stuff, this makes it harder a lil bit, and www.boratproxy.com becomes useless in that case. On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher tles...@digium.com wrote: On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote: Hmm...so by open book, that means access to the internet? Possible to get own notes ? Yes, you have access to the Internet, but your access is proxied, and the administrator of the test can see everything that you access. So it's best for you stick with only general guides and not look for crib notes. If your test proctor believes you cheated, you fail. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain dump sites. Or go to www.boratproxy.com and confuse their proxy. ah too fun. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pascal B. http://www.kameleonlabs.com/ Mike Ditka http://www.brainyquote.com/quotes/authors/m/mike_ditka.html - If God had wanted man to play soccer, he wouldn't have given us arms. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Platform decision ...
It all depends what are you going to use Asterisk for. Sounds like it is for conferencing. Would you care to elaborate? CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio Ferreira Brasil Sent: Tuesday, August 18, 2009 10:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Platform decision ... Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;): The main points listed on Asterisk's CONS that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT traversal; * Asterisk doesn't use SpanDSP; * Use of no longer maintained Berkeley DB1 engine as its internal database; * Asterisk doesn't allow CSRC entries in RTP; * Asterisk doesn't have an universal jitterbuffer for use with any channel type; * Asterisk doesn't use POSIX realtime extensions (having dependency with Zaptel timing); We were considering Asterisk as the chosen platform, but after reading this I got a little worried. The comparison considers 1.4 old version of Asterisk. So, can someone give me an update on what have changed for this items considering new 1.6 version ? Maybe someone can point me a site with an updated comparison. As long as I could see by now SpanDSP is present on new version of Asterisk, so this item isn't a difference any more. Right ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [SPAM] RE: newbie questions
Let me see if I get you: you inserted the installation CD, then you restarted the computer, and now you want to know what to do next? CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Poe Sent: Friday, June 19, 2009 11:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] newbie questions I have an Asterisknow.org CD. When I boot up, it seems ready for me to choose update, console, etc. I'm assuming I need to do something at the CLI prompt. Is there a tutorial that would take me from loading CD to making first test call? Computer is Dell Optiplex GX260 50GB free disk space 1.5GB RAM P4 processor external mic speakers Skype is on board, and would be good to use it, if possible. If I want to use Skype, do I need anything additional? Would it be better to install CD on my hard drive? Any help appreciated. Tom ___ -- 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 ___ -- 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] [SPAM] RE: SIP hacked connection?
Very few calls have been made this way, trivial cost, but it is slightly worrying. That's what I thought when they hacked into one of my systems, but it is not the cost of the calls, it is the purposed of the calls you should watch out for. The FBI contacted the owner of the PBX, and inquired him about calls being made from his company doing credit card soliciting. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Redstone Sent: Thursday, June 11, 2009 3:30 PM To: Asterisk User Subject: [asterisk-users] SIP hacked connection? Hi Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns). SIP ports mapped through firewall as we often connect from outside, but all SIP accounts have good passwords. However our telecoms provider picked up a few suspicious calls to places we do not normally call at times we do not often call. Looking at Asterisk logs it shows SIP session from the internet connected in and making calls with account IDs we do not recognise - definitely none of ours. Very few calls have been made this way, trivial cost, but it is slightly worrying. Anyone any ideas on how this could be happening? Thank Paul ___ -- 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 ___ -- 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
Re: [asterisk-users] Asterisk, SQL Database Update
Nothing is difficult my friend. If you dedicate a few cups of coffee to it, a couple of days, and do some good googling, you will get it done yourself. Good luck! CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, May 25, 2009 3:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk, SQL Database Update Thanks for your helpful reply. I am not so good in coding. simply all i need is as follow When a call comes, goes into an IVR, and then depending on the entry option it will connect to a remote SQL Database, to check the pre-existed data, and in the end of the IVR the caller will enter an option that will need to be written in the SQL Database. Can you please give me a general scenrio how this will be achieved. and which files that i will need to modify. Thanks a lot. Date: Sun, 24 May 2009 22:15:31 +0200 From: philipp.kemp...@amooma.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk, SQL Database Update Torintino T schrieb: Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Depending on what you are trying to do there are various solutions: Channel Event Logging (CEL) - http://www.asterisk.org/node/48358 AGI System() ODBC_*() functions Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 _ check out the rest of the Windows Live. More than mailWindows Live goes way beyond your inbox. More than messages http://www.microsoft.com/windows/windowslive/ ___ -- 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
Re: [asterisk-users] CDR feature not working properly for failed call attempt
As far as I know, just reinstall 1.4.20, and your problem goes away.
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas
Sent: Wednesday, April 22, 2009 4:12 AM
To: asterisk-users@lists.digium.com; asterisk-...@lists.digium.com
Subject: [asterisk-users] CDR feature not working properly for failed call
attempt
Hi Asterisk Developers/users,
I am facing a problem while using the cdr feature of asterisk(version
asterisk1.4.24.1). Whenever I make a call using a *.call file and it gets
failed , it don't produce the CDR for that channel as it falls into
OutgoingSpoolFailed channel As there is no such channel defined for
OutgoingSpoolfailed.. I am using this line in extension.conf for capturing
the failed atempt:
exten = failed,1,Set(CDR(userfield)=${HANGUPCAUSE})
I did a bit of googling and found that this problem is coming out with
Asterisk1.14.24 and it was quite OK in 1.4.17.
http://forums.digium.com/viewtopic.php?t=64832
http://forums.digium.com/viewtopic.php?t=64832highlight=outgoingspoolfaile
d highlight=outgoingspoolfailed
It seems to be bug with this new release , can any developer look into this
or give any workaround so for this feature to work in failed cases also.
Best Regards,
Parveen Jain
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Re: [asterisk-users] issue with sip 180 responses
I am having a similar issue. Asterisk does not show ringback tone and I investigated this due to it not reading sip invite 180. (or supposedly not receiving it).. My solution is that now I am using h323 (ver 1.4.19) CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yehavi Bourvine Sent: Sunday, April 19, 2009 4:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] issue with sip 180 responses Maybe it is buffering issues with the kernel? Does it happen only when there is a peak in the new calls rate? Do all 180 messages get dropped? __Yehavi: 2009/4/19 Nir Levi n...@bezeqint.co.il Hello, SIP invites are accepted from imitator , but 'SIP 180' is not responded back to imitator. By inspecting the issue , we can *see* the response is generated and sent from asterisk (via asterisk logger (sip debug )) , but while sniffing the interface with tcpdump, we can't see 180 response on the interface. We don't have errors on the interface, firewall is disabled , seems there's no packet lost (checked with ping with low interval ) , and routes are ok . By our tests we can see there is a direct connection between the mass of the calls, and between the lost of sip 180 responses. We're using Asterisk 1.4.4, with real-time configuration, also we made few *modifications* in asterisk source code (changed app_dial.c, /app_macro.c, /func_cdr.c). We're not sure if the problem on the OS Level (Centos 5.2) or in the asterisk application. Please assist. ~Nir _ This message was enriched by Impactia Technologies Ltd.www.impactia.com http://www.impactia.com/ Please do not enrich http://impactia.bezeqint.co.il/exclude.asp?email=asterisk-us...@lists.digiu m.comdomain=bezeqint.co.ilid=777 emails sent to me. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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
Re: [asterisk-users] retransmision error con asterisk 1.4.24.1
Alex, tu hablas español? CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Sunday, April 12, 2009 9:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] retransmision error con asterisk 1.4.24.1 Parece que Asterisk no recibe el ACK, el cual es necesario para establecer la llamada totalmente. Tal vez hay problema con un firewall o algo asi? troxlinux wrote: señores alguien le ha presentado este problema al acceder al voicemail o al hacer una llamada a la pstn 1940 Playing 'vm-received' (language 'es') -- SIP/111-08d91940 Playing 'digits/yesterday' (language 'es') -- SIP/111-08d91940 Playing 'digits/at' (language 'es') -- SIP/111-08d91940 Playing 'digits/8' (language 'es') [Apr 12 18:29:20] WARNING[3204]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded on transmission 4d242299a2de8...@192.168.10.23 for seqno 2468 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 12 18:29:20] WARNING[3204]: chan_sip.c:1998 retrans_pkt: Hanging up call 4d242299a2de8...@192.168.10.23 - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (netsoluciones, *981, 2) exited non-zero on 'SIP/111-08d91940' saludoss -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 ___ -- 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] [SPAM] RE: $20 Bounty
Dear Sir: Me no peak englisss…. 20 pesos??? ok , tank you sir, tank you sir From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kinjal Dixit Sent: Tuesday, March 03, 2009 9:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $20 Bounty I wish my family and I could live on $40 a week... simplify, simplify, simplify -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- 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
Re: [asterisk-users] Videoconference one-to-many
Asterisk compatible One to many video is achieved with VidPhone. You can download the web embedded video component free by signing up an account on my website www.itntelecom.com. Any help on usage, just send me a note and I will be glad to help you set it up. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Sent: Monday, February 02, 2009 7:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Videoconference one-to-many Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and video. So I can establish a voip + video connection *one-to-one* onlyit works OK. But I'd like to implement a videoconference *one-to-many* in order to intercommunicate many clients, is it possible with Asterisk 1.4 ??? (multicast is better than brodcast in this situation of course) Thanks a lot, Alejandro ___ -- 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 ___ -- 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
Re: [asterisk-users] TAPI and Asterisk
I recently completed PhoneClient 1.2 which is a Windows executable that interfaces with Asterisk, with capacity to receive numbers from the clipboard via a hotkey. PhoneClient is also a call manager and meeting room interface. If interested please contact me off the list. C. Savinovich -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Friday, January 30, 2009 10:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TAPI and Asterisk Funny how a topic will come up that you have never dealt with before, and suddenly it comes up from multiple directions at the same time. I was recently involved in a meeting where TAPI (which I understand only vaguely) was proposed as way to link a custom application to Asterisk for outbound and inbound call processing, much like SugarCRM and probably others are doing. Today I was asked by an existing client if I knew a way to synch their mobile device contacts with the system in some way so that they would have quick access to speed dial or otherwise call up a personal directory on their (Polycom) phones that could be synched in this manner. It struck me that the Polycom directory interface is a bit kludgy for such things, having no search capability and no sorting capability once loaded that I am aware of. Given the meeting last week it seems that a more elegant solution would be to link Outlook itself with Asterisk, so right clicking a contact makes it possible to launch an outbound call. That would take care of integrating a WHOLE LOT of devices, as (sadly) the MS contact database would be the go-between that all of these devices synch with in one way or another already. Is TAPI the right protocol to investigate for this purpose? Would something like Fonality's HUD software bridge this gap? Has this wheel already been invented? Hoping for some thoughts! Cheers, j ___ -- 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 ___ -- 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
Re: [asterisk-users] TAPI and Asterisk
Yes it is shareable. Thanks for the interest :)... just that I feel I am a few days away from the announcement CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, January 30, 2009 3:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] TAPI and Asterisk Why off-list? If you have an open-source application to share with us, why not share it here? If it's a commercial application, maybe it would be better to contact your prospect directly. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Friday, January 30, 2009 12:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] TAPI and Asterisk I recently completed PhoneClient 1.2 which is a Windows executable that interfaces with Asterisk, with capacity to receive numbers from the clipboard via a hotkey. PhoneClient is also a call manager and meeting room interface. If interested please contact me off the list. C. Savinovich -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Friday, January 30, 2009 10:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TAPI and Asterisk Funny how a topic will come up that you have never dealt with before, and suddenly it comes up from multiple directions at the same time. I was recently involved in a meeting where TAPI (which I understand only vaguely) was proposed as way to link a custom application to Asterisk for outbound and inbound call processing, much like SugarCRM and probably others are doing. Today I was asked by an existing client if I knew a way to synch their mobile device contacts with the system in some way so that they would have quick access to speed dial or otherwise call up a personal directory on their (Polycom) phones that could be synched in this manner. It struck me that the Polycom directory interface is a bit kludgy for such things, having no search capability and no sorting capability once loaded that I am aware of. Given the meeting last week it seems that a more elegant solution would be to link Outlook itself with Asterisk, so right clicking a contact makes it possible to launch an outbound call. That would take care of integrating a WHOLE LOT of devices, as (sadly) the MS contact database would be the go-between that all of these devices synch with in one way or another already. Is TAPI the right protocol to investigate for this purpose? Would something like Fonality's HUD software bridge this gap? Has this wheel already been invented? Hoping for some thoughts! Cheers, j ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] how to build a small asterisk pbx
Type how to build an asterisk PBX in google CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nightduke Sent: Sunday, January 25, 2009 9:09 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to build a small asterisk pbx Hi i must build a small phone pbx system. My friend has : 3 phone analog lines 6 phone extension How can i build that thelephone system? Nightduke ___ -- 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 ___ -- 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
Re: [asterisk-users] bridge 2 calls
None of these examples actually create a 3-way call, which is, unless I am mistaken, the original request. An incoming/outgoing call gets bridged to a local channel alright, but then how do you bridge that call to yet another call?. I did try some alternatives and the only way I found is by using a meeting room. Not too elegant in my opinion although it works nicely. If anyone knows of a better way please tell me. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Wednesday, January 14, 2009 6:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bridge 2 calls I use post variables. I found this on the web. Forgot where I got it from (sorry that I can't give you credit). ?php //Connect to the Asterisk Manager $socket = fsockopen(127.0.0.1,5038, $errno, $errstr); fputs($socket, Action: Login\r\n); fputs($socket, UserName: username\r\n); fputs($socket, Secret: password\r\n); fputs($socket, Events: off\r\n\r\n); fputs($socket, \r\n\r\n); fputs($socket, Action: Originate\r\n); fputs($socket, Channel: SIP/.$_POST['first_call'].@my_peer\r\n); fputs($socket, Context: mycontext\r\n); fputs($socket, Exten: .$_POST['local_exten'].\r\n); fputs($socket, Priority: 1\r\n); fputs($socket, Callerid: 5551212\r\n); fputs($socket, Timeout: 10\r\n); fputs($socket, Variable: FOO=.$my_var.\r\n); fputs($socket, \r\n\r\n); fputs($socket, \r\n); fputs($socket, Action: Logoff\r\n\r\n); fclose($socket); ? - Original Message - From: Nick Wolf new...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 06, 2009 12:18 PM Subject: Re: [asterisk-users] bridge 2 calls I am also interested in establishing a three way conversation using a simple webpage. I wonder if anyone can provide some help on that. On Tue, Jan 6, 2009 at 7:29 AM, amit mehta amit.magn...@gmail.com wrote: Hi Rilawich, I worked recently on it and that is why can give you the idea how i achived it. You can write an PHP script to get the number and name of the customer.You can phpself to the script.Then you can use an API script to use that number to orignate the call.The channel will be used to call the asterisk internal agent and the other line will call the number that was input by the customer and bridge the call. Hope this might help you. Regards, Amit Mehta Cell: +91 9898340962 On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] Needs more cpu usage
Your asterisk is using 99.9% of cpu and you want it to use more? Do you mean you want asterisk to use LESS cpu? CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Kim Sent: Friday, December 19, 2008 11:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Needs more cpu usage Hi, I am running * on centos5 using 4core cpu. When it is busy, * uses 99.9% of cpu max. How can I make * to use more cpu power? Thanks. ___ -- 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 ___ -- 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
Re: [asterisk-users] Authorize Microsoft SQL
Greg's question is this: - Does anybody has a sample on how to open and query a Microsoft SQL database from the dialplan?(and which are the correct drivers/addons to install?) Thanks CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Wofford Sent: Thursday, December 18, 2008 11:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL There is some code somewhere on the Asterisk/Linux box getting the SQL data, be it a program, script or batch file. There is something initiating the T-SQL code... SELECT * FROM supportcases WHERE id = 123456789 This code comes from the client, not the server. The Asterisk box will have the database drivers (ODBC...), but that just allows a connection, there is something that tells the server to return data (via the query). You are going to have to write the script (middleman) and pass it on from SQL to Asterisk. I don't know of anything like this ready-made. 1. DialPlan collect @number from caller 2. Call script, program etc and use the @number as a parameter 3. The script, program etc will the create the SQL Query to query the database: SELECT COUNT(*) FROM supportcases WHERE @number = 123456789 4. The script, program etc will then get the number of rows returned, hopefully 1 or 0 and assign it as a variable. 5. Your script, program etc with then use the following logic: If @variable = 0 Then Play enter your case again Voice Prompt ElseIf @variable = 1 Then Connect to Agent... HTH, Steve Wofford www.uctrlit.com P.(949)743-0233 Ext. 200 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack Sent: Thursday, December 18, 2008 20:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL Steve, my friends setup does not utilize perl/php code. His communication is directly between asterisk and mysql, there is no middle man. This is what I was hoping for with ms sql. But it doesn't sound like that will be the case. Thanks for everything! Greg -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Wofford Sent: Thursday, December 18, 2008 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL This is exactly what you need. Get your friends perl/php script and the SQL code will be near identical, or at least you will have no problem changing it yourself even if you don't know SQL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack Sent: Thursday, December 18, 2008 20:13 To: f...@teamforrest.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL This much I already know. This information is easily found through a simple google search. What I'm looking for is if anyone knows what a dialplan would look like that would perform an ODBC query to an ODBC database. I've seen minuet documentation on ODBCget, which is what I'm thinking will do the trick, but as I said the documentation on this is so vague that I'm not quite understanding it. There's also the possibility that there is another option here that I'm not seeing. One idea Steve gave me, was to create a perl/php script that does the query and returns a result code. Basically acting like a middle man between asterisk and the MS SQL database. Greg -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posner Sent: Thursday, December 18, 2008 9:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL All you need is odbc and freetds. Then it will integrate very smoothly. Fred Posner f...@teamforrest.com Direct: +1 (503) 914-0999 -Original Message- From: Steve Wofford s...@uctrlit.com Date: Thu, 18 Dec 2008 19:46:36 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Authorize Microsoft SQL ___ -- 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 ___ -- 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 No virus
Re: [asterisk-users] Text messaging and Asterisk
Thanks, excellent point. Furthermore, a google search on fastsms.conf yielded the existence of a couple of 'Asterisk SMS gateways'..wow CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Tuesday, October 14, 2008 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Text messaging and Asterisk C. Savinovich wrote: Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) Thanks C. Savinovich I don't use it but on my Asterisk 1.4 slug there was a file /etc/asterisk/fastsms.conf which had info about connecting to SMS services for about 4c per txt. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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 ___ -- 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] Text messaging and Asterisk
Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) Thanks C. Savinovich ___ -- 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
Re: [asterisk-users] Text messaging and Asterisk
I mean is if someone know of an sms server or service that allows me to send outgoing text messaging. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Monday, October 13, 2008 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Text messaging and Asterisk C. Savinovich wrote: Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) IIRC, asterisk currently supports sending text messages only when voice call is already established, so not very usefull yet... PJ Thanks C. Savinovich ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] Text messaging and Asterisk
I am trying to send text messaging to one caller, maybe about 40 of those per day, whose phone service have expired. All these callers are calling from their cell phones, and I have their caller ids. I will like to send each of them an individual text message (not an email) saying you attempted to use the service, but your service has expired Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Chamberlain Sent: Monday, October 13, 2008 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Text messaging and Asterisk On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote: I mean is if someone know of an sms server or service that allows me to send outgoing text messaging. Are you sending SMS to known users or to any mobile phone user? If you are sending to a fixed user base, track down the email to SMS gateways for their carriers. Then sending an SMS is no different than sending an e-mail. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- 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 ___ -- 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] Question on using DMZ
I am tinkering with a new router, a Linksys wrt610n dual-band, etc. But the when I connect it, the softphones(x-lite) on the computers don't even register. After a couple of hours of hassle, I found out that if I dmz the router to the computer I am using, the softphone starts to work. Problem is, there are about 6 computers in this office, all using x-lite. Can anybody suggest what to do here to so that I can enable all 6 computers connected to this router? Thanks CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Sent: Wednesday, October 08, 2008 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] retransmitting NAT Hi, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it stops there until it reached the timeout i set then it will say unavailable. is there a way that realtime will know that the phone is not registered anymore? or what could be causing the retransmitting of NAT? has anyone encountered the same prob? thank you regards, nhadie ___ -- 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 ___ -- 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
Re: [asterisk-users] Seemingly easy question: NPA/NXX
I don't see where it is difficult to figure out. First of all, system keeps looking up on the table as user dial each number. When number starts with 1, expect USA. When number doesn't start with either 1 nor 0, expect USA too. When number starts with 011, and as country code and city code is identified, expect as many numbers as determined by country+city code (once you know country and city code, you know how many local digits to expect) CS Question: How does the local Telco know you're done dialing a seven digit number? Easy you may say: If your dial string begins with 1, the parser expects 11 digits total, otherwise seven, 011 is international. The reason suspect it's more complex is that: 1) International numbers can vary widely in length and 2) Our local analog Telco will route a ten digit NANP numbers with no leading 1 and with no terminator--seemingly instantly Obviously this could be done with 'timeouts'--implicitly 'sending' after a delay. But it works so well I suspect there's more logic in there. For example I have dozens of ATA's provisioned with timeouts, and I find it difficult or impossible to replicate the Telco dialing experience (Either the delay is too long, or you have frequent 'reorder' tones because it 'sent' before you were finished). Therefore I assume that there is something more 'fancy' going on. Can someone validate, debunk or clarify this? Theory 1 Is it all done with timeouts, but they're CONDITIONAL timeouts. i.e. give a LONG timeout if the number: -did not start with a 1 and is still shorter than 7 digits, -started with a 1 and is still shorter than 11 digits -started with a 011 and is shorter than the theoretical international minimum lenght Theory 2 As you know, a few years ago the 2nd digit of the NPA was always 1 or 0. Therefore the switch could easily determine(without the leading 1) if your first three digits were an NPA or just an NXX (exchange). They were nationally unambiguous. Now that's no longer true. STILL, it could be possibleto consider all known valid NPA's and exchanges so they can determine via context what you're trying to do, and thereby optimize the dialing experience? Can anyone speak to this? I would very much appreciate any knowledgable input. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
It's amazing... the man starts the thread with a simple question: Can anybody tell him if Asterisk can do the same things that the Cisco Unity Server can do?, if it can do some better, some the same, and/or some worse, can someone indicate which ones? Also, can Asterisk complement the Cisco call manager functionalities?... I wish I knew the answers, and I am myself interested in the educated straight opinions of some of the members of this forum. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Tuesday, July 22, 2008 2:15 PM To: Asterisk Users Subject: Re: [asterisk-users] Cisco vs Asterisk Alex Balashov schrieb: The question is: 1. What are you trying to do? 2. Can Asterisk do it? 3. Can Asterisk do it well? 4. Can Asterisk do it at the scale, volume and scope you're looking for? The question is NOT: 1. Is Asterisk basically like a free version of CallManager? 2. Can Asterisk duplicate CallManager? Come on. People want simple answers. So: Can Asterisk duplicate CallManager? [y/n] *scnr* Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MagicJack quality
I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Better handling of the packets, that's for sure. Also, the algorithm is smart, and flexible... that being said, it opens more questions than answers. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, July 11, 2008 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality On Fri, 11 Jul 2008 17:13:15 -0400, C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? More memory CPU power then the average hard phone? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Yes, I have designed two different webphones, granted, using third party libraries, and magicjack's quality is better. I acknowledge that. Thank you, but referring me to someone's review won't help me much... I am interested in the internals. Regardless, their technique has a twist, and I am a naturally very curious *technical* fellow. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Friday, July 11, 2008 5:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MagicJack quality I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
As per the ads, if people ignore them or not, doesn't matter. Advertisers will fall in love with the idea that the venue reaches 1 million people, or more. As per the price of the service, they might be calculating the fact that the average monthly consumption of minutes on a softphone could be lower that the average monthly consumption on hardphones. After all, having to have that cpu on to make the call, is a drag. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, July 11, 2008 6:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality I don't see Magicjack being around long. The business model isn't sustainable without tons of ads, and even then, people will either ignore them if they are audio or if they are popups, they will simply close them or disable them. I might buy one just to hack it. Has anyone sniffed it or poked around at all on lists? Thanks, Steve T On Fri, Jul 11, 2008 at 6:03 PM, C. Savinovich [EMAIL PROTECTED] wrote: Yes, I have designed two different webphones, granted, using third party libraries, and magicjack's quality is better. I acknowledge that. Thank you, but referring me to someone's review won't help me much... I am interested in the internals. Regardless, their technique has a twist, and I am a naturally very curious *technical* fellow. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Friday, July 11, 2008 5:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MagicJack quality I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centile ipbx, anyone heard of this?
Of course you can ssh. And you can trace whats going on at 3 different levels. You can also open a trouble ticket with them. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Tuesday, June 24, 2008 2:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Centile ipbx, anyone heard of this? On Tue, Jun 24, 2008 at 5:58 AM, C. Savinovich [EMAIL PROTECTED] wrote: To be fair, Centile is better geared than asterisk for virtual pbx hosting. It comes with a system to manage virtual pbxs... it also handles the provisioning of most ip phones adequately, it is a totally different pbx although linux based. Interesting. Yes, it has a few phones it knows how to provision. I am using generic SIP device for both the phones currently in use. Although I don't know the details of your setup, it would not surprise me to see Centile accepting 2 different phones with the same extension on the same pbx. Well, my 4AM brainstorm didn't help. The phone I'm having trouble with is my favorite one, a Siemens S675IP. It is registered and works perfectly with 5 other SIP providers. On the Centile pbx, it can make calls but it can not be called. The web admin interface shows the correct public and NAT ip addresses and shows the phone in service. Calling it from another phone rings once and then goes to congestion, or at least that's the signal I hear. (It's wierd not being able to ssh in and see what's happening.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centile ipbx, anyone heard of this?
I do. I spent about a month doing contract work for a company that provides pbx hosting. Their hosting is based on Centile. What do you want to know?... they had about 80 customers (virtual pbxs), 320 ip phones... it seemed to be running ok. I could have done the same thing with asterisk :) CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Monday, June 23, 2008 3:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Centile ipbx, anyone heard of this? On Mon, Jun 23, 2008 at 7:31 PM, EdPimentl [EMAIL PROTECTED] wrote: They have been around for over 8 years, and their HQ is now in France... As is mine! But does anyone know anything about this other than their own docs? r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centile ipbx, anyone heard of this?
To be fair, Centile is better geared than asterisk for virtual pbx hosting. It comes with a system to manage virtual pbxs... it also handles the provisioning of most ip phones adequately, it is a totally different pbx although linux based. Although I don't know the details of your setup, it would not surprise me to see Centile accepting 2 different phones with the same extension on the same pbx. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Monday, June 23, 2008 9:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Centile ipbx, anyone heard of this? On Mon, Jun 23, 2008 at 10:01 PM, C. Savinovich [EMAIL PROTECTED] wrote: seemed to be running ok. I could have done the same thing with asterisk :) Basically, I was really curious why a company would use this instead of asterisk. Apparently, as I said, there is a reference in the docs, something like if you use asterisk on the same machine (or maybe network), ... I understand why people use SER, but wanted to know about this. When you google centile there's little info. I couldn't sleep, woke up in the middle of the night,, I think I may have solved the problem I was having but I can't test the solution until it's a decent hour (now=4AM). The problem was totally idiotic. I had left a phone on another site and it was registering to the same account as a phone on this site. I believe the way Centile deals with NAT may be part of why it's different from asterisk. Thanks for your comments, /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Click-to-call client
Check the web embedded click-to-call solution from videoreps.net. It is free. It includes click-to-video, click-to-call, and click-to-did CS From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Wednesday, April 16, 2008 7:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Best Click-to-call client Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? Thanks ___ -- 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
Re: [asterisk-users] cdr_custom outout to serial port
No problem. The program is in Windows. Contact me off line to make arrangements to send you the installation files. C. Savinovich Long ago, I wrote a nice program that reads CDR output from any legacy PBX via the serial port. Not much in use lately, but I will be happy to furbish it with mysql output to anyone who asks. Yes, please. What OS does it run under? Thanks! Doug ___ -- 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
Re: [asterisk-users] cdr_custom outout to serial port
Anything that you can make available? I'd love to be able to get our Definity's CDR into a MySQL database. Long ago, I wrote a nice program that reads CDR output from any legacy PBX via the serial port. Not much in use lately, but I will be happy to furbish it with mysql output to anyone who asks. C. Savinovich ___ -- 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
Re: [asterisk-users] Click to call
You can try the click-to-call from www.videoreps.net... it is asterisk based. The sample provides you with an actual pc-to-pstn call... of course calls to internal extensions are easier. There is click-to-call, and there is click-to-call-with-video CS somebody knows some application web that allows me to call to my internal extensions of my asterisk, example click to call. I was proving the click to call of this example but it doesn't work ___ -- 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] Asterisk connect to Cisco As5400 gateway
Yes... and there is plenty of information about sip-to-sip communications if you do research CS i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of using the E1 PCI cards in asterisk box ,is this practically possible? can i use SIP in the connection between Asterisk and Cisco AS 5400 Gateway? ___ -- 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
Re: [asterisk-users] Voicemail playback on iPhone
It is doable. The iPhone uses a subset of the Apple OS. Sometime ago I reviewed the file structure of the iPhone. It is just a matter of placing the voicemail files from * into the voicemail folder of the iPhone. Somebody with more time than me though :) CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Lixfeld Sent: Wednesday, October 24, 2007 7:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail playback on iPhone Sorry, it's clear my question was far too vague. To clarify, is there a recipe to make * record voicemail in a format that allows playback on iPhone's media/music player playback for voicemails that are received say, in an email message. It seems the * voicemail defaults don't work. This link seems to describe codecs that do work, however I haven't seen any indications as to whether * voicemail can be tweaked to record in any of the supported formats: http://www.kehlet.cx/ Any success out there? On 22-Oct-07, at 7:38 PM, Ron Stephan wrote: Trick question I assume? It was mind numbingly simple on my iPhone...(though none of the voice mail worked when London a few weeks ago). - tap voice mail - - tap speaker (upper right) until it turns blue (is activate) - tap the message you want to playback - use assorted controls to delete - replay etc. Now...if the question is ... how do you get asterisk voice mail to show up on an iPhone...I am all ears. Groovy concept - if anybody has a hack - I'd love to see it. Elvis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Jason Lixfeld Sent: Monday, October 22, 2007 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail playback on iPhone Anyone managed to get this to work? What's the recipe? ___ --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 __ NOD32 2607 (20071022) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --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 ___ --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 ___ --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] Announcing: Click-to-Call with VIDEO
Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. The video technology integrates with Asterisk, and it also allows the agent to video-conference up to 10 callers at the same time, making it an ideal tool for online seminars, demos, shows, tech support, etc, etc. Click-to-call WITH VIDEO can be used in: 1) Customer service mode (web click-me buttons) 2) Conference mode (meetings of up to 10 participants) 3) Video-Call mode (enter user destination number and video-connect to him/her) It is an ideal add-on for pbx vendors who want to add video services to their pbxs. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Here is how it works: Register online and you will receive an email with the confirmation link. With your new username you will be able to enter the operator's work area. Please fill your profile with your web site's ip (so that the buttons work), your destination phone (so that the phone call works) and a meeting room username (so that the videocall works). Go to the demo buttons page and click on anyone you like to download the button and kit with a couple lines of php code you need to place the button in your web page. Next, assuming you have a camera connected, click on GoOnLine and you will be online!! (don't eat at your desk!). I appreciate your kind input on this web site. I am at my desk most of the time (logged in as a video agent), and I will be glad to be available when you click on our web site button, or when you enter my number 19176135931 on the meeting room for quick-connect videophone. However, hyper as I am, I would prefer to coordinate a demo time with anyone who wishes a demo session. Best Regards Christian Savinovich VideoReps.net Note: Being an ActiveX component, please use internet explorer. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] Announcing: Click-to-Call with VIDEO
OK gentlemen, thank you very much. Best Regards C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, September 20, 2007 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO On Thursday 20 September 2007, C. Savinovich wrote: We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. I am one of the longest running developers of Asterisk and I agree with Anselm's assessment. Please post announcements like this in the future to the asterisk-biz list, as that list is for business discussions. This list is for users of Asterisk, not for commercial announcements. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] Announcing: Click-to-Call with VIDEO
Dear Dean: It is a very cool technology indeed, and please, do not see me as your competition, but as a friend. I know you have a click-to-call product, and if there is any way I can be of help with providing the video technology for you, I will be glad to set it up for you. You are most welcomed to use videoreps for free until you establish how it can benefit you. C. Savinovich VideoReps -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, September 20, 2007 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO I was interested in it - commercial or otherwise.but only because I used to work for the competition. Commercial or otherwise it looks like a very cool technology and something I'd be interested in - but only as a one of purchase price rather than an ASP. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Smith Sent: Thursday, 20 September 2007 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO He changed the title of his response. Your post remains intact on the list in its original form. In the interest of letting others decide, I think it was a spammy post as well. Others have decided... Hence the title of the list Asterisk Users Mailing List - Non-Commercial Discussion. They decided it a long time ago. When you included pricing, your email became commercial, an advertisement, and spam. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Thursday, September 20, 2007 4:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz list, and I do not read it for a purpose), but I do not use the integrated exploder for lack of the necessary obfuscation system on my work machine. Please do not send commercials, ads and product information to this list. It might very well be considered SPAM. Just and only because _some_ readers might be interested there is no legitimation for sending it (else every pen15-en1argm3nt _might_ trigger interest at some readers). Thanks Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list
Re: [asterisk-users] [PHISH] Re: ***SPAM*** Announcing: Click-to-Call with VIDEO
Please honor your statement below, We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. I take your request as an official demand from the community to provide the service for free for a longer time. We are happy to work with you guys. Herewith, for anyone who signs up until Sunday September 23, will get 3 months free. I will personally go through the database and extend everyone who has signed with the committed time. We sincerely think we were making our 2 cents contribution to the asterisk community by announcing an innovative concept on this forum. We apologize if we have inconvenienced anyone but are nevertheless glad to be of any help. Don't want to abuse our welcome here anymore, so we will be happy to respond to personal inquiries. Best Regards C. Savinovich -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, September 20, 2007 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Speaking for the Asterisk community as a whole, we demand that it be free forever. Please honor your statement below, We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. C. Savinovich wrote: Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz list, and I do not read it for a purpose), but I do not use the integrated exploder for lack of the necessary obfuscation system on my work machine. Please do not send commercials, ads and product information to this list. It might very well be considered SPAM. Just and only because _some_ readers might be interested there is no legitimation for sending it (else every pen15-en1argm3nt _might_ trigger interest at some readers). Thanks Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Partitioning DSL input
Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). Many Thanks C. Savinovich ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] Partitioning DSL input
Looks good. a lot of initial work, but looks worth the effort. Do you find that it improves the quality of your VOIP calls? C. Savinovich From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar Sent: Monday, September 10, 2007 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [PHISH] Re: [asterisk-users] Partitioning DSL input pfSense works very well for this. You can use it to setup VLANs (one for your PCs, the other for your VoIP equipment), and it has a traffic shaping/queuing mechanism for prioritizing VoIP. AR On 9/10/07, C. Savinovich [EMAIL PROTECTED] wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). Many Thanks C. Savinovich ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ http://www.astricon.net/ --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 http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] Quintum tenor configuration with asterisk help
I don't have much details on your set-up, but I assume that since quintums had performance troubles with SIP (about 2 years ago) your best bet is to get them to work with h323. For that your first step willl be to install h323 support on your asterisk box. I may be a little rusty on this, so if anyone has better advice, welcome CS Hi I need help configuring a quintum box with asterisk. Anyone has it working ? Thanks, Please let me know what I should do. I want to be able to register the asm200 with an extension, and be able to hopoff calls when calling from my asterisk, Thanks, On 9/9/06 6:47 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Another (quick) Polycom 501 question (Kevin Smith) 2. RE: asterisk-users Digest, Vol 26, Issue 54 (FRANCISCO PEREZ-LANDAETA) 3. Re: Call Processing Slow 11 seconds ([EMAIL PROTECTED]) 4. Re: Zaptel-1.2.9 compile error (Samy Antoun) 5. Problems configuring Polycom 301 (Jim Freeze) 6. Re: Zaptel-1.2.9 compile error (Nigel Godfrey) 7. ztdummy installed but choppy audio warning on load (Nigel Godfrey) 8. Re: ztdummy installed but choppy audio warning on load (Daniel Pocock) 9. Re: Zaptel-1.2.9 compile error (Samy Antoun) 10. Scope of contexts (Rene) 11. Re: What don't I get about SIP? (John Marvin) 12. Re: Scope of contexts (Doug Lytle) 13. Re: Scope of contexts (Moises Silva) 14. Re: Grandstream GX-2000, doesn't send calls to free lines (Zeeshan Zakaria) 15. Re: How to send correct Caller ID on PRI (Zeeshan Zakaria) 16. Re: How to use Grandstream GX-2000 phones for paging (Zeeshan Zakaria) 17. Re: Grandstream, how to use the configuration tool (Zeeshan Zakaria) 18. Re: Roundrobin not working on PRI (Zeeshan Zakaria) 19. Using option 'r' in queue doesn't announce frequeny etc. (Zeeshan Zakaria) -- Message: 1 Date: Sat, 09 Sep 2006 15:24:44 -0400 From: Kevin Smith [EMAIL PROTECTED] Subject: Re: [asterisk-users] Another (quick) Polycom 501 question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Mike, As far as I know, you need to at least start the dialing (ie New call, speaker, etc) for the digitmap to even come into play. The only settings that I am aware of that you can try to change are dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf. Kevin Mike wrote: Hi all, That's my last one for a while (I hope). How can I (if at all possible) make the 501 turn on the speaker phone as soon as a digit is dialed (if the handset is not lifted)? Sort of like what a normal speakerphone does. The reason I want this is I want the 501 digitmap to be taken into consideration even if the handset isnt lifted and the speakerphone button isn't consciously pressed. For all those users who don't want to press send, but like dialing without lifting the handset (and can't be bothered to press the speakerphone button). Yes I know it's capricious, but we have the users we have... Yes, I have read the admin manual, but couldn't find the info. I am assuming I just don't know what to look for, but that this functionality exists. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Message: 2 Date: Sat, 09 Sep 2006 19:48:27 + From: FRANCISCO PEREZ-LANDAETA [EMAIL PROTECTED] Subject: [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed hi i need helpl configuring a quintum tenor analog gateway using sip with asterisk. anyone, help is appreciated the model of the gteway is asm200 i need the settings to configure it with asterisk. for some reason it registers with asterisk but when try to call the extension from the quintum it is not recognized. help help help thanks From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: asterisk-users
[Asterisk-Users] Anyone knows how to receive a SIP call without registering gateway?
Hello everyone, I am pulling my hair here because a carrier threw me curve early today. They want to send calls to my asterisk server using SIP. Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation. Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox SIP way which is: client registers to my proxy. Guess what, I can't find any samples on this!!, Can anyone please help?, I will probably need a sample sip.conf. and then, to make a test call, I can use another asterisk box and try asterisk to asterisk sip calls (without register) via the cli prompt. But I have no idea and I am intrigued. Thanks CS ___ --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: test-ignore
Randy: Yes, you are right, thank you and consider this the end of this 'unfortunate' thread :) . See, I acknowledge it was wrong to send the test message, but that doesn't mean people have to be rude in pointing it out. Bye everyone CS -Original Message- From: Randy Bush [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 12, 2005 11:40 AM To: Christian Savinovich Subject: Re: test-ignore Look, please leave me alone. Don't bother me. Get a life. I started this thread with the title test-ignore, and in the body I wrote This is a test, please disregard. yes, sc is a rude ass. but posting test messages to a list is also rude and a well-known weenieism. perhaps you can resist having the last word in this stupid thread? randy ___ 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] test-ignore
This is a test, please disregard ___ 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] test-ignore
And what does that do? I am not testing no filter, I am testing the change of my name as it shows up on the list. I can't think of any other way. However, if turning off HTML has the desired result, I thank you for the tip. BTW: I can only dedicate 5 minutes to this issue. Thanks CS On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote: This is a test, please disregard Steven Critchfield [EMAIL PROTECTED] Next time you post, make sure you turn off HTML. ___ 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] test-ignore
Brian: Do you mean Don Quixote and the windmill? CS Dante and the windmill? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 11, 2005 7:49 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] test-ignore Dante and the windmill? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 11, 2005 5:44 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] test-ignore On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote: This is a test, please disregard Next time you post, make sure you turn off HTML. And not that I expect anything productive to come of this part of a rant... If you(collective mass of people sending test messages lately) are testing filters, couldn't you just wait the 10-15 minutes it takes for more messages from this list to come in? -- Steven Critchfield [EMAIL PROTECTED] ___ 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] test-ignore
You know, you have a choice. C. Savinovich ITN-Telecom I don't know. Dealing with some people here seems like I am in hell tilting at dragons. On Tue, 2005-01-11 at 20:02 -0500, C. Savinovich wrote: Brian: Do you mean Don Quixote and the windmill? CS Dante and the windmill? I don't know. Dealing with some people here seems like I am in hell tilting at dragons. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 11, 2005 7:49 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] test-ignore Dante and the windmill? Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 11, 2005 5:44 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] test-ignore On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote: This is a test, please disregard Next time you post, make sure you turn off HTML. And not that I expect anything productive to come of this part of a rant... If you(collective mass of people sending test messages lately) are testing filters, couldn't you just wait the 10-15 minutes it takes for more messages from this list to come in? -- Steven Critchfield [EMAIL PROTECTED] ___ 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] ATA Adaptor
I don't mean to be patronizing at all, but one thing I've learned is that if a customer does not want to spend $150 on anything, either he/she is an incredibly shrewd businessman (because he is going to make you bust your balls for free until he sees results), or he is really not interested and he is cheap. In either case it is best to beat it and wait till the next customer. That's my grain of salt advice, maybe not for you, but for any developers who are initiating in the consulting business. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Taylor Sent: Monday, December 20, 2004 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ATA Adaptor On Mon, 20 Dec 2004 11:09:09 -0800, TC [EMAIL PROTECTED] wrote: Hi, I am new to asterisk and I am trying to get things set up so I can prove to the boss it works and get the budget to do a full implementation. Does anyone have an ata adaptor or an ip phone laying around they would be willing to sell me for around 30-50 dollars, I will need 2 of them. why not just spend the 75 bucks per ata http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-31891743744.htm is $150 too much for the boss's proto-type test ? ___ 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 Set up your Asterisk, load the X-ten softphone on a PC, call me and I'll loan you a real-world DID number to play with for a few days. If you can't prove it up with this, you're out of luck. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ 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] Help setting-up X-Pro behind a proxy
I am trying to set up XPro behind a Squid Proxy. What should I put in outbound proxy?, what is a STUN server? Thanks C. Savinovich ITN-Telecom ___ 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] Using meetme video mode with SIP ? Now a $2000bounty
Dean: We have a commercial videoconference product. The closest we can get is to initiate the VC based on the phone call started by asterisk, which would be really cool, but there will be a charge for the video software. C. Savinovich ITN-Telecom 212-865-9118 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Peter Svensson Sent: Wednesday, December 08, 2004 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Using meetme video mode with SIP ? Now a $2000bounty On Wed, 8 Dec 2004, dean collins wrote: There doesn't seem to be any interest in using asterisk and video. I posted a $1,000 bounty to get video meet me working without a single reply. I have now just bumped this to $2000 http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid eo+conferencing This is a legitimate commercially binding bounty, I hope this might inspire some people to develop at least something. There is something in OpenH323 for video conferencing. Perhaps some ideas can be learnt from that implementation. Peter ___ 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
