[asterisk-users] SIP FXS ATA with Gigabit ethernet bridge port,
Hi Folks, Is there any SIP FXS ATA with Gigabit ethernet bridge port, in the market ? -- Isamar Maia Cel. VIVO SSA: (55) 71-9940-2012 Cel. TIM SSA: (55) 71-9289-5128 Cel. Claro SSA: (55) 71-9146-8575 Fixo: (55) 71-4062-8688 Skype ID: isamar.maia A vida é muito curta para ser pequena (Benjamin Disraeli) -- _ -- 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] Bind port
Yes. More than 1 port as source and port forward doesn't work. Isamar On Mon, 4 Jul 2005, Carlos Alperin wrote: Eric, This is the Sip.conf section where you define the port. Do you want to use more than one port as Source? ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 69.39.69.183 ; Address to bind to canreinvite =no context = pstn ; Default for incoming calls And this is what I get from the port #2: SIP1-MI*CLI sip show peer carlos2 * Name : carlos2 Secret : Set MD5Secret: Not set Context : pstn Language : FromUser : FromDomain : Callgroup: (0) Pickupgroup : (0) Mailbox : LastMsgsSent : -1 Dynamic : Yes Expire : 340717 Expiry : 900 Insecure : No Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : xxx.xxx.xxx.xxx Port 5061 (This is my Real IP address outside the NAT, and it receive it on 5061. Defaddr-IP : 0.0.0.0 Port 5060 Username : carlos2 Codecs : GSM ULAW ALAW G.726 G.729A Status : UNKNOWN Useragent: Sipura/SPA2000-2.0.13(g) Full Contact : sip:[EMAIL PROTECTED]:5061 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Monday, July 04, 2005 6:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bind port Carlos Alperin wrote: Sipura boxes uses 5060 5061. I don't see why you cannot use something like that. But anyway, that depends on how you 'll going to register them. They use that for the SOURCE port of packets sent by the SIPura. It still uses 5060 as the destination port. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] Bind port
Yes. That's what I'm trying to sort out with SER. I just need to forward the packets. Anybody with a sample ser.cfg to do that? Isamar On Tue, 5 Jul 2005, Tzafrir Cohen wrote: On Tue, Jul 05, 2005 at 08:13:58AM +0900, Isamar Maia wrote: Yes. More than 1 port as source and port forward doesn't work. Why isn't port-forwarding good enough? Maybe run a separate SIP proxy on another port? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Bind port
Dear All, I need to bind two different ports at the same time for SIP. 5060 and another port number. Is it possible ? It would be something like port=5060,5062 Isamar ___ 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] What to use h323 or oh323 ???
It's a little bit hard to compile but Try oh323 first. Although, There will be some few situations that H323 will work better than oh323. So, have both. Isamar On Sat, 2 Jul 2005, Adeel -31 wrote: I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... i m comfortable in using sip iax softphones but there is no h323.conf in /etc/asterisk/ i read that i've to compile some files but i m confused regarding h323 oh323 .. which one should i use.. plz tell me or atleast give some helpful link __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soyo G688
Yes. it's a PA1688. IMHO, it works well for Home users but don't even think to use it for business applications. The great thinkg is that it works with IAX2. Isamar On Sat, 28 May 2005, Waldo Rubinstein wrote: I was referred to this URL: http://www.thevoipconnection.com/store/catalog/ product_16221_SOYO_G668_VoIP_Telephone.html - Waldo On May 27, 2005, at 7:32 PM, Isamar Maia wrote: Do you have any link? Isn't it PA-1688 Chip? Isamar On Fri, 27 May 2005, Waldo Rubinstein wrote: Has anyone had any experience with the Soyo G688 phone? I'd like to use it as a agent's phone. Is it reliable? How well does it work with *? How's the quality? Features? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soyo G688
Waldo, The external material quality is business-level. It looks like a toy. Depending on the business you will provide this as a solution, it will not be acceptable. But... it can be relative. For business, I would recommend http://www.act-tel.com.tw Forget the support from Soyo. Just confirm with them the model of PA1688 and download the firwmare you want from http://www.aredfox.com Isamar On Sat, 28 May 2005, Waldo Rubinstein wrote: I read about the PA1688 and, yes, it says to support IAX2. However, reading the PDFs on the Soyo G688, I found no reference to IAX2 at all. How certain are you that the Soyo G688 is based on the PA1688? Also, why do you not recommend using it for business apps? Thanks, Waldo On May 28, 2005, at 7:58 PM, Isamar Maia wrote: Yes. it's a PA1688. IMHO, it works well for Home users but don't even think to use it for business applications. The great thinkg is that it works with IAX2. Isamar On Sat, 28 May 2005, Waldo Rubinstein wrote: I was referred to this URL: http://www.thevoipconnection.com/store/catalog/ product_16221_SOYO_G668_VoIP_Telephone.html - Waldo On May 27, 2005, at 7:32 PM, Isamar Maia wrote: Do you have any link? Isn't it PA-1688 Chip? Isamar On Fri, 27 May 2005, Waldo Rubinstein wrote: Has anyone had any experience with the Soyo G688 phone? I'd like to use it as a agent's phone. Is it reliable? How well does it work with *? How's the quality? Features? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soyo G688
Do you have any link? Isn't it PA-1688 Chip? Isamar On Fri, 27 May 2005, Waldo Rubinstein wrote: Has anyone had any experience with the Soyo G688 phone? I'd like to use it as a agent's phone. Is it reliable? How well does it work with *? How's the quality? Features? Thanks, Waldo ___ 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] Problems with Public IP
Miranda, Looks like you have a codec problem. Be sure that all terminals are talking the same codec and if their settings in sip.conf or whatever they using has the allow= for the codec in use. Ex: If you are using G711u: allow=ulaw Um abraco! Isamar On 5/25/05, Virmones P. T. Miranda [EMAIL PROTECTED] wrote: HI All asterisk user I Have one Asterisk with this scenario: i have two ip Address one Private IP one Public IP, my internals terminals using private IP works very fine but my terminals using public ip don't work audio , make rings but streamer don't work. thks for you attetion best Regards ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] LOOKING TO HIRE
Ergo, the assertion Good Programmer = Compiled Languages is *pure bull*. Good programmer = assembly language! Just kidding ;-) Good programmer is who makes the things working well *as planned* in the time-limit planned beforehand, having good results for the *business* in the end-of-the-day. The rest doesn't matter. Isamar ___ 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] LOOKING TO HIRE
Good programmer is who makes the things working well *as planned* in the time-limit planned beforehand, having good results for the *business* in the end-of-the-day. The rest doesn't matter. Isamar Sometimes you have to do things in a boring and unelegant way. You want to do something fun and exciting but you can't be wasting other people's money. You have to understand that software development does not get 100% of the budget. Yes. You are absolutely right. In fact, this topic can be also related to the asterisk. A GPL system that works gracefully but its code is nothing that can be called perfect, even written in C :-) Isamar ___ 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] Voice Recognition - Cases of success
Hi Folks, I am planning to make a little project of voice recognition. I already browsed Voip Wiki and found some solutions. Before putting my hands on it to just do a little demo menu, I would like to hear from the list any succesful case using voice recognition and Asterisk. Best Regards, Isamar Maia ___ 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] Predictive Dialier
Hi Folks, Where can I find a list of Predictive dialer solutions for Asterisk? Thanks, Isamar ___ 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] Linksys/Cisco buys Sipura
I guess the prices will go up like a rocket Isamar On Wed, 27 Apr 2005, MF Hulber wrote: Have you seen this story? Cisco definitely wants to own the VoIP market. I wonder what effect this will have on Sipura products. http://story.news.yahoo.com/news?tmpl=storyu=/nf/20050427/bs_nf/33554 MARK. ___ 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] japanese voice files
Anybody would have the japanese voice files for *? I need now the number's recording at least. Thanks, Isamar ___ 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] Aculab
Jochen, Recently I contact Aculab in UK about that and They asked me to call Digium Sales. I called Digium Sales and they told me that nothing is confirmed yet about a deal between Aculab and Digium. Maybe something changed Isamar On Mon, 11 Apr 2005, Jochen Witte wrote: Hello, on http://www.voip-info.org/wiki-Aculab it has been said, that there is a Aculab card, which works with Asterisk. Two questions: 1. Which card is this? 2. How do I configure it with Asterisk / Linux? If anybody has any experiences regarding this, I would very much appreciate to get some more information on howto use it with Asterisk. Regards Jochen -- Jochen Witte [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
Re: [Asterisk-Users] Sangoma VS. Digium
Isamar Maia wrote: Technically speaking not. But Sangoma's support seems to be pretty much better. My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. They're glad to use Asterisk as a selling point for their hardware, but unwilling to donate anything back to the Asterisk community. I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. I don't understand this *love* for Digium. Digium is a commercial institution, period. If we need to be thankful for Mark Spencer for giving asterisk to the world as many say, I understand and agree. But to protect them specially in my case since I am in Japan and Digium products don't(and it seems that will never) have any support for NTT lines, is kinda no sense. I would better support the Asterisk Fork development that seems to be happening in the underground. BTW, anybody knows their mailing list? I'll be glad to contribute. Isamar Maia ___ 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] Sangoma VS. Digium
Isamar Maia wrote: I don't understand this *love* for Digium. Digium is a commercial institution, period. Yes, but. They are a commercial institution which took an enormous risk by giving away for free what is undeniably their most valuable product. So, if Linus Torvalds had a company I would need to buy products from him? If they assumed this risk, great! I will remember to send a postcard in the Christmas to them. More hardware companies support Asterisk with Zap drivers, cheaper will be the boards, better quality will be provided and in the end of the day, the community will have all the benefits. The name of it is competition. Or it's a monopoly? Maybe Japan or other countries with own crazy standards are not a commercial interest of Digium like they are for Avaya, Dialogic, Aculab and stuff... the open and free competition should happen because the world is not USA and AFAIK it's GPL. I'm sorry you have trouble understanding this. I feel that for many of us it is pretty clear. Yes. I see. Very clear. Isamar ___ 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] Sangoma VS. Digium
Remco Barende wrote: It would be nice if Digium would accept the bristuff patch at some stage and include it in asterisk. GPL code cannot go into the Asterisk distribution. Yes Steve. That's right. I have heard that any code going inside Asterisk distribution needs to give a paralel license to Mark Spencer. If it's true...This kind of thing Digium *lovers* should take in consideration. Isamar ___ 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] why even use SIP
For a easier comprehension, nowadays, H323 is like english. SIP is like spanish and IAX is esperanto. You can IAX. It's wonderful, modern, lot of advantages, pass through any firewall, blah...blah..blah... but you can find only some strange guys using that. :-) Isamar ___ 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] H323 and Cisco: one way problems
hi folks, I am calling from Asterisk to some Cisco gateways (as5350, 2600) and I am having one way problems with chan_oh323. With other provider that uses cisco also and I running the same chan_oh323, it works perfectly. I've tried also with chan_h323 and it does not work as well. Asterisk cvs head, chan_oh323 0.7.1 Have anyone experienced this problem? Thanks for any help. Isamar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + Call hangup
Giovani, Are you using a X100P ? In my case here for a similar situation, the same happens because the Zaptel takes sometime to understand the call was hangup. Try to play with Busydetect/busycount option in zapata.conf Isamar On Fri, 11 Mar 2005, Giovanni Miano wrote: Scenario PSTN - ZAP CHANNEL - ASTERISK - SIP When i recive call i fwd it to SIP Phone - SIP PHONE ringing If From External Line PSTN hungup call SIP Phone Ringing too, why ? ___ 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] FAX
I am using Underwood's fax system for fax on demand and it's very cool. I am planning to do the following and I would like to know if it's possible before putting my hands on it. For a specific application, I want to dialout thousands of numbers searching for fax machines. If somebody takes the call(voice), I would flag that number as bad in the DB. If it's a voice only answer machine, I would flag that number also as bad. But if it's a fax or an answer machine with fax, I would flag that number as valid fax number for future use. Is that possible? Thanks a lot, Isamar Maia ___ 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] FAX
Actually, it was requested to me to build a fax number database. The real purpose is unknown. I am an IT guy, not marketing guy. Isamar On Sun, 20 Feb 2005, Torsten Krueger wrote: Hello, On Sun, 20 Feb 2005, Isamar Maia wrote: For a specific application, I want to dialout thousands of numbers searching for fax machines. If somebody takes the call(voice), I would flag that number as bad in the DB. If it's a voice only answer machine, I would flag that number also as bad. But if it's a fax or an answer machine with fax, I would flag that number as valid fax number for future use. Is that possible? You are definitely in need of app_faxspam_harvest.so or am I wrong? Torsten Krueger ___ 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] FAX
Ok. I will be burned in fire.. :-) Or better.. I won't go to the heaven... Isamar On Sun, 20 Feb 2005, Andrew Kohlsmith wrote: On February 20, 2005 08:30 am, Isamar Maia wrote: I want to dialout thousands of numbers searching for fax machines. You are an evil, evil man. Worse than the goddamned telemarketers, IMO. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
Are you using PC or mac? Isamar On Thu, 20 Jan 2005, Wilson Pickett wrote: I would also suggest that while it is possible to do something, it is not always wise :) See the significant volumes of reports in the archives regarding multiple zaptel cards in one system. I must be lucky: I have 2 X100P and a TDM400 with zero IRQ or other issues. And double NAT for the voIP part. :) ___ 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] Japanese FXO card
Mick San, I wish you luck. I am in Japan also and like others in the list, what you have to worry less is about the configs.. get some samples in the wiki site and go for it. If you need some help, feel free to ask. But, be prepared for some headache to make it working well with NTT FXO lines Again, good luck! Isamar On Wed, 1 Dec 2004, Asterisk users wrote: Hi folks, Im totally new to * but I went ahead and told my boss that it was the way to go for our new telephone system :) now I have a test box and two cisco phones and a brand new modem card. Im having plenty of trouble with learning all the config stuff but ill leave that for another day. ie: a few days after I rtfm. My modem card, once installed in the box (FC2 by the way) was detected by linux and installed perfectly no probs. but now I dont know how to make * recognise it? How can I tell if it is even compatible with *? I dont think all the usual options of buying the compatible cards are open to me because im in Japan. We have a bunch of ISDN lines and TAs to use, so if im out of luck with the modem I bought perhaps Id have a better chance with an ISDN card? your thoughts/comments/suggestions are appreciated. cheers, Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323/*/IAX - Firewall - IAX/*/H323
Hi Folks, I have two H323 Polycom video conference system with a Linux firewall Iptables in the middle. I am not getting to make H323 working in this setup and I was wondering to put two * servers as a bridge to jump the firewall using IAX. The idea basically is: h323 Polycom IPTABLES VideoConference Device -- *(LAN) --- *(WAN) H323 Polycom chan_h323 chan_iax chan_h323 or chan_oh323 or chan_oh323 Question before spending some time with it... should it work ? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID for Japan?
Yes. It is possible. But a driver was not implemented for that yet. Isamar On Fri, 12 Nov 2004, Kuniyoshi Murata wrote: Hi, Does anyone know if it's possible to make Asterisk's Caller ID function to be compatible with Japan's Number Display system? TIA Kuni -- Kuniyoshi Murata.iChat/AIM:macwebcaster English-Japanese Interpreter mailto:[EMAIL PROTECTED] Macintosh Webcast Specialisthttp://www.macwebcaster.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem facing on Firewall, NAT and asterisk
Hi Prasad, Install a Asterisk in your DMZ and one Asterisk inside of your Lan. Set them to use IAX between them passing through your firewall. A) Your SIP Phones in your lan will connect to your LAN's *. B) The SIP Phones in the internet will connect to your DMZ's *. C) A connnects to B through the Asterisk's IAX connection SIP doesn't work with firewalls. Also, next time, post this kind of message in the asterisk-users list. Isamar On Wed, 3 Nov 2004, prasad_s wrote: Hi all, I am using asterisk, which is running on one machine having static(global) IP. I have another machine(Internet server with global IP, with firewall) working as gateway for internal machines having local IP starting with 192.168.xxx.xxx. My SIP client(xten-xlite) is on LAN machine and registers to the asterisk server through this sip phone. All machines on the LAN, having sip phone are registered to asterisk server. But the problem is when I call internally between two sip client I don't get voice path between these two sip phones, i.e. I can not talk and hear from both phones, though I get message on the asterisk server connected. Is this because of Firewall and NAT between my sip client and asterisk server? But then how I get register to asterisk server? Is there any workaround for this problem regards Prasad Somwanshi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using Voipjet?
[EMAIL PROTECTED] wrote: I've used them for calls terminating in the US with good results. I happened to put through a call to Romania today and it seemed the person was hearing me very much lagged behind. The actual asterisk IAX figure given was like 80 ms which is usually pretty decent for talking to someone. I'm using Voipjet from last week only, I regulary call to Spain, France and Peru and the quality is very good. I've used ulaw and ilbc with a ping time to their server of about 160ms. Using ulaw, the other side said that there's some echo of his voice, but with ilbc it is almost gone. With so long distances, there is nothing better than G.729. Isamar ___ 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] ATCOM froze
Hi folks, While upgrading the firmware of an ATCOM AT-323, the power was cut. Now, it just shows Booting... in the panel and freezes. Any thoughts? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phones -India
What I heard is that you can sell pones but you cannot provide VOIP or termination service over there. Isamar On Wed, 20 Oct 2004, Henry Devito wrote: HI I am in the US and have a customer using * in the US they just acquired a call center in India. Does anyone know if I can legally sell/ship Grandstream IP phones and IAXy's to India? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quicknet Linejack Asterisk PBX
It doesn't work. Period. If want more, tell me your address and I will send a couple to you. Dialout is not well or totally implemented in asterisk for this board. Isamar On Mon, 11 Oct 2004, FRANCISCO PEREZ-LANDAETA wrote: Hi, I am in the process of setting up an Asterisk PBX with some Quicknet Linejacks that I have. Has anyone been successful with this setup ? I have a PC with 7 Linejacks and would like to set it up as a PBX with two incoming lines and 7 extensions. These two lines will be to dial out and to receive incoming calls. For this setup I don¹t need voicemail or any of the fancy features, just a phone that rings so that one can pick up the call and transfer it. I am just wondering if the linejacks (7 of them) will work ok with Asterisk I am 100% sure that digium will work but I am not sure about the linejacks. I need to do this project and have it ready this week so your help and cooperation is appreciated. I know that many people don¹t like the Linejacks but this is a project and I must make it work. Are there any tricks to transfer the calls to other phones ? Thanks guys ! Frank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Free G.729 ready for download
Umar, I agree with you. The license price for each G729 is very reasonable and I don't see any reason or merit for this thread. I think Digium and Asterisk should have other priorities. One of them is the callerid stuff not working in some countries, specially Japan. Isamar On Sat, 25 Sep 2004, usedcanon wrote: Hi All, I consider the License fee charged by digium for G.729 as very reasonable, and hope people agree and do nothing to jeopardize this project. Right now I don't use G.729 at all, however if and when I do, I have no reason to seek an alternative to what Digium provides. At the very least I would be confident that I am in no way breaking the law, and have the satisfaction of have contributed back to the product, be it in a very small way. Umar -Original Message- ___ 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] spandsp / I get only garbage in my faxes
I am trying to use spandsp 0.0.1k-whole. I have 2 X100P working well for inbound/outbound calls. I have tried libtiff 3.6.1 and 3.5.7 With 3.6.1 I get only faxes all black, and 3.5.7, I get blank vertical lines and the rest all black also. During the transmittion, apparently there is nothing that can indicate any error except some Training errors. The sender fax thinks that the fax was sent successfully... Kernel 2.4.21, Slackware, Processor Duron 850. Any thoughts? Isamar ___ 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] Audiocodes Mediant 2000
Hi FOlks, I am trying to setup remotely an AudioCodes Mediant 2000 MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Audiocodes Mediant 2000
But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Google found this it may help http://corp.deltathree.com/productsandservices/manuals/bizlink.pdf I have seen that already... looking something more objective. I just read that and didn't understand anything Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT Easicom - Andy Powell
Sorry. I was not following the thread, but... What justifies this phone has this price of 99US$ while others are for retail from 75 to 85US$ .? Isamar On Sat, 4 Sep 2004, SeshKanuri wrote: Try this Phone at http://ipphone.eezeephone.com/ This Phone is listed now on ebay for sale at http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=11908item=5718863004rd=1 - Original Message - From: Andrew Newton [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 02, 2004 2:08 AM Subject: [Asterisk-Users] BT Easicom - Andy Powell Hi, I have been looking for info on * and the BT Easicom 1000 without much luck when i found a post to this list from Andy Powell saying that he had the phone working quite well. Before i go buy a shedload of these things I would like to know what problems/sucesses people have had with these phones and * in the UK. What they can/cant do with * y Also does anyone know of any good ADSI Scripting resources/tutorials? Many thanks Andrew Newton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma Card Support
Hi Folks, I found some old postings about Sangoma card support in * but nothing indicative if this is supported or not for dialin/dialout. I found only support indication for VOFR using Sangoma... Anybody other driver available for Sangoma even not free like chan_dialogic ? Thanks, Isamar ___ 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] Fax on demand
Hi folks, Anybody making fax-on-demand with * ? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax on demand
I wanna do it through IVR. I know how to create the .call files for normal outbound calls but how to attach the .tiff files ? Isamar On Mon, 2 Aug 2004, Brian McManus wrote: Yes I've implemented a simple web interface that generates a . call file that faxes generated .tiff files a Crontab checks against a database to generate the tiffs and .call files. B Isamar Maia wrote: Hi folks, Anybody making fax-on-demand with * ? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Linejacks
Me too. I spent several months to make it working and I figured out it cannot dialout... only dial-in. Sell it to someone else and buy and buy a TDM-04b. Isamar On Thu, 22 Jul 2004 [EMAIL PROTECTED] wrote: Nope...I scrapped that idea and just bought a Digium card. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of greg Sent: Thursday, July 22, 2004 2:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Linejacks I found a message from you to the asterisk users mailing list from 2001. I was wondering if you got (or still have) an asterisk system working with the linejack? If so, would you be willing to assist me with mine? I seem to have things working, and * says that caller ID is coming in, but I can't get * to actually answer the call. Thanks, Greg -- NetIO.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B Dead?
Just for curiosity, Let us know how much time you'll gonna get a RMA of it. Isamar On Thu, 22 Jul 2004, Andres Junge wrote: I had the same problem, and it was that the power suppply coudn't handle the new card. My solution (until i get a new power supply) was to unplug a very big fan that i have in the case. Salu2 Andrés Greg Hulands escribió: Hi, I just received in the mail my TDM04B card and put it in the computer, now the computer won't even show the video card bios or the post screen. From the digium website I could not find any specific requirements for the pci card, like 32 or 64 bit slot. The motherboard for the computer I put it in is an Asus A7V333 with PCI 2.2 compliant slots. I am thinking that maybe I just got a dud card. Is there anything I need to change or I can test to see why it is not letting the computer boot? Any help is greatly appreciated. Regards, Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cordless Phone Problem
I have one TDM04b(4FXO) that BTW came with a broken module and I'm sending the module to RMA. The other channels work well with one phone but with some specific brand/models don't work. For example: Sharp CJV-743W http://www.sharp.co.jp/products/cj/index.html#cjv743w Using the cordless phones or not, the sound in some calls is very low and sometimes is like when you put a shell in your ear. Other calls work perfectly. So, the problem is intermitent but with a big frequence. I did already some measures: 1) All the phone wire in the building was changed 2) I put the boards in another machine 3) The boards are not sharing IRQs 4) I'm using 2 wire cable 5) I already tried to change rxgain to several values 6) I have two of those Sharp phones with the same problem and I trashed already some other thinking that it was a phone problem. 7) I am using the latest CVS zap and * 8) I am using aggressive echo cancel with the new algorithm This machine has 1 TDM40b and 1 TDM04b and actually I don't know if it's a problem in one or other. The phones directly connected to the line works perfectly. Did anybody have a similar problem? Thanks, Isamar ___ 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] The Cellphone companies getting prepared
Looks like that the cellphone companies are getting prepared to any possible competition... http://www.thefeature.com/article?articleid=100878pos=1ref=1859764 Isamar ___ 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] Penalty in queues.conf
I have already read explanation about that in some places but I don't have still a clear image about the meaning of Penalty parameter inside of queues.conf What means that? Thanks, Isamar ___ 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] T100P in Japan (eh?)
I'm planning to buy a T100P for a project in the company where I work for but my concern is about the japanese ANI. Can I get somehow japanese(NTT) ANI working with T100P ? Feasible? Impossible ? Thanks, Isamar Maia ___ 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] Get back a failed transfered call
Hi Folks, I have the following situation: I received an inbound call in my extension A and transferred it to the extension B. But B was busy and I want to capture the call back to my extension. How should I proceed? Thanks, Isamar ___ 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] Playing the invalid extension input
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Any suggestions?
Thanks,
Isamar
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[Asterisk-Users] chan_dialogic
I'm planning to buy Dialogic licenses for one of my dialogic boards to use with *. I have already that in the drawer and it's boring me to keep it there with no use. Although, I have heard that it doesn't work for dialout and I would like to confirm if it's true... my plan is the following: Definity --- Asterisk w/ Dialogic -- Asterisk w/ Dialogic --- Definity D-ChannelVOIP/IAX D-Channel Since, I don't have VOIP in the Lucent Definity machines, I think it would be perfect integrated with asterisk and my dust cloud dialogic boards. So, I just want to confirm if it would work with the current chan_dialogic. Thanks, Isamar ___ 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] FAX x Echo Cancellation
I installed a TDM04b and a TDM40b with aggressive echo suppression and it's working almost perfectly. The problem is that all extensions are fax machines and people uses it for both purposes, voice and fax. AFAIK, I cannot use aggressive suppression for fax extensions, but when I turn it off terrible echos happen. Is there any workaround for this case? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to force G729
allow=ulaw Why don't you remove this? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell 400SC and X100P
Thanks a lot for replying. I turned on the ACPI in the CMOS and it got better. At least I call receive several calls in sequence and call out but it hangs up right after the person gets the phone in the other side. So, something is still missing. What is your ACPI mode in the CMOS ? S1 or S3? Which kernel version are you using? Can you send me your .config ? Thanks again, Isamar On Thu, 24 Jun 2004, Martin List-Petersen wrote: Is your kernel ACPI enabled ? The motherboard in the PE400SC is basically the Dimension 8300, which i use for my development box with 1 X101P, 1 TDM400P and two ISDN cards here at home and that works without problems. One thing to make sure with these boards is that ACPI is enabled, since they are ACPI only. Kind regards, Martin List-Petersen On Thu, 2004-06-24 at 02:58, Isamar Maia wrote: I have a Dell PowerEdge 400SC with a X100P and a TDM01b. The board works wonderfully in another machine but in this brand new one, it just get in nuts. The problem is: 1) Zaptel recognizes it perfectly 2) No IRQ conflicts, two-wire new cable. 3) Asterisk starts up and listen the ring and answer the cal 4) RIght after answering the call, it's dropped. 5) The following calls, even with asterisk off, the driver(???) answers the call and hang it up. With the * running, it doesn't even get any ring, and the call is answered and dropped right away. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Video/H323/SIP
Nakano San, Have you tried to make * only to route the connection and they just talk point-to-point without * bridging? Isamar On Thu, 24 Jun 2004, Masakazu Nakano wrote: I tryed it. but callee cannot answering with video in SIP. # surely videosupport=yes in sip.conf H.323 is works well but I think stilln't support over * yet. mack_jpn. On Thu, 24 Jun 2004 14:03:10 +0200 Michael Devenijn [EMAIL PROTECTED] wrote: I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based solution to establish a video connection with * and sip protocol? - Original Message - Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ 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 DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] How to force G729
Sorry guys... These are all great tips, but also this doesn't work: the gateway is not under my control, it is actually a real phone switch, which isn't owned by us. Unfortunately I can't tell them to add a second IP ... :-) As I could understand so far, you wanna do G729 passthu from a SIP connection and the PSTN running in the asterisk. What I am asking to myself now if it is technically possible without transcoding or having G729 licenses. Isamar ___ 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] Video/H323/SIP
It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Isamar ___ 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] SIP Registration problem
Hi Folks, I'm having problem with GS registering in Asterisk. My setup is the following: [1755] type=friend incominglimit=10 qualify=no nat=yes insecure=no secret=X dtmfmode=rfc2833 username=1755 host=dynamic canreinvite=no defaultip=192.168.0.1 context=sip-incoming I have dozens of phones running the above configuration. All GS-BT101. The problem is that some of those phones, in the other side of the world, only register themselves during the boot and become unreachable after some minutes not re-registering themselves periodically what would be the right process. Registration time is 5 min. Firmware version 1.5.0.0 Asterisk version is 7.2 Anyone has any clue? Isamar ___ 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] Active extensions via Web
Hi, There is already any CGI script to show the active online extensions through the web? Thanks, Isamar Maia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk behind Iptables: What's the magic?
I tried some combinations of setup seen in some postings and didn't get success on this yet. I have grandstream phones outside the network trying to call an * server inside my network through NAT/Iptables. The problem that I'm facing is one-way audio. Any suggestion? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P: Sharing IRQS?
This isn't directly related to your question, but I just recently committed some more documentation on IRQ sharing to the http://www.asteriskdocs.org book. Feel free to check it out, it may be of benefit. Finally, TDM400P also has IRQ issues, right? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 fallback
AFAIK.. it shows up a crazy error... The G.729 crying for more licenses... Isamar On Tue, 1 Jun 2004, Mike Heininger wrote: Hi, if the G.729 codec runs out of licenses does * fallback to another codec? TIA, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P: Sharing IRQS?
I had a little nightmare playing with X100Ps and IRQs and I decided to buy TDMP400P/FXO and FXS. The question is, can I put multiple boards in the same motherboard without worrying about IRQS? TDM400P shares IRQs with other boards? Isamar ___ 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] Portuguese Sounds
From where can I download the portuguese sounds? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linejack dialout
Yes. Give away your LJs to some university for research... They are not for business... and don't buy X100P. Buy TDM400P. It has the same price of a LJ and have 4 FXOs instead of only one. Isamar On Wed, 19 May 2004, Jer wrote: Dear all I read on the list back in 2003 that * does not support IXJ LineJACK dialout yet is this still the case? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?
I don't know what to tell you, other than to echo the statement that you'll probably be better served by installing a 4 FXO TDM400P card, even though that's gonna cost you another US$400. You might try asking here on the list if anybody wants to buy some X100P boards... Sullivan, Thanks a lot. I'm trying to find a motherboard with more IRQs until today. If I don't get it, I'm gonna order the TDM400P(4FXO), otherwise I would need to have two machines what'd be a big mess. Isamar ___ 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] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?
Hi folks, I'm trying to make an * PBX for a customer using 4 X100Ps and 1 TDM400p(4FXS). The problem I'm facing is to make one unique IRQ for each PCI slot/board since shared IRQs create all kind of weird noises and echos. Anybody got any workaround for that? Any recommended motherboard to accomplish that ? Currently, I'm playing with an ASUS A7V600. Thanks for any tip, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Isamar Maia wrote: | Hi folks, | | I'm trying to make an * PBX for a customer using 4 X100Ps | and 1 TDM400p(4FXS). | The problem I'm facing is to make one unique IRQ for each | PCI slot/board since shared IRQs create all kind of weird noises | and echos. | Anybody got any workaround for that? | Any recommended motherboard to accomplish that ? | Currently, I'm playing with an ASUS A7V600. Have you looked at possibly using the TDM400P with 4 FXO modules? Then you would only need to have 2 cards (currently) in the system and possibly have room for expansion in the future, if needed. That's an excellent idea, and maybe the unique way out. But, what do I do with all my X100Ps that I bought from Digium? Give them back and get my money back and buy a TDM400P(4FXO) ? :-) Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P and TDM400P non-USA Caller ID
How many of you are prepared to do this? Can you nominate your country needing this this feature! I can contribute financially with it also. Or saying better, I am crying for it for more than one year. I'm in Japan. NTT FSK and I have already all the documentation in english. I talked to Digium one time about that, but I've heard that it's not a priority. One year is passed. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing by Called interface
On Sat, 2004-05-08 at 10:52, Chris Wilson wrote: I want to run different lines directly to different extensions on two FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to extensions 102 Does anyone know of a way to do this? Yup! Check your trash folder. This was discussed on this list in the past 7 days. I didn't get this yet. The helicopter noise still sounds I'm changing the cables today to eliminate any interference possibility. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DSL vs X100P
I am trying to forward an inbound call to go out through another X101P and I get nothing but a noise like a helicopter sound... Inbound and outbound are ok if done separately. I already checked IRQs and they are fine. Updated the drivers and asterisk and they seem to be ok too. Turned on and off echo cancel. Both lines are coming from an ISDN line,channels A and B respectively. Should it be cable problem or another issue, in this case with ISDN lines? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSL vs X100P
I had a similar problem after a CVS update and had to set the rxgain to -2 to reduce the time the echo canceller kicked in... The problem is that my settings now only work well with rxgain=+15 txgain=+15 Setting rxgain to -10, the noise disappeared but I can hear only one side of the line. Isamar [EMAIL PROTECTED] Nagoya/Japan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] -- MARK --
On Mon, 19 Apr 2004, Michael Welter wrote: Every half hour I get -- MARK -- in the syslog. Is this normal behavior? This has nothing to be with asterisk, but with your linux installation. Yes, it is a normal behavior and it is harmless... It is just a half hour stamp to your syslog... I think it was because of MARK Spencer... burn him! :-) Isamar ___ 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] CTI
I found some messages in the history about * and CTI but nothing concrete explaining how to do that. I want to receive calls and pass their info to application running on a windows client machine. I am using Visual Basic 6.0 and .NET Where Can I find the TAPI 2.0 API like in http://www.mail-archive.com/[EMAIL PROTECTED]/msg30982.html ? or something like http://www.mail-archive.com/[EMAIL PROTECTED]/msg02388.html ? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group
Me too. Isamar On Mon, 29 Mar 2004, Paul Mahler wrote: Where and when is the rollout meeting? I'd love to attend. Thanks! Paul Paul Mahler [EMAIL PROTECTED] ___ 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] Zap to Zap
I have two X100Ps for testing. I want to receive a call from one and dial out through the other available. When I make a call, it try to dial to the channel in use. How to solve this? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P dial in/out to sip phones
Caller ID does not work in the UK, well not on my BT or Telewest line's. What I didn't understand yet about * + X100P with caller id not working in some countries is, it's a hardware or software limitation? Isamar ___ 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] Call priority
I am trying to make some call routing... I have the following rules in the same context: ; 1st rule exten = _1800.,1,Dial(SIP/..) exten = _1800.,2,Congestion ; 2nd rule exten = _1.,1,Dial(SIP/..) exten = _1.,2,Congestion The problem is that some 1800 calls are still going to the second rule. What is the best way to accomplish that? Thanks, Isamar Maia ___ 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] Grandstream / SIP - IAX2 / Voicepulse
Dear Folks, Trying to make calls from a GS behind NAT using SIP through my * server talking IAX2 to Voicepulse and no success. From GS to Zap/PSTN is ok and vice-versa. From ZAP to Voicepulse(IAX2) no problem... but.. not getting to connect SIP-IAX2 and the problem is not only with VoicePulse but with another provider as well in the same situation, GS(SIP)- * - IAX2 - ITSP -- Call accepted by 66.234.228.132 (format G729A) -- Format for call is G729A -- IAX2[voicepulse]/2 is busy -- Hungup 'IAX2[voicepulse]/2' == Everyone is busy at this time -- Executing Congestion(SIP/1604-4f72, ) in new stack What should it be? Thanks in advance, Isamar Maia ___ 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] call2ua Dial parameter
Hi, I am trying to connect through call2ua with no success. It seems to be authentication problem. Anyone could inform to me how is the Dial/H323 parameter to authenticate with them and dial? Thanks, Isamar ___ 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] 800 numbers / Skinny - IAX2
Hi Folks, I'm trying to route IAX2 calls to 800 numbers from a Skinny channel and the log says: -- IAX2[69.73.19.178:4569]/4 stopped sounds -- Asked to indicate 'Stop tone' condition on channel Skinny/[EMAIL PROTECTED] And no audio happens. It's working for Zap and SIP channels, though. Tried to google about that but didn't find anything. Any thoughts? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing 800 numbers with VOIP
Use fwd.pulver.com On Mon, 9 Feb 2004, Matt Lawson wrote: Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800 numbers. Are 800 numbers treated differently somehow? Or is there a business reason for disallowing them? It makes the ringing sound but never connects. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P
I only have 1 server which is dual p3 1ghz. Its mobo only has 2 64 bits pci. :( Not good. Throw it away in my house's trash can and I send you 4 Pentium 2... more details in pvt. :-) Isamar ___ 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] Prepaid Calling Card
I am planning to sell prepaid calling cards to my service. The system is already working but I wanna print a good quality prepaid calling cards for it. Anyone would recommend me a good and cheap pre-paid card printing company anywhere in the world? Thanks in advance, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
On Thu, 5 Feb 2004, Clif Jones wrote: No they do not. I am managing an installation running 7960 SIP release 6.0 and the phones are on about 4 different subnets. Half of these are on remote VPN connections at people's homes. Currently, The Cisco 7960 SCCP can hear me but I cannot hear him. Both are in Public IP address without any firewall. What the problem should be? I tried different codecs and no change. In the asterisk side I'm using a X100P. Isamar ___ 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] Cisco 7960 : No one-way audio
One friend with Cisco 7960 with public IP address connect to my * box and I called me to my home phone through a X100P. He can hear me clearly and I cannot hear him. I thought the problem could be a NAT in the middle.. but there is no NAT. Any thoughts? Isamar ___ 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] Quintum Tenor a800 and *
Dear All, Anybody is using successfuly Quintum Tenor a800 with Asterisk? If yes, if possible, I would like to know ping response in ms between the two points and codec used. Thanks, Isamar Maia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip phones
not to mention, fortune cookies are included! :) Hey Chinaman... I was wondering if the following SIP phone is just a Grandstream's OEM or just a japanese copy... http://sipphone.livedoor.com/ What do you think? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip phones
I am in Japan and I was just going around in some shops in the web... Isamar I found a site somewhere that referenced the livedoor sipphone to: LivedoorSIP phone terminal development original page Http: //www.grandstream.com/y-product.htm The manual it is Which means Livedoor sip phone is infact the grandstream phone. Not even a clone, it's the same thing. :) How did you come across that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Isamar Maia Sent: Wednesday, February 04, 2004 4:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voip phones not to mention, fortune cookies are included! :) Hey Chinaman... I was wondering if the following SIP phone is just a Grandstream's OEM or just a japanese copy... http://sipphone.livedoor.com/ What do you think? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P limit per PC
I know that it was commented here already but how many X100Ps I can plug per PC? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has Nufone gone belly-up
Which is the best way to contact Nuphone.? On Sun, 25 Jan 2004, Jeremy McNamara wrote: John Baker wrote: I tried a couple times to talk to them about service. How much it costs, how it works, etc. Just common stuff you might find on a website. I left a message and nobody returned my call; I went with voicepulse instead. No messages were ever received from you, thus we never called you back. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem registering FWD
NOTICE[245776]: File chan_sip.c, Line 2837 (sip_reg_timeout): Registration for '[EMAIL PROTECTED]' timed out, trying again Does anyone else have any problems with FWD? Terence This is password problem. I have suffered this problem before. Check the password and try again. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing list growth
I have my doubts about breaking the list... Another day, for ex., I posted a mail in asterisk-devel and I didn't get no answer. Some days after, I posted the same mail in the main list... and bm! I got my reply. I still think we need something more fine grained. I think we can add the asterisk-biz list, and eventually something akin to a newbie list, but need a more appropriate name, IMHO. Mark On Fri, 9 Jan 2004, Panny Malialis wrote: Just a suggestion, Could Digium make lists.digium.com accessible like a news server (NNTP) somehow ? Thanks Panny Malialis Hotlinks Internet Services ___ 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] I wanna buy a new X100P
I'm trying to buy a new X100P but http://shop.store.yahoo.com/bsdmall/wisifxoin.html is failing to check the order Anybody knows any other way to purchase it? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD / Timed out
Hi Folks, I can ping fwd.pulver.com with no problem but not getting to register with them... NOTICE[4101]: File chan_sip.c, Line 2476 (sip_reg_timeout): Registration for '[EMAIL PROTECTED]' timed out, trying again Anybody knows how to fix that? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_iax2.c Warnings
Hi Folks, I'm setting up my Iaxtel connection now and I'm getting some annoying warnings What means: WARNING[7176]: File chan_iax2.c, Line 436 (iax_error_output): Ignoring unknown information element 'Unknown IE' (31) of length 4 ? And how can I fix it? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallWaiting CallerID
Hi all, In order to get the CallerID from PSTN (X100P) I have modified callerid.c file like that: callerid.c [line 256] from: case 3: /* Number (for Zebble) */ to /*case 3: Number (for Zebble) */ Without this modification my own number was displayed as the inoming call CallerID. Now I want to go further. I have activated CallWaiting support on the POTS line. When someone calls and I am in another PSTN call, the displayed CallerID is asterisk. How can callerid.c file be changed in order to correctly display CallerID information during CallWaitiing? One question that I would add to yours is: the caller id functionality with X100P is completely software based or there is some hardware based feature working to make it possible? I'm still working with NTT Japanese functionality described in english at http://www.ntt-east.co.jp/gisanshi/analog/edit5e.pdf Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Double 3's Problem - H323 . Very weird
Hi Folks, I have a X100P with Asterisk running connection to a non-asterisk device in the other side. It was working perfectly with H323(chan_h323)+ G.729 in the last weeks. Suddenly, I am getting double 3's in the other side's POTS. Any number is not repeated, only the 3 is being repeated. I guess I'm sending it correctly and no change was done in the dial plans recently. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Party in Paris
Mark - I am more than happy to put together whatever you need done. Obviously could do with knowing how many people, what time of day, do you want to eat - drink or just somewhere quiet. Depending how we take this forward can supply full contact details mobile, personal email address. I went to Paris 3 years ago in the winter and it's really cold... A good night club that I found and liked was one named Terra Samba in Bastille neighborhood pretty cool. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)
I would like to hear from anyone else that has real world experiences with both chan_h323 and asterisk-oh323. For 6 months, I didn't know what was a perfect connection using both except using G711 with oh323. Few weeks ago, a big mind from Australia solved the problem with 1 or 2 lines of code, and now I'm using chan_h323 almost perfectly. I think there is some setup adjustments to be done by myself but that's not the case. The important now is that I'm happy with chan_h323 and the things seem to be easy having that as a channel module inside the dist. I just don't understand why the both projects merge having more synergy making the things happen faster... I'm pretty sure that there is no technical explanation for that :-) Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
