[asterisk-users] Registering Asterisk to a SIP Provider
Hey, It's late out here and I'm trying to setup a personal Asterisk server and have it register to a SIP Provider. Whilst the account works perfectly find from a softphone I keep getting a This account is not valid IVR. The provider appears to be running PortaSIP. Anyone with suggestions? -- Regards, Sahil Gupta Director Tigercom Pte. Limited 998 Toa Payoh North #07-22/23 Singapore 318993 -- _ -- 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
[asterisk-users] Dedicated Servers
Hi,I am looking for a reliable provider that can provide 3 dedicated linux servers asap. Unfortunately, the provider I have used for YEARS has become way too slack in recent times and we have to move on. Cheers, Sahil ___ -- 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] Off Topic: 8x FXO Gateway
Hi, I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist -- Regards, Sahil Gupta Corporate Advisor TigerCom Pte. Limited 296 River Valley Road Singapore 238337 ___ -- 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] Hylafax
Hi, We seem to be having some teething issues with a new Hylafax - happy to pay someone to complete the installation. Please contact offlist. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 ___ --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] Alcatel - Asterisk setup
Hi, You need to enable overlapdial. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 On Tue, 29 May 2007, Carlos Hernandez wrote: Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into Alcatel) If / when we manage to get anything from Alcatel, we get just the first digit of the number the user is intending to call.. Asterisk expects the whole number at once, so it fails.. Most of the time we get nothing at all from Alcatel, we think something is missing, so Alcatel sees the link is down. Please let me know if you have done this type of work before. We are not wanting to involve the Alcatel people, unless really required. Is there any special way to set up zaptel/zapata so Alcatel detects the PRI to be operational? Is there any special way to receive the calls once the PRI is up? Right now asterisk is set with: pri_net Any information or hints will be greatly appreciated Thank you, Carlos NZ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Could two Asterisk servers connect through VPN
Hi, Yes they can - relatively straight forward. Regards, Sahil Gupta VoiceValley On Mon, 7 May 2007, Tielin Xu wrote: Hi list: Has anyone done to set up two servers in different remote offices through VPN in order to get the VoIP communication? Thanks for your information. Tielin Xu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SpanDSP (RxFax)
Hi, We had an install working quite well of SpanDSP on our machine until recently where it has began spitting out an error stating unable to translate from unknown to unknown. Any ideas ? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Hangups on PRI Interface
Hi, I seem to be having an issue with a PRI at present whereby the call works fine for 90% of the users however, when a customer begins dialling DTMF tones over the channel with the ASTCC application - the call seems to disconnect from the PRI Interface end: -- Channel 0/5, span 1 got hangup request -- Hungup 'IAX2/voicevalley-7' == Spawn extension (context, 099092428, 1) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' I have upgraded to the latest versions and have also ensured that busydetect and callprogress are turned off. Any ideas? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Hangups on PRI Interface
Hi, I seem to be having an issue with a PRI at present whereby the call works fine for 90% of the users however, when a customer begins dialling DTMF tones over the channel with the ASTCC application - the call seems to disconnect from the PRI Interface end: -- Channel 0/5, span 1 got hangup request -- Hungup 'IAX2/voicevalley-7' == Spawn extension (context, 099092428, 1) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' I have upgraded to the latest versions and have also ensured that busydetect and callprogress are turned off. Any ideas? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic: Hardware Required
Hi, Apologies for the off-topic post, is there anybody in NYC with a bunch of video cards lying around that I might be able to get picked up this evening or early tomorrow ? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PABX Setup
Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PABX Setup
Thanks mate. All going well. Regards, Sahil Gupta VoiceValley On Tue, 6 Jun 2006, Boris Bakchiev wrote: Samsung PABX? Its TEPRI probably configured in overlap mode so you need to configure asterisk span that is connected to PABX to overlap mode as well. When user selects the outside line in overlap mode PABX connects to asterisk and then sends the digits to it as the user presses the key's. If overlap mode is not configured in asterisk switch is not started by asterisk and it just thinks that empty dial string was sent to it. Just use: overlapdial=yes in your zapata.conf Make sure you have exten = s,1,Busy() exten = s,2,Hangup in your 'samsungincoming' context so that users get a busy signal when they didn't enter any digits in allotted time otherwise you'll get a hanging channel in Samsung. We use that setup with OfficeServ 500 and it works really well. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Tuesday, 6 June 2006 21:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PABX Setup Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [Asterisk-Users] Prices of g729 codec
Hi, I couldn't quite understand what was so wrong if someone was moving a bit of hardware around and requested key changes. After all, the keys have been paid for and the registered person was requesting for the keys to be reset. It was a while back... All good otherwise. Regards, Sahil Gupta VoiceValley On Mon, 5 Jun 2006, Jon Lewis wrote: On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - Sahil Gupta [EMAIL PROTECTED] wrote: We recently had around 60-80 licenses become useless because Digium refused to renew the keys on that. That was a bit of money kissed goodbye. Unless you had been clearly abusing the key licensing system, our support department will never refuse to enable a new registration on your license key(s). There is no 'renew the keys', though, since they don't expire. I hope that's the actual official policy now. There seems to have been some internal conflict or communications failure at Digium a few months ago as to whether or how many times a g729 license key can be reset. As a service provider (you could call us an Asterisk ASP), we regularly build host systems for customers, retire/upgrade systems, swap out hardware, add interfaces, etc. which causes problems with the g729 licensing. In one attempt a few months ago to get a license reset, I was initially told it was now policy that Digium would only reset the registration count once, and after that, you were SOL (or forced to play MAC address changing games or as someone else posted, try hacking around the license key code). In that particular case, the customer's server had suffered a 2 disk RAID failure, and to get them back online, I moved them to a lower end system (what was readily available) while we waited for parts to get their dual xeon server back online. Both motherboards had built-in dual ethernets. IMO, locking the licensing to a piece of system thats often built-in, has been very annoying. I think I'd be happier if it was locked to some sort of dongle (parallel, or more likely today, USB). At least that way, we could easily move the key anytime we needed to. It would be a bit of a pain any time a system needed to quickly be transfered to hardware already at another location. The TRX idea sounds appealing, but I wonder how they'll handle servers that don't have internet access. Not all VOIP servers are on the internet. I've actually wondered if we could legally use Intel's code in cases where we have licenses bought from Digium, but they're not re-registerable because Digium wouldn't reset the use count. -- Jon Lewis | I route Senior Network Engineer | therefore you are Atlantic Net| _ http://www.lewis.org/~jlewis/pgp for PGP public key_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Developer
Hi, I need a few things modified on the current version of astcc. If there is someone competent, please contact me off-list. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
We recently had around 60-80 licenses become useless because Digium refused to renew the keys on that. That was a bit of money kissed goodbye. Regards, Sahil Gupta VoiceValley On Sat, 3 Jun 2006, Chris Mason (Lists) wrote: I have no problem with paying Digium the $10 for G729 licenses, everyone has to make money. It's the administration of the licenses that sucks. I experiment with different hardware a lot, and make up demo machines to install for customers with available hardware. I have to put G729 licenses on them, usually $100 each time, and when I install the real hardware for the client, I can't transfer the licenses. If I scrap that machine or change the interfaces, that's a $100 loss. I believe when you buy a number of licenses, that should determine how many instances you can use, regardless of how you want to deploy them. In short, the method of enforcement is poor and leads to resentment from customers. Surely Digium can construct a better system? -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contract Work : On-site NYC
Hi, We require a technical person to do some on-site installation work for us in New York, must be proficient with Linux and Cisco. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canada Termination
Hi, We will have a DS3 of capacity available directly into Canada with a Tier-1 Carrier there - we will almost certainly be able to thrash your existing rates for high volume traffic. If anybody is keen on routes into Canada, please contact me off-list. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Colocation Denmark
Hi there, Is there anybody on the list that offers or can put me in touch with somebody that offers quality colocation services in Denmark? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Work available - India
Hi there, If there is anybody on-list looking for VoIP related work in India, please contact me off=list with your details. Positions are of a full-time nature. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Part-Time work available
Hi, I'm looking for someone to do time-to-time mantainence on some of our machines going up in New York. The person *MUST* be stationed in New York. Areas of expertise required: - Proficiency in Linux: Slackware, Fedora - Proficiency on Cisco Routers If anybody is interested, please contact me off-list. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco AS5350
Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP in India
http://www.dov.gov.in Regards, Sahil Gupta VoiceValley On Wed, 25 Jan 2006, Code Lover wrote: Hi all, I would like to set an VoIP Gateway in India. Could any one tell me, is VoIP is legal in India? How I can obtain the license to start my VoIP gateway? -- Thank You, Code Lover ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP in India
Hi, This is not entirely correct, though it is a slightly costly excercise and India is making progress on it is definately possible. If you hold an ILD license in India you can terminate whatever you like and there will be no objections held. Not having the license and running grey routes (whilst quite common) leaves you wide open for prosecution. The cost of the ILD license has significantly reduced in India and has fallen from a previous fee (paid in the form of a bank guarantee) of 25 Crores to 2.5 Crores - so yes PSTN termination is possible if you have 2.5 crores plus capital outlay in your back pocket. There are at present only 4 ILD operators in India (Airtel, BSNL, VSNL, Reliance) - this means all other carriers Hutch/Essar for e.g. must buy their outbound termination for their mobile network off an ILD operator. That said, if you are running a 100% VoIP Network, you need not purchase off an ILD operator but are still required to purchase off a local ITSP license holder (there are a dozen of them in each city). Hope that clarifies the fog. Regards, Sahil Gupta VoiceValley On Thu, 26 Jan 2006, Vamsi Pottangi wrote: Nope, convergence with public phone network is not yet legalized in India. You could use VoIP for your local network. This is how call centers in India work. They use VoIP to connect to outside world but not to India PSTN. Hope this is clear. ~Vamsi On 1/26/06, Code Lover [EMAIL PROTECTED] wrote: Hi all, I would like to set an VoIP Gateway in India. Could any one tell me, is VoIP is legal in India? How I can obtain the license to start my VoIP gateway? -- Thank You, Code Lover ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] (Fwd) hi there
Hi, Not very reliable for commercial setups, they do have issues hanging up ports etc. Quintum over Antek any day. Regards, Sahil Gupta VoiceValley On Mon, 2 Jan 2006, Rehan AllahWala wrote: www.antek.com.tw Had 4 port fxo, for around 200 to 250$ They are OEM, and can change things if u need. I tested it breifly in there office last year in Computex 2005 You can contact [EMAIL PROTECTED] for wholesale. Rehan On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? I bet it has an fcc id which can be looked up at fcc.gov. If you get the first 3 letters it tells you who the vendor is. Maybe a ruse about not believing that it has all those compliance certifications and you want to guarantee the FCC certification for use in the US ... I would google for the name on the sticker, which is 'fxo-04'. This returns people talking about teh Asotel(Dinamyx) fxo-04. There is also a 'stargate fxo-04'. On and on ... If I had to guess I would say it looks like: http://www.chinanetphone.com/newchanpin/fxo-04.asp or http://www.repotec.com/voip/RP_FXO02A.htm My guess is that you should be able to find out more on your own :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group --- End of forwarded message --- --- End of forwarded message --- Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check how many G729 codec license installed
Right :) Regards, Sahil Gupta VoiceValley On Sun, 13 Nov 2005, Angelito Manansala wrote: *CLI show g729 No such command 'show g729' (type 'help' for help) this means i have no g729 codec installed, right? On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote: That's easy... Just go into asterisk cli and type show g729 It will tell you how many are active and how many you have in total Regards Zafer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelito Manansala Sent: Sunday, 13 November 2005 10:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to check how many G729 codec license installed Guys, is the any CLI commands or info files where you can check how many g729 codec license installed. Regards, Lito ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] codecs
You simply need to have g729/g723 codecs. Asterisk comes with gsm by default. Regards, Sahil Gupta VoiceValley On Wed, 9 Nov 2005, Olivier Taylor wrote: Right, I must suppose I need gsm codec to hear gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Angelito Manansala Envoyé : mercredi 9 novembre 2005 12:28 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] codecs i think gsm you mention is gsm sound files not gsm codecs. On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote: Hi all, We use asterisk as a local pbx and we connect to a pstn/sip provider for calls to pstn. Since the messages on asterisk are on gsm format, we need gsm, but to call pstn, we need g729 or g723. How can we enable both codecs to be able to call pstn and hearing voicemail messages for example? Any idea is welcome. Olivier ___ --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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] didgium card in india
Such hardware I believe incurs a stock standard duty of 35% plus some other charges. All up, AFAIK it will cost you $2300USD to import the card (based on the $1495 price for a 4 E1 card). You can try guys like Drishti in Delhi, they can help out. Regards, Sahil Gupta VoiceValley On Sat, 24 Sep 2005, Capt MS wrote: where can i buy the digium or any other card to work with asterisk in india and what is the cost like __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - Dying Signal 11
Hi, Asterisk keeps dying reporting error signal 11. There is no segmentation fault etc and full logging reports nothing with respect to reasons of why it restarts. Any ideas? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX
Hi Kapil, AFAIK, there are no such PDF's that exist unless someone has really spent time compiling such information, which will be great to see. However, if you check out www.voip-info.org, its a complete mine of useful information regarding doing what you wish to. Regards, Sahil Gupta VoiceValley On Wed, 21 Sep 2005, kapil dhawan wrote: Hi List I am very new to Asterisk but have been alloted a job to replace my traditional PBX with it. Kindly provide me some useful info (PDF's etc) to setup Asterisk with FXO and FXS both. I have to cater some 60 users with 10 simultaneous calls. Regards _ Biography of Shah Rukh. His profile, awards, films. http://server1.msn.co.in/Profile/shahrukh.asp Find more here! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internet connection between Africa and Europe
FlagTel offer dedicated circuits between Egypt and Europe, if that helps... Regards, Sahil Gupta VoiceValley On Thu, 15 Sep 2005, [ISO-8859-1] Stéphane LAVRI wrote: Hi I'm looking for a company who can provide me an Internet connection between africa and Europe. Plesa If someone can give me some contact name or company dont hesitate to send me a mail at [EMAIL PROTECTED] Best regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oh323 and Asterisk with MERA
Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) -- H323 -- Asterisk -- IAX -- Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it again. This is a problem that crops up randomly and goes away randomly as well. A permanent solution would make life easy Regards, Sahil Gupta VoiceValley ___ --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] Call drops
Hi, We are facing an issue with ALL calls simply dropping during peak times (this is happening upto 10-13x an hour) on certain gear: We have a setup like this: Client --- SIP --- Asterisk --- IAX --- Asterisk --- ISDN --- Provider Any ideas? Regards, Sahil Gupta VoiceValley ___ --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] Balancing traffic between two routes
Hi,
We are currently running our own equipment to break calls out in a
location I need to balance the calls out between two sites so that one
site doesn't keep getting hit again and again.
So currently we have something like this:
exten = _1.,1,Dial(IAX2/pop1/${EXTEN})
exten = _1.,2,Dial(IAX2/pop2/${EXTEN})
But the above.. would hammer pop1 any tips ?
Regards,
Sahil Gupta
VoiceValley
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RE: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations
Hi, Whilst the talks are on regarding DS3's, what is the maximum number of simultaneous channels Asterisk should be able to push through in pure pass-through mode? Regards, Sahil Gupta VoiceValley On Wed, 13 Jul 2005, Brian C. Fertig wrote: Trust me dude.. You don't want a lucent TNT. If your going all out for an DS3 and you don't want to multiplex it then you will need something that will take a DS3 which I don't believe TNT's do. Purchase an AS5400HPX they will and work very well. Set yourself up with some dialpeers etc and your good to go. Trust me. I have done it. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Sent: Wednesday, July 13, 2005 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations At 10:06 AM 7/13/2005, you wrote: Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM - VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features are needed, I think it can handle the routing. If not, I can whip up a SER box. We currently have a Cisco 7206VXR (1 voice resource) and a Cisco AS5300 (120 voice resources). The DS3 will also have SS7 signaling on it. Recommendations/comments/concerns/rants are graciously welcomed. Lucent TNT Thanks, Matthew ___ 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ 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] Systems Admin; Telecom Newbie - What do I need?
Why not look at getting a provider that can port your numbers to their network and buying the DID's off them over VoIP? Regards, Sahil Gupta VoiceValley On Wed, 13 Jul 2005, Ed Pastore wrote: Thanks for all the great replies. I guess I over-asked my question (since so many kept popping up). For now, what I really need to determine is what I need to budget for a full implementation. Unfortunately, I don't have time now to do testing and analysis... I just need to get my budget submitted. So I'm trying to figure out what all I'll need to buy and budget for. Obviously this is pretty hard, since I understand so little about telecom. So that said... Can anyone help me in determining what all I will need? The only thing I really need is one ballpark figure for a grand total cash outlay. However, it it is too low, I may be hosed. If it is too big, the project may be cut out of the budget. So I'd like to get within, say $5K of the actual expected cost. The items I had identified in my original post were: - A server, running Debian Linux or OS X (our preferred operating systems here) - A good network. We're on switched 100 Base-T, but will move to gigabit next year. - A T1 or some dedicated channels of a T1 - Gateway PCI cards or devices (in the case of OS X, only devices I guess) - VOIP phones or phone software (I'd like to use software and USB handsets) Are there more things I need? Or does someone have a rough estimate of what it costs to implement an Asterisk system in a small business? We have about 50 users and currently have something like 20 POTS lines coming into our PBX. Thanks again. ___ 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] Using G729 in pass through mode
Hi, If you are terminating the call from/to a T1/E1 card or modifying the call in anyway e.g. playing IVR prompts not just voice in - voice out, you will require the codec. Regards, Sahil Gupta VoiceValley On Thu, 7 Jul 2005, Obelix wrote: Is it possible to use G729 on asterisk without the license? It is to connect devices which use the codec to termination providers in a phone card application. Will decoding the DTMF tones from the caller require G729 processing? This message was sent using IMP, the Internet Messaging Program. ___ 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] res_config_mysql.so in CVS asterisk-addons broken?
Hi, I spent quite a few days with this and in the end I find that the 1.07 release is by far the most stable. I had a lot of trouble with the CVS release. Ofcourse, thats just in my case, what do the others feel on this? Regards, Sahil Gupta VoiceValley On Thu, 7 Jul 2005, Christoph wrote: Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph ___ 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] g.729 codec -- open source?
Check out http://www.readytechnology.co.uk/open/g729/ Regards, Sahil Gupta VoiceValley On Wed, 6 Jul 2005, Juraj Bednar wrote: Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar. ___ 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] cdr_mysql
Hi, Something seamless has become rather painful lately: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:162:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:162: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:162: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:162: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 A search on google says to use an older release, done that, no help.. any ideas guys? Regards, Sahil Gupta VoiceValley ___ 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] Colocation/Telehousing
Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley ___ 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] Error with app_addon_sql_mysql.c
Hi People! Having interesting issues with app_addon_sql_mysql.c: [EMAIL PROTECTED]:/usr/src/asterisk-addons# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 This is a installation of Slackware 10.1 with Mysql 4.1.12 (source). Any ideas? Regards, Sahil Gupta VoiceValley On Fri, 1 Jul 2005, Brian West wrote: You could have just done ln -s asterisk-1.0.9 asterisk and it would have fixed that. It should by default do -I../asterisk /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote: Marcel van Kaam, Fonetica wrote: I had the same problem with installing addons. I checked out in the file cdr_addons_mysql.c what the location of the asterisk.h must be and changed the cdr_addons_mysql.c to the location of the asterisk.h file. After this it worked. Also to be sure do: locate asterisk.h to check or you have the file on your system. Marcel Yes, that worked. For the record, it had to be #include ../asterisk-1.0.9/asterisk.h -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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] Can't build cdr_addon_mysql.
Hmm.. I'm having this problem today: /usr/local/mysql/lib/mysql/libmysqlclient.a /usr/local/mysql/lib/mysql/libmysqlclient.la /usr/local/mysql/lib/mysql/libmysqlclient.so /usr/local/mysql/lib/mysql/libmysqlclient.so.14 /usr/local/mysql/lib/mysql/libmysqlclient.so.14.0.0 cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz-L/usr/local/mysql/lib /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 I've tried many many things to get it going but have failed (incl. reinstalling zlib)... any ideas? Regards, Sahil Gupta VoiceValley On Fri, 1 Jul 2005, Brian West wrote: You could have just done ln -s asterisk-1.0.9 asterisk and it would have fixed that. It should by default do -I../asterisk /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote: Marcel van Kaam, Fonetica wrote: I had the same problem with installing addons. I checked out in the file cdr_addons_mysql.c what the location of the asterisk.h must be and changed the cdr_addons_mysql.c to the location of the asterisk.h file. After this it worked. Also to be sure do: locate asterisk.h to check or you have the file on your system. Marcel Yes, that worked. For the record, it had to be #include ../asterisk-1.0.9/asterisk.h -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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
[Asterisk-Users] TE100P
Hi, I have a Gateway running in TE (terminal equipment mode as slave that I need to connect to my asterisk server using a TE100P card. Can anybody give a quick run up of how to run the TE100P's in Network Termination mode to have this working sucessfully? Cheers! Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
E-mail me off-list, we'll help out :-) Regards, Sahil Gupta VoiceValley On Sun, 26 Jun 2005, trixter http://www.0xdecafbad.com wrote: On Sun, 2005-06-26 at 23:29 +0200, Matt Riddell wrote: Andres wrote: So it looks like Livevoip went Bankrupt Sh1t. Looks like the Daily Asterisk News will need a new host. So, unless anyone can donate space for a custom php and mysql based site, it will be hosted in either New Zealand or Italy. Offers? sourceforge asterisk daily news documentation project? They have some bandwidth, file space, php and mysql are reported to work... Dunno if this will fit your goals though. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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 with Asterisk
Hi, We seem to be having an interesting issue with Asterisk whereby, it keeps routing calls coming in to the 'default' context regardless of what changes occur to h323.conf. SNIP [POP-A] type=user host=1.2.3.4 context=international /SNIP == Starting H323/ip$1.2.3.4:12914/16313 at default,12126599878,1 failed so falling back to exten 's' == Starting H323/ip$1.2.3.4:12914/16313 at default,s,1 still failed so falling back to context 'default' Any help would be appreciated... Regards, Sahil Gupta VoiceValley ___ 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: Re: Digium Website Update: Asterisk Business Edition
Take this off list please.. Regards, Sahil Gupta VoiceValley On Sun, 12 Jun 2005, Bob Goddard wrote: On Sunday 12 Jun 2005 16:10, trixter http://www.0xdecafbad.com wrote: On Sun, 2005-06-12 at 15:06 +0100, Bob Goddard wrote: On Sunday 12 Jun 2005 08:56, trixter http://www.0xdecafbad.com wrote: On Sat, 2005-06-11 at 13:47 -0700, Daryll Strauss wrote: On Sat, 2005-06-11 at 13:10 -0700, trixter http://www.0xdecafbad.com wrote: Look at 'big evil corporations' like apple. They did in a year with mach what the FSF/GNU wants to do with HURD and still cant (to quote stallman 'its really hard' while explaining why after 10 years HURD still doesnt exist). Apple was able to do this largely because they paid people to do it. That money had to come from somewhere. While apple did release darwin (the mach microkernel+ BSD components - but no mac components so largely not highly useful) under a license even the FSF claims is 'free'. Had it not been for the 'big evil corporations' that would not have existed at all. You're fairly off base with that paragraph. you're fairly stupid. I wasnt giving a history lesson I was talking about the fact that both apple and FSF tried to do the same thing. Apple did it in about a year (from the time mach actually became available to use the way it is) and FSF is stil trying and stallman is still whining that its really hard and that is why he cant get hurd done. You are the one who is fairly stupid. Apple took Mach, BSD and X and got them to talk to each other. The FSF, have taken Mach and are attempting to write another BSD. Thank you for repeating me and leaving out the fact that FSF *cant* geti t to work, to quote stallman on the problem its relaly hard and that is why they cant get it working. My whole point was that apple *did* it. You have so eloquently proven my point about your intelligence. For the last time, Apple took a ready written O/S in FreeBSD, the FSF are doing effectively a full rewrite of FreeBSD. A year my arse. Few people are working on Hurd where as with *BSD and Linux they are a cast of thousands. ___ 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] G711 ( alaw or ulaw ) pass-thru
Hi, Both of those are fully uncompressed codecs and free to use. Regards, Sahil Gupta VoiceValley On Fri, 10 Jun 2005, Edgardo Bermejo wrote: Hi, Its possible to make a pass-trhu conection with alaw or ulaw? Thanks -- Este mensaje ha sido analizado por C4I S.A. Mail Server en busca de virus y otros contenidos peligrosos, y se considera que está limpio. MailScanner agradece a transtec Computers por su apoyo. ___ 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] English vs American voice files
Like to share who can record NZ / Australian voices? Regards, Sahil Gupta VoiceValley On Wed, 8 Jun 2005, Mark Phillips wrote: I think you miss the point Andrew. She's not from NZ but from England. She speaks English. Says six and not sex etc. Mark Andrew Thrift wrote: I also have someone in New Zealand who has done some for our own Asterisk server. Mark Phillips wrote: I've found a woman whom is happy to help make English voice files! Ironic that she should be in New Zealand. More when I have the files. ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 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] Message Playback
Hi, I'd like to know how I can playback a pre-recorded message to a user using our system without answering the call. I want to do the above in the scenario where the user dials a number and the number has been dialled incorrectly. Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP--h323 conversion
This is relatively straight forward, you can either use Nufones Implementation or the OH323 package. Both work relatively well. However, I've had issues presenting a GateKeeper ID from Asterisk to carriers that authenticate based on that in the past. Regards, Sahil Gupta VoiceValley On Mon, 16 May 2005, Micko wrote: Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sipASTERISKh323-GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could someone help me with configuration of Asterisk? I installed [EMAIL PROTECTED] 1.0 and oh323 0.6.5 Thanks! ___ 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] voipjet anyone?
VoipJet are not too bad, little pricey though.. theres better around.. a matter of looking :-) Regards, Sahil Gupta VoiceValley On Fri, 13 May 2005, Andrew Latham wrote: Personally I thought that VOIPJET has the best service and documentation including simple up to date CDRs also. They do not offer incoming, at least not to me If you follow the instructions on their site it will work, if you are useing AAH then maybe you should look into editing the files by had. Checking the server ping time will help. Everyone with a nonroutable IP address will be surprised which ones are faster. On 5/13/05, JD [EMAIL PROTECTED] wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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] voipjet anyone?
International calls must be prefixed as 011 to voipjet. Regards, Sahil Gupta VoiceValley On Thu, 12 May 2005, JD wrote: Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m, [1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640 May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such context/extension May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received reject Outbound settings: notransfer=yes auth=md5 context=from-pstn host= 66.246.220.19 secret= md5hashstring type=friend ; also tried peer and user username=1234 Im using [EMAIL PROTECTED], but that shouldnt matter; people have this working or is it me? JD ___ 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: Who's happy with their voip service?
We've recently broken off with RNK, major service issues.. For some weird reason, during test time all carriers are great. When they get the money of you all of a sudden, the quality goes bad, the account manager is on holiday, the NOC is down and the list goes on... Regards, Sahil Gupta VoiceValley On Tue, 10 May 2005, Michael D Schelin wrote: Please Give me a call. I'm the owner of Shelltel. 626-814-2354 We're not the same cookie cutter VoIP carrier. Bryce W Nesbitt wrote: I started out happy as a clam with my new Broadvoice account and asterisk machine. About 10 days ago things began to change Who's happy with their voip service using asterisk? Where do you get reliable DIDs? The 'carrier partner' they speak of.. can you get the did directly from them? Are all the voip providers this flakey? I've tried 5 providers, and I can't say that I'm happy with any of them. I'm far to small to deal with 'carrier partners' directly (e.g. Level 3, XO or RNK). So I have to deal through resellers. And they all seem to be operating on shoestrings and duct tape. I'm OK with the awkward setup, confusing configuration, and (for Asterisk) all but useless documentation. But high latency, dropouts, unplanned outages, lack of clues, echos, all take the shine off things. With Asterisk I have very few tools to monitor connection quality, especially on the outbound leg of my calls. I at least want to know when it sucks, and have some control over parameters. I keep a POTS line at home. ___ 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] VoIP A-Z Carriers
Hi, If you recommend any good carriers, please let me know :-) Volume is no problem, prepayment is no issue. We require good quality routes with high ASR. Preferably on ulaw. Regards, Sahil Gupta VoiceValley ___ 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] Music on Hold
[EMAIL PROTECTED]:~# mpg123 -v High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! Regards, Sahil Gupta VoiceValley On Sat, 7 May 2005, Matt Riddell wrote: Sahil Gupta wrote: Assuming you have 0.59r (which you should), which codec is the call using? ulaw to the box. And do you definitely have 0.59r? Paste us the output you get when you type mpg123 -v -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Music on Hold
ulaw to the box. Regards, Sahil Gupta VoiceValley On Sat, 7 May 2005, Matt Riddell wrote: Sahil Gupta wrote: Hi, I've been trying to get music on hold going on one of our servers: Upon dialling extension 005, it plays: -- Executing WaitMusicOnHold(SIP/parssyd1-4dbe, 30) in new stack -- Started music on hold, class 'default', on SIP/parssyd1-4dbe However, no music in the background Assuming you have 0.59r (which you should), which codec is the call using? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Music on Hold
Hi, I've been trying to get music on hold going on one of our servers: Upon dialling extension 005, it plays: -- Executing WaitMusicOnHold(SIP/parssyd1-4dbe, 30) in new stack -- Started music on hold, class 'default', on SIP/parssyd1-4dbe However, no music in the background MPG123 is intalled.. musiconhold.conf shows: default = mp3:/var/lib/asterisk/mohmp3 The directory has?: [EMAIL PROTECTED]:~# ls -al /var/lib/asterisk/mohmp3 total 6589 drwxr-xr-x 2 root root 160 2005-04-21 10:25 ./ drwxr-xr-x 8 root root 216 2005-02-17 22:48 ../ -rw-r--r-- 1 root root 1939812 2005-04-21 10:25 fpm-calm-river.mp3 -rw-r--r-- 1 root root 2582496 2005-04-21 10:25 fpm-sunshine.mp3 -rw-r--r-- 1 root root 2217563 2005-04-21 10:25 fpm-world-mix.mp3 Any clues ? Seems like it actions things but isn't playing the mp3 files.. Regards, Sahil Gupta VoiceValley ___ 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] SPANDSP
Hi, I'm having troubles getting SPANDSP working with Asterisk (for faxes), on a search of google.. I came up with a few links but the rxfax and txfax modules wouldn't patch or compile into asterisk Any hints? Regards, Sahil Gupta VoiceValley ___ 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] Cisco 7940G
We have an interesting issue with relation to Confrence Calling on the 7940's. If a call is made to an internal number say 004 and then if we try and confrence a local number say 30184200 it dials it out via the international provider. However, just dialling 30184200 it dials it out via the correct provider. Regards, Sahil Gupta VoiceValley ___ 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] Digium Card Issues
Hi, I'm trying to configure a digium card here. Got everything working sweetly apart from the last bit.. dmesg shows: TE110P: Span configured for ESF/B8ZS Calling startup (flags is 4099) Registered tone zone 1 (Australia) whilst /etc/zaptel.conf has: span = 1,1,1,ccs,hdb3,crc4 bchan = 1-10 dchan = 16 defaultzone = au loadzone = au Any ideas? Regards, Sahil Gupta VoiceValley ___ 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] Cisco ATA Help
Hi, I have a Cisco ATA 186 that I bought on my recent overseas trip and its the I2 series which has higher impedance than the New Zealand standard 600ohm. Is there something I can do to make it listen to my DTMF tones? Regards, Sahil Gupta VoiceValley ___ 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] wcfxo problem
I'm having similar issues using an X100P Ambient Chipset Clone Card any ideas? Regards, Sahil Gupta VoiceValley On Mon, 11 Apr 2005, Dave Weis wrote: I've got a X100P in a compaq proliant 3000. My system stops taking calls and making calls. I had been getting the FXO PCI Master abort before updating, I am now running a cvs head checkout from a week or so ago. Now I still have the problem but get more error messages: Found a Wildcard FXO: Wildcard X101P Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) FXO PCI Master abort wcfxo: Out of space to write register 05 with 02 wcfxo: Out of space to write register 05 with 03 wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a Any solution? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as protocol conventer beetwen SIP and H.323
Hi, Try the OH323 implementation, we found it works better. Everyone has different experiences oviously.. Cheers, Sahil On Sat, 9 Apr 2005, Adam Rybak wrote: Hello, have successfully installed Asterisk 1.o with H.323 driver and made configuration: GW (Hardware)- GnuGK - Asterisk and i call into asterisk from the PSTN network and it's work fine, but i need to make conversion from SIP small gateways to H.323. I need to make configuration like that: (Normal Phones - SIP Gateways -) x many - Asterisk - GnuGK (H.323) - Gateway (H.323) SIP Gateway and H.323 Gateway supports g.729 - i need the g729 codec into Asterisk? Can i mark sip gateways that i will can see on h.323 gateway witch from SIP gateway it comes? Can you write sample configs for me? Im Asterisk newbie :) Regards, Adam Rybak ___ 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
