Re: [asterisk-users] Multi-registration ?
On 27/03/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Asterisk can handle multiple registrations for the same account. Both should ring when calls come in. No it can't - the latest registration 'wins'. To achieve simutaneous ringing of more than one phone (hard or soft), you need a SIP account for each and an entry in the dialplan which rings both. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?
On 16/03/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Here's a better question: why make everyone join another list when this one already works perfectly well? Have you experienced any difficulty asking or answering questions about Asterisk 1.4 here? -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
The same ones which, by total coincidence, you just advertised on asterisk-biz, perhaps? What are the chances of that? On 14/02/07, Sam Tam [EMAIL PROTECTED] wrote: Drop me an email I know some GSM Gateway that has a direct serial port for SMS Sam -Original Message- From: Jon Pounder [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 14, 2007 10:50 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending SMS from Asterisk Quoting Patrick [EMAIL PROTECTED]: On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote: Singer Wang wrote: by your .ca address I assume your in Canada.. both Telus and Rogers have a email-to-SMS gateway... Well, those are notoriously unreliable. I've had messages take hours to arrive when sent by the email-to-SMS gateway. I was kinda hoping for something more direct. Rogers prioritizes internal SMS messages over e-mailed ones. we do this with the vmobile.ca gateway (which is just using the actual bell cellular network), and only a handful of times in several years hasn't it been instant. I get the sms before my desktop mail reader has even picked up the same messages in most cases. What I'd like is some kind of SMSC -- or something that accomplishes the same thing. Maybe http://www.kannel.org/ provides some useful info. Regards, Patrick ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
On 14/02/07, Stephen Bosch [EMAIL PROTECTED] wrote: Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere (they are not interested in helping an individual customer and technical staff Tiers I and II have no idea what I am talking about). Are there SMS aggregators that I could use for sending messages to this particular phone over the Internet? There are plenty - I've used 2sms.com, clickatell.com and csoft.co.uk. bayhamsystems.com have a service tailored for Asterisk users. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway promotion from £69GBP
On 14/02/07, Dave Cotton [EMAIL PROTECTED] wrote: On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote: Hello All This month we would like to offer our GSM Gateway range for less to clear up some spaces. etc Perhaps, you could explain what is NON COMMERCIAL about your post. He does this all the time, and never bothers to respond to objections. Doesn't answer questions about how he mis-describes his products, either. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
On 14/02/07, Stephen Bosch [EMAIL PROTECTED] wrote: Peter Bowyer wrote: There are plenty - I've used 2sms.com, clickatell.com and csoft.co.uk. bayhamsystems.com have a service tailored for Asterisk users. These are all based in the UK. What if I'm in North America? Does it matter? What matters is whether they can deliver to your target users - check what countries + networks each one quotes in their footprint. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsterikNow vs Trixbox
Trixbox is easier to spell. Apparently. On 11/02/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Comments? People's opinions -- Thanks http://www.sqlhacks.com The SQL knowledge base ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/SIP gateway for Belgium and western Europe
On 19/01/07, Jan Dewerchin [EMAIL PROTECTED] wrote: Dear all, I'm not sure if this is the correct place to put it, but can I announce you the possibility of using a new, lost-cost trunk for Belgium and western Europe ? Maybe it's a shameless commercial plug, but have if you don't know it exists, how can you all benefit from this ? asterisk-biz is the correct place. This isn't. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate displayes on web interface status Offline
On 11/01/07, Markus Amann [EMAIL PROTECTED] wrote: Hi i have a trunk up and running with Asterisk and Sipgate.de and i can make call out but no call in but the Enddevice Status on the Sipgate Webpage says offline. Maybe somebody had the same problem in the past and can give me some hints ? You haven't registered with them. What does 'sip show registry' say? -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Subscription Bug?
On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. Yes, because the subscribe is against an extension, which is translated to a SIP (or other technology) user via the 'Hint' entry for that extension in the dialplan. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Subscription Bug?
Why? You're saying 'please update me on the status of extension '1234'' when there's no such extension. Where's it going to get the data from? Better to get a 404, know something's wrong and correct a typo than let it succeed and just not work. Peter On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: Asterisk, imho, should still accept the subscription request from user A. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Subscription Bug? On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. Yes, because the subscribe is against an extension, which is translated to a SIP (or other technology) user via the 'Hint' entry for that extension in the dialplan. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
On 24/12/06, Al Bochter [EMAIL PROTECTED] wrote: So I will try get you on my point of the message! It would appear to be 'unlimited doesn't mean unlimited'. Surely this doesn't come as a surprise to someone who has been in the industry as long as you claim to have been? Move on, nothing to see here. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
This is getting funnier by the minute. Way to go, Al. On 24/12/06, C F [EMAIL PROTECTED] wrote: I Find It Funny, So I Decided To Let Others Laugh As Well -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Sun, 24 Dec 2006 14:01:06 -0500 Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: [EMAIL PROTECTED] This is off the list C F, You are an ass Bret is a scammer you can take that to the bank from a PI. Sorry I never stated what I do for a living. Did I? I will be dealing with Bret. And 2007 is not going to be a good year for that scammer. So why are you hiding use a real email address. And a real name. Looks like you have an in with Bret Master of Cybercrimes May have to my homework on you to. What is you think? I really don't care if you if you trust me. Your reply is only a pop out trying to save your ass. Please stay on the POINT! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: Al, Nobody Cares About Your Problems With Bret. Most People Here Know And Trust Bret More Than They Do You. All You Have Done So Far Is Made A Fool Out Of Yourself. At This Point All I Can Think Of Is That If Bret Does Hold Some Of Your Money That It Is A Significant Amount And He Wont Ever Give It To You. Move On And Dont Make A Bigger Fool Out Of Yourself. Swallow Your Pride Its Not Fattening. For You I Can Say: Temper Is What Gets You Into Trouble Pride Is What Keeps You There. On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --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 Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 1:41:46 PM ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Oh no, the game's up - Al's found my IP address. Wait - no he hasn't - he's found an IP address that belongs to McAfee Security in Spain - with whom I have no connection at all. (Hint: whois ip address) Those PI classes really paid off, Al. Supposing you had managed to find out one of my IP addresses (which isn't really too hard, I have NIC handles at ARIN and RIPE, and hold addresses on behalf of more than one major organisation), what were you going to do with it? I'm done with this. I thought we were discussing VoIP provider scams? On 24/12/06, Al Bochter [EMAIL PROTECTED] wrote: Peter, This is off the list? it looks like ip: 62.189.112.129 Country GB: Britain AM I close? Anyways This is off my point! And should not be posted to the list. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Peter Bowyer wrote: This is getting funnier by the minute. Way to go, Al. On 24/12/06, C F [EMAIL PROTECTED] wrote: I Find It Funny, So I Decided To Let Others Laugh As Well -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Sun, 24 Dec 2006 14:01:06 -0500 Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: [EMAIL PROTECTED] This is off the list C F, You are an ass Bret is a scammer you can take that to the bank from a PI. Sorry I never stated what I do for a living. Did I? I will be dealing with Bret. And 2007 is not going to be a good year for that scammer. So why are you hiding use a real email address. And a real name. Looks like you have an in with Bret Master of Cybercrimes May have to my homework on you to. What is you think? I really don't care if you if you trust me. Your reply is only a pop out trying to save your ass. Please stay on the POINT! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: Al, Nobody Cares About Your Problems With Bret. Most People Here Know And Trust Bret More Than They Do You. All You Have Done So Far Is Made A Fool Out Of Yourself. At This Point All I Can Think Of Is That If Bret Does Hold Some Of Your Money That It Is A Significant Amount And He Wont Ever Give It To You. Move On And Dont Make A Bigger Fool Out Of Yourself. Swallow Your Pride Its Not Fattening. For You I Can Say: Temper Is What Gets You Into Trouble Pride Is What Keeps You There. On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --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 Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 1:41:46 PM ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
:-) On 24/12/06, Steve Totaro [EMAIL PROTECTED] wrote: Peter Bowyer wrote: Yeah but it's good sport yanking his chain :-) On 24/12/06, Steve Totaro [EMAIL PROTECTED] wrote: What a tool. Al Bochter wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialplans for Asterisk?
On 21/12/06, Doug [EMAIL PROTECTED] wrote: Does anyone know the maximum number of digits for an international phone number? Doing some searching, it looks like 16 numbers including the 011 is the maximum number, because 17 is just not found: OK:1234567890123456 http://www.google.com/search?q=011X Not OK:12345678901234567 http://www.google.com/search?q=011XX Why would you imagine that people in non-US countries would list their phone numbers on their websites in US International dialing format? Especially when more countries use '00' for their outbound international prefix than use '011'. As has already been mentioned recently, at least one country (Germany) has no hard limit on the length of a number - extra digits after the base number are delivered to the CPE for internal routing - kind-of self-administered DDI ranges. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Match a Numer - then continue with dialplan
On 20/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: Bzzt. In order to call SetVar, I have to match the extension dialled. When that happens, there is NO WAY to continue searching the dialplan after that point for another extension to match. Can you not use either Goto or the Local channel, maybe a combination, to restart the dialplan with your variable set? (Might need a _ or two on the variable name to get it to survive) Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing CALLERIDNUM on the fly
On 19/12/06, Doug Crompton [EMAIL PROTECTED] wrote:
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and this plays havoc with my redial if the 3 digit area
code matches a local 3 digit extension. All my outside calls are 10 digits
or 1+10 digits.
Doug
[from-pstn]
exten = s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
exten = s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
exten = s,3,set(__CALLERIDNUM=1${CALLERIDNUM}) if not add 1
exten = s,4,noop(${CALLERIDNUM}) and this still displays without
I tried no, one and two underscores with the CALLERIDNUM variable.
gonzales*CLI show function CALLERID
gonzales*CLI
-= Info about function 'CALLERID' =-
[Syntax]
CALLERID(datatype)
[Synopsis]
Gets or sets Caller*ID data on the channel.
[Description]
Gets or sets Caller*ID data on the channel. The allowable datatypes
are all, name, num, ANI, DNID, RDNIS.
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] ASterisk and SER
On 04/12/06, Arun Kumar [EMAIL PROTECTED] wrote: HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. Your dialplan. (Since you didn't get around to posting any configuration or log information, that's about as close as anyone's going to get to your problem). Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP GSM Gateways
Not very good at answering followups to your ads, are you, Sam? On 01/12/06, Peter Bowyer [EMAIL PROTECTED] wrote: On 30/11/06, Sam Tam [EMAIL PROTECTED] wrote: We do have @cough VoIP GSM Gateway for sell as well @ cough Try to search on ebay for gsm voip gateway and you will see some in there As far as I am concern it is cheaper than 2n. And if you are looking for multi ports then it will come off as RJ11 ports rather than voip and they are £100 per port with a max of 16 ports in 1 chassis. It's cheaper because it's not the same thing and only does half the job - what you sell is an analogue-GSM adapter. It needs an FXS port to interface with Asterisk, and isn't actually a VoIP GSM gateway at all. If you must plug it here, please be honest about what it is and what it's not. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP GSM Gateways
On 30/11/06, Sam Tam [EMAIL PROTECTED] wrote: We do have @cough VoIP GSM Gateway for sell as well @ cough Try to search on ebay for gsm voip gateway and you will see some in there As far as I am concern it is cheaper than 2n. And if you are looking for multi ports then it will come off as RJ11 ports rather than voip and they are £100 per port with a max of 16 ports in 1 chassis. It's cheaper because it's not the same thing and only does half the job - what you sell is an analogue-GSM adapter. It needs an FXS port to interface with Asterisk, and isn't actually a VoIP GSM gateway at all. If you must plug it here, please be honest about what it is and what it's not. Peter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk: SIP Gateway or Proxy
On 01/12/06, yusuf [EMAIL PROTECTED] wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Ports (1000 to 2000 works)
On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before So is there any standard ports Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] skype and SIP hardware for linux
It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. Peter On 05/11/06, Thufir [EMAIL PROTECTED] wrote: I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under either Skype or as a SIP phone? The information I have on the driver, skypemate, is a bit sketchy. According to A-Link, the phone complies with SIP, http://www.a-link.com/us_us/IPU1.html, but the details are sketchy. No information is provided as to the interface for configuring SIP. The user manual, http://support.a-link.com/phonemate/Manual/IPU1manual_for_Linux.pdf, details using Skype but not SIP. Any user experience with this phone? For instance, has anyone used it with gizmo project or free world dialup, or even Skype? thanks, Thufir ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk upgrade
On 16/10/06, Simone Ruffilli [EMAIL PROTECTED] wrote: at the moment (fortunately) i'm not experiencing any kind of particular problem, do you suggest me to upgrade asterisk? #1 sysadmin rule: If it's not broken, just don't fix it. Slightly older and wiser sysadmins consider the importance of staying with a supportable version of software, especially if it's open source. If there's a security-related bug found in your version, will it get patched, or will you have a forced upgrade several versions ahead on your hands in a hurry? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EMAIL PROTECTED] problems
Probably best change the login and password from the defaults now you've posted this - your admin interface is wide open On 09/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Alex...I do not have FreePBX. What I have is this: http://70.89.124.237/ Ed ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming sip line with INX (internationalnumber.com)
On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote: Hi, I can't get my INX line working for incoming (outgoing is working fine). When I dial this number from my home phone, asterisk sends the call straight to extension 101, for some reason it doens't read what my extensions.conf is saying. SIP.CONF register = number:[EMAIL PROTECTED]/101 ;number is a replacement for my line number Didn't you think that the '101' there might be a clue? Your register statement tells the provider to deliver the call to '101'. Replace that with a different number and something different will happen - perhaps the rest of your dialplan is expecting the call to come in with a destination which matches your DID - in which case, put the DID number there instead of the 101. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate
On 09/10/06, Joseph [EMAIL PROTECTED] wrote: I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me Asterisk 1.2 is not ready for PRIME TIME. And that new-fangled electricity will never catch on - lets stick with gas-lamps... -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate
On 09/10/06, Joseph [EMAIL PROTECTED] wrote: On Mon, 2006-10-09 at 08:59 +0100, Peter Bowyer wrote: On 09/10/06, Joseph [EMAIL PROTECTED] wrote: I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me Asterisk 1.2 is not ready for PRIME TIME. And that new-fangled electricity will never catch on - lets stick with gas-lamps... Very funny! Though, it seems to me that my crashes after today's upgrade to 1.2.12.1 are related to this bug: http://bugs.digium.com/view.php?id=7972 Fair enough - that's a bit different to 'Asterisk 1.2 is not ready for PRIME TIME' though, isn't it? There are plenty of stable 1.2 releases, all of which have many fewer bugs than your 1.0.x version. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming sip line with INX(internationalnumber.com)
On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote: Hi Peter, Thank you for your answer. I did: register = DID:[EMAIL PROTECTED]/DID exten = DID,1,... Now when I call the DID number It doesnt reach the Asterisk. sip show registry shows me the line is registered but when I dial out from my softphone (eyeBeam) I get the 500 error - disconnected and the message the person you are calling is unavailable. Please, what do you suggest me to do? Have you matched up the 'context= ' entry for your SIP provider in sip.conf with the right context in extensions.conf where the 'exten = DID' is? Do a sip debug and see what it's telling you about the call, post it here if it doesn't help. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk act as a proxy ?
On 06/10/06, ram [EMAIL PROTECTED] wrote: Hi can some one clarify does the aterisks act like a SER http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
On 05/09/06, Jay Moore [EMAIL PROTECTED] wrote: Perhaps if answering the simple things politely is too difficult for you, you'd be better off not answering at all. Someday, I hope, you'll find that 'simple' is a relative term. Perhaps if receiving accurate answers without biting off the hand of the person helping you is too difficult for you, you'd be better off paying for a support contract with some reputable organisation? That way you can do no work whatsoever yourself and enjoy never-ending handholding at $150 per incident. That may suit you better. Around peer-support lists, you tend to find an aversion to telling people things they could easily look up or find out for themselves in a few keystrokes. You'll also notice that I took the trouble not only to answer your question, but to come back and re-phrase my answer when I saw you hadn't understood my explanation. You got all that for free. Enjoy! Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Right, I guess I was wondering if it's possible to include a file without it being in a context. The goal I wanted to achieve was to have as few contexts in the main extensions.conf file as possible. Did you try it? It would take... perhaps 30 seconds? A minute if you're a slow typist... Yes, you can do this. #include is a literal text include, as the last poster said. -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Marco: Ah I see. There's a [general] context. I'm pretty new to this Asterisk stuff and I didn't realize there was a general context that you could do things like global includes. Thanks, I'll give it a shot when I'm back in the office on Tuesday. Peter: No need to be an ass about it, pal. Not all of us are as adept at this as you are. You've still not got it. #include is a general text include - can be used anywhere. Well, perhaps it has to be at the start of a line. Contexts, not even the [general] section which isn't actually a context, has any relevance. It will insert the contents of the included file as though it was in the main file, wherever you put it. You could put the whole of the sip.conf file in an #include'd file. The whole of one context. One and a half contexts. 2 lines out of the [general] section. And so on. All of which, to repeat, could be experienced with a small investment of your time. It really does pay to experiment with the simple things, you find your learning curve is so much flatter than if you ask questions in a vacuum. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Jobs Update
As a couple of people have pointed out already, unless we doing something wrong, there seem to be no jobs. Haven't you any comment on that, other than to post another announcement about how great it is now the employers have to pay? On 20/08/06, Matt Gibson [EMAIL PROTECTED] wrote: Hello All, After a brief summer vacation, the Asterisk Jobs staff have returned and are gearing up for an eventful fall season here in North America. Asterisk Jobs (www.asterisk-jobs.com) has removed the free access for new employers after a successful 4 month promotion. Asterisk Jobs will continue to function free for all employees or other freelancers searching for employment. Asterisk Jobs (www.asterisk-jobs.com) is always upgrading and changing the site. Look forward to more announcements in the near future. The next planned release is a complete revamp of the site to include tags and other fancy stuff that will make searching for that dream job involving open source telephony a reality - quicker, and easier! For more information or to start looking for Open source Asterisk VOIP employment head over to http://www.asterisk-jobs.com Thanks, Asterisk Jobs Staff http://www.asterisk-jobs.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Port Forwarding SIP rtp
If someone asked your for help finding their front door key, would your proposed solution be to leave the door unlocked? On 11/08/06, Rosli Sukri [EMAIL PROTECTED] wrote: just disable iptables - if use redhat/fedora #service iptables stop On 8/11/06, Siqhamo Sifo [EMAIL PROTECTED] wrote: I need help with SIP,RTP port forwarding , I can connect using SIP and make calls but there is no audio even though my kernel has sip support and I suspect that it has to do with iptables. Siqhamo Sifo NewLunar Technology Solutions 5th Floor SmartXchange 5 Walnut Road Durban http://www.newlunar.co.za ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
What do Teliax support say?
On 11/08/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi friends,
We have installed Asterisk in our organization. We registered with Teliax
and got our DID number. We are making calls to USA successfully through
Asterisk. We are making outgoing calls to US. But, we are unable to receive
incoming calls to our DID number. When I executed the sip show peers
command, it is showing that my Asterisk server is registered and displaying
Teliax IP address also. I checking by doing ping to voip-co1.teliax.com.
Pinging is also going fine.
Here I am giving the configuration files. Please tell me a solution.
SIP.CONF contents:
[general]
register = xyz.abc:[EMAIL PROTECTED]
[authentication]
auth = xyz.abc:[EMAIL PROTECTED]
[teliax]
context=default
type=friend
username=xyz.abc
user=xyz.abc
host=voip-co1.teliax.com
secret=xxx
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[105]
type=friend
username=105
secret=rani
callerid=Ranikumar
host=dynamic
context=leader
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all
EXTENSIONS.CONF contents:
[leader]
exten = 105,1,Dial(SIP/105,15)
exten = 105,2,Voicemail(u105)
exten = 105,3,Voicemail(b105)
exten = 105,4,Hangup
exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
[general]
exten = 3031234567, 1, Answer()
exten = 3031234567, 2, Dial(SIP/105,15)
Please tell me the solution. Looking forward to your response.
Thank you.
Regards,
Chandra.
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Get on board. You're invited to try the new Yahoo! Mail Beta.
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Re: [asterisk-users] RE VoipNow 1.2.0 Beta
If developers only ever get feedback from other developers, how will they ever produce something that the market needs? Shouln't you also listen to your customers? And surely someone who uses Plesk already is ideally placed to give an opinion on whether it's suitable? Peter On 08/08/06, Matthew Warren [EMAIL PROTECTED] wrote: Yes it is an addon of Plesk, thats stating the obvious. But while your complaining about people writing stuff to use what are you doing. If your not a developer don't critisize the developers. I see nothing more than you displaying that you are the Vice President of a 2 man consulting firm. Which means you have to sell other peoples developed products. Not to mention you are being critical of plesk, yet you use to host you websites for your business. Dude, we all have opinions, like crapholes they all stink, your's just stood out. ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bugs.digium.com
I hate to say this but you might just have hit a 'reap what you sow' moment - you don't hesitate to trash Asterisk on this mailing list when you can't make it do what you think it should do, and just maybe this affects how the developers treat requests from you on the bug tracker? Just a thought. Peter On 27/07/06, Douglas Garstang [EMAIL PROTECTED] wrote: I opened bug #0007490 the other day. The issue was that when you do a 'sip debug' on the Asterisk console, there was no way to have this output go _only_ to the messages file. Someone with the id of 'russell' in his infinite wisdom has deemed that this isn't a bug, closed it, and given me -2 karma points. WTF??? It clearly is a bug, or at the VERY least, a limitation that needs to be fixed. So why the hell did he give me -2 karma points and say 'not actually a bug'. Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. Its damn irritating not being able to have 'sip debug' output go to a file only, and this is what the options in logger.conf imply you should be able to do, which is another reason I don't understand why he took this irrational action. In a PRODUCTION environment, you can't be running a sip debug to your console. Doug. ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to decrease answer time !
When you hear the phone ring, run faster so you reach it more quickly. On 13/07/06, Pablo Mora [EMAIL PROTECTED] wrote: Pablo Mora, Ing. GERENTE DE OPERACIONES ESPOLTEL S.A. Malecón 100 y Loja Telf.:2514477 ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context
That's the fourth time you've asked the same question in the space of a few hours - please have a little more patience and wait for someone to answer. On 12/07/06, Khaled Chehab [EMAIL PROTECTED] wrote: Since I make call forward to an extension l by default it will attach your DIAL(local/[EMAIL PROTECTED]) from the context from-internal which is linked to a trunk , The script is located at /var/lib/asterisk/agi-bin/dialparties.agi I $dialstring = 'Local/'.$extnum.'@from-internal'; How can I let it find the context ? automatically $context ? Instead of '@from-internal' Please help regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with making outgoing calls
On 12/07/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi, We could make calls to USA using Teliax service upto 11, July 2006 with Asterisk. But, since 11, July 2006 evening, we are unable to make calls sometimes and could not connect to Teliax server sometimes. I have realized that Teliax server was down for few hours. Currently our Asterisk server is connecting with Teliax. But, When I am trying to make call to USA, Its giving me one ring and being disconnected. I could not understand what could be the problem? Is there any problem with my connection to Teliax server? What did Teliax support say? I presume they were your first port of call, since they're the people prividing you with service -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International Dialing setup in extensions.conf
On 05/07/06, Kai Fürstenberg [EMAIL PROTECTED] wrote: Just dial the international number completely (e.g. for Germany 0049etc.) In your extension above a number beginning with 011 is being dialed. That is not an international number. Where were you assuming the OP was dialling from? -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mail loop?
Yes - every message I've sent to the list in the past few weeks is now arriving back here. I'd ignore it, it's harmless... Peter On 27/06/06, Mike Fedyk [EMAIL PROTECTED] wrote: Is anyone else getting messages from the lists.digium.com mail server with errors about a mail loop? I've been getting this for the last few weeks, but I don't have any list software on my server. Any ideas? ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID Matching in extensions.conf
On 23/06/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten = 9220370/1234,3,Playback(tt-weasels)
You have it backwards. The callerid to match goes after the extension,
not before.
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Re: [Asterisk-Users] MWI not working
On 15/06/06, Walid Azab [EMAIL PROTECTED] wrote: Hi everyone, I noticed that the waiting message indicator does not lit when I have a message in my voice mail. Any suggestion why this is happening? You probably need to change the bulb. -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)
SIP is a UDP protocol, and telnet is TCP. You can't test it like that. Have you tried connecting with a SIP client? Peter On 13/06/06, John Klimek [EMAIL PROTECTED] wrote: I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried telnet localhost 5060 but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says connection refused. Finally, I've tried changing the bound address in sip.conf to 127.0.0.1 and 192.168.0.1 but I am still unable to connect using all the methods mentioned above. What else can be the problem? Can I have some sort of iptables problem? ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conference
Have you sent this enough times yet? On 13/06/06, Khaled Chehab [EMAIL PROTECTED] wrote: Any one knows how to make a call conference using a voip gateway connected to asterisk. In mean what should I press (extension) to have another line and make the conference . regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to connect to Asterisk? (simple[?] question)
Try this: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules Peter On 13/06/06, John Klimek [EMAIL PROTECTED] wrote: Ahhh, that would explain it. I setup my firewall (eg. Shorewall) to allow incoming TCP connections to port 5060. I've changed it to UDP port 5060 and it works great! (well, Asterisk says Forbidden, but that's just a simple config problem I'm sure) Which other ports do I need to forward/open to get Asterisk working properly? I'm guessing I only need port 5060 open to my local network... Do I need any ports open for connections from the internet? (eg. incoming connections) On 6/13/06, Peter Bowyer [EMAIL PROTECTED] wrote: SIP is a UDP protocol, and telnet is TCP. You can't test it like that. Have you tried connecting with a SIP client? Peter On 13/06/06, John Klimek [EMAIL PROTECTED] wrote: I'm trying to setup Asterisk on my Linksys WRT54G router and it appears to startup successfully (no errors) and it says it is listening on 0.0.0.0 port 5060, but I am unable to connect to it. I've tried telnet localhost 5060 but it just says connection refused. I've also tried connecting from another machine on my network (eg. telnet 192.168.0.1 5060) but it also says connection refused. Finally, I've tried changing the bound address in sip.conf to 127.0.0.1 and 192.168.0.1 but I am still unable to connect using all the methods mentioned above. What else can be the problem? Can I have some sort of iptables problem? ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AEL #include
On 30/05/06, Douglas Garstang [EMAIL PROTECTED] wrote: Yes, like asterisk-addons not compiling, which for anyome that wants to use cdr-mysql, or realtime, makes it useless. It's a development snapshot, you can't expect it not to have issues. -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is NuFone Really Dead? Same as voipjet
On 25/05/06, Kerry Garrison [EMAIL PROTECTED] wrote: Just because their email address is [EMAIL PROTECTED] doesn't mean its fast, or is even answered. It should be /dev/[EMAIL PROTECTED] I agree. Others seem to rave about them, but I've had no luck attracting their 'fast' support staff's attention, despite many emails and direct followups to their promotional postings on the -biz list. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB headsets?
On 24/05/06, El Flynn [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Well, IMO if the soundcards are already crap to start out with, there's no way a fancy-schmancy USB headset -- or any other headset, for that matter -- will sound good when plugged in to the laptop. Because, remember, it's the soundcard that generates the audio and sends it out the heaphone port. E no - that's the point of using a USB headset - it has soundcard functionality built-in and doesn't use the on-board card. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..
On 22/05/06, Steve Kennedy [EMAIL PROTECTED] wrote: On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote: If you want to roam between GSM and WiFi while on a call, the GSM carrier is going to have to support it. There is a protocol for this (UMA), however few operators as yet support it. T-Mobile offer a webnwalk tarrif (unlimited data access for a fixed monthly fee), but they are are going to (if not already) block VoIP calls - they've realised that users are using VoIP (probably Skype) and not making GSM voice calls - and the voice revenue is declining. They block VoIP and IM, supposedly to protect their users from a poor quality experience. Of course, it's really to protect their voice and SMS revenues. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP useragent?
On 19/05/06, Remco Barende [EMAIL PROTECTED] wrote: Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? sip show peer peername -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help
If you're looking for real-time help, maybe the irc channel would be a better place? Peter On 15/05/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: is their nobody here with a h323 terminal, netmmeting ... I just need a h323 terminal register with asterisk/oh323/gnugk just five minutes just aggressive because of I'm feeling tired --- Administrator TOOTAI [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: hello, I have to test asterisk/gnugk is their somebody, sur cette putain de liste, with a h323 terminal ? No need to be aggressive like that, I don't think it will help your request. And if you think what you wrote, feel free to unsubscribe. -- Daniel ___ --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 ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP or IAX2 via NAT
On 09/05/06, James Nunnerley [EMAIL PROTECTED] wrote: I've installed successfully freePBX with Asterisk, and got various internal extensions working, however… recently my internet facing IP address has been removed by my ISP (for various reason) and I'm not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static IP? I'm sure Voiptalk support would help you with this in not time at all, but... if you use the Voiptalk control panel you can route the DID to your Voiptalk ID hen 'register' to the Voiptalk ID from Asterisk. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 Message Waiting Light
On 01/05/06, Jeffrey Macko [EMAIL PROTECTED] wrote: Does anyone know the secret to get the GXP-2000 Message waiting lamp to illuminate? No secret - just set a 'mailbox' line in the appropriate peer entry in sip.conf. Later GXP-2000 firmware shows the number of messages waiting on the LCD display as well as flashing the MWI lamp (can't remember which firmware version introduced this). Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
On 30/04/06, Steve Totaro [EMAIL PROTECTED] wrote: My question is, how can I throttle the FTP (Standard with dist) transfers using out of the box CentOS4.3 (or any easy to use, low learning curve package)? I thought about FTPing the files at less frequent intervals but that just makes the issue less frequent but last longer. It's a while since I've looked at it, but I seem to recall that ProFTPD has options for bandwidth limiting per login - you could take a look at that. I just took a glance at the online docs - 'TransferRate' - 'The TransferRate directive is used to set transfer rates limits on the transfer of data. This directive allows for transfer rates to be set in a wide variety of contexts, on a per-command basis, and for certain subsets of users. Note that this limit only applies to a single connection, and not to the overall transfer rate of the server.' www.proftpd.org Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. You're trying too hard - unless you tell it not to, the Dial application will do what you're asking. As Olle said, this is the default. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: So, is there any other option that prevents that from happening? Something that I might have turned on and makes Dial work trough asterisk? I already even removed asterisk completelyu from system and reinstalled it to be fresh new... still all RTP goes trough Asterisk machine. And the server really can't handle many connections this way. What options are you using? Post an extract of your dialplan and sip.conf. And how are you determining that the RTP is going through Asterisk? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a good VoIP Provider in the UK-
Telappliant and VoipTalk are the same people, and are great for retail volumes. They have good integration with Asterisk, national DID coverage and reasonable termination rates - you can get pretty near the same rates from US providers at the moment, though. If you have wholesale volumes, Magrathea Telecom are great. Peter On 17/04/06, Maxx Lobo [EMAIL PROTECTED] wrote: Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK customers can call them as a local call AND they can dial out UK numbers as local calls. The obvious side benefit would be that US employees could call UK customers and vendors as a local call as well. I've looked at Telappliant, VoipTalk and PipeCall so far, and I'd like to get some feedback before going with one or the other. I'd be grateful for any opinions on the quality of (these, other) services, how responsive they are to problems, and if they are as easy to setup with Asterisk as TelaSIP. Recommendations are appreciated, of course. Thanks- --Maxx ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Trial Version of Asterisk Interface Available
On 17/04/06, Vikram Rangnekar [EMAIL PROTECTED] wrote: you can fix issue number 3 by running the install script sh ./install.sh or manually running the command touch /var/log/asterisk/druid chmod 777 /var/log/asterisk/druid You'll have difficuly persuading any professional unix admin that 'chmod 777' is a good solution to a problem. It might be a temporary workaround to help confirm where the problem is, but you need a better solution for the real world. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: [dmuars] Eh up - March 144 results altered
Here you go, Ian..-- Forwarded message --From: G3RIR [EMAIL PROTECTED]Date: 05-Apr-2006 20:54 Subject: [dmuars] Eh up - March 144 results alteredTo: [EMAIL PROTECTED] What's going on here. The results of the MArch 144 UKAC have been re-published and we have lost out considerably. Either I don't understand the rules or we have been robbed We scored 1159 G8VHI 928 G3RIR 154 G0TPH 133 G4OIG 333 G4ARI/P 98 G3CWI/P Totalling 2805 Cray have 2158 G4DBL 238 M3RCV 192 G3SPJ 27 G0KPZ 16 M3CVN/P Totalling 2631 Now we won so have 1000 points Cray should have (2631/2805)*1000 = 938 points They have been given 991 points! Why! Perhaps Peter can point out my error before I raise the issue with the adjudicator. Neil, G3RIR SPONSORED LINKS Craft hobby Hobby and craft supply Ham radio De montfort university YAHOO! GROUPS LINKS Visit your group dmuars on the web. To unsubscribe from this group, send an email to:[EMAIL PROTECTED] Your use of Yahoo! Groups is subject to the Yahoo! Terms of Service. -- Peter BowyerEmail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [dmuars] Eh up - March 144 results altered
Oops! Fat fingers, sorry, all. On 06/04/06, Peter Bowyer [EMAIL PROTECTED] wrote: Here you go, Ian.. ___ --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] Who is on a call?
On 03/04/06, Douglas Garstang [EMAIL PROTECTED] wrote: The 'sip show channels' and 'show channels' command aren't exactly easy to interpret, especially if one of the numbers has pic codes and rate centers inserted (the rest is truncated on the output), or you have a proxy involved in the call. Wish someone with some C knowledge would fix that. Did you post a bug? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 2.0 Where to download
On 02/04/06, Nguyen Trung Tin [EMAIL PROTECTED] wrote: Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. That version is a year and a day old now, isn't it? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Building Asterisk embedded device
On 31/03/06, sam [EMAIL PROTECTED] wrote: Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Depends what horsepower you'll need - many people have had good results with the Soekris NET4801, running Astlinux. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Vonage
On 29/03/06, Steve Jones [EMAIL PROTECTED] wrote: I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! I'm not a Vonage customer, but I did spot this: http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credentials.asp Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Regulatory Ruling about Caller-ID
On 29/03/06, Matt [EMAIL PROTECTED] wrote: Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? A ruling in what jurisdiction? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
On 27/03/06, James Harper [EMAIL PROTECTED] wrote: Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. It strikes me as really strange that GSM/Wifi would be available while GSM/DECT is not so much. DECT is a voice technology, while wifi isn't. 1) Because the phones do so much more than voice calls. Would you run a web browser over DECT, or would it work better over wi-fi? 2) How many public DECT hotspots do you know about? 3) How many companies have deployed DECT in their buildings? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] AMP and ABE
On 20/03/06, James Sturges [EMAIL PROTECTED] wrote: Google is a good friend, unfortunately the system admin who represent the company we are installing is not so. They a requiring an audited stable platform, aka Asterisk Business Edition. So when we say we need to install non-certified package onto their Enterprise Server, they say na! Then shouldn't you be requesting support from the supplier of that audited, stable platform, instead of requesting community support? Isn't that why you (they) bought it? No much point otherwise. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] regexten
On 17/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, I finally got it to work. Such a shame I can't use it. I didn't realise it until I'd expended all the effort, but this approach doesn't give you a HA asterisk solution. If the server that the phone is registered to goes down, no Asterisk system knows the location of the phones that where registered to it. If you had 3 Asterisk boxes, 1/3 of your users suddenly can't receive calls. Not HA! It does if you combine this with an IAX switch or DUNDI, and phones which re-register fast. Did you read how this was explained to you a few days ago? -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Feedback from VON expo! Info on * HA and Polycomphone!!
On 17/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: David, How's DUNDi make this redundant? The way I understand it, a phone is only ever registered to a single Asterisk box at a time. If that Asterisk box where to fail, callers lose the ability to contact users that where registered on that box. Assuming you're using regcontext/regexten, if a phone isn't found locally then a DUNDi switch will allow it to be found elsewhere - assuming it has re-registered. It will only be unavailable during the re-registration interval. You need to go and try this, not keep posting about how you think it won't work. Then we can help you get it going. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Feedback from VON expo! Info on * HA andPolycomphone!!
On 17/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Peter, Sorry, can't do inline replies with web based Outlook. Yeah - tell me about it. Why not get a Gmail account for doing mailing lists - 2000% better that OWA. Yes, I know... I'd say that if the phone wasn't available between the period the server went down, and the time it re-reregistered, then it's not a HA solution. I discussed with the people that make the decisions this evening, and having downtime like that is not an option. Fair enough, they're the folks with the money. Our job to make sure they understand what they're spending it on. You'd also have no reliable BLF functionality as the subscribe and notify messages would be spread amongst multiple systems and just plain wouldn't work. Yes, I can see that would be a problem. Might be some work-around. Can you imagine what sort of traffic, say even 1000 phones re-registering every minute would be like? That's SEVENTEEN new registrations per second. Yeah - sounds a lot, but only when you're watching a SIP debug. What traffic do 1000 PCs produce against a Windows server? I wouldn't discount it simply on that factor. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Clustering
On 13/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... Like Kevin already said: If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. Use an IAX2 switch for a small, known number of servers. Consider DUNDi to extend into a larger, more dynamic 'cloud'. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Clustering
On 13/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Now that I've read that paragraph of Kevin's a few times, it strikes me that this is not a redundant configuration. If the call is handled by the Asterisk system where the phone registered, what happens if that system becomes available? Can another system (one that did not handle the registration) process the call? (Any chance you could format your emails for easier quoting? Thanks) Something like this: Server A [sip-registrations] exten = peer1,2,Dial(SIP/peer1) exten = peer2,2,Dial(SIP/peer2) include = switch-server-b [switch-server-b] switch = IAX/user:[EMAIL PROTECTED]/sip-registrations So a call arriving in context sip-registrations will hit any peer which has registered (with the regcontext trick), and fall through to the 'switch' for any which hasn't. Server B has the opposite. This won't help with a failure for an in-progress call, but should automatically distrubute calls around your peers which are registered with one server or the other. If the phones know how to re-register in the event of a server failure (and I think you said you use a SRV-based system for this), then something good should be able to happen. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Clustering NEW THREAD, Almost Working
On 14/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: DUNDi... pfft... forget it. No docs... it's useless. How about qualifying these blanket put-downs something like: DUNDi - I couldn't find any docs that helped me with it, so I decided not to invest any more time ? (Which is an entirely OK position to take - your call - declaring it as useless without giving it a try, however, is not really helpful or accurate). Others are busy finding it very useful indeed. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Advice on configuration
Hi Paul I am looking to see if this is possible and any pointers if it is. It seems straight forward but not too sure. I have 4 extensions 2000 to 2003 I have one voip external account with Sipdiscount. I want any of the 4 extensions to share that single sipdiscount account. 'share' as in dial out through? Assuming they're SIP phones several ways to do it, here's my favourite sip.conf [phone1] .. context=sipphones ... [phone2] .. context=sipphones ... [sipdiscount] stuff about your sipdiscount account extensions.conf [sipphones] other-things-you-want-them-to-be-able-to-dial include = sipdiscount-outbound [sipdiscount-outbound] exten = somepattern,1,Dial([EMAIL PROTECTED]) etc I also have 2 voip incoming numbers through another company (sipgate). I want one of these to ring 3 phones and the other one to ring the 4th extension if dialled. Is that possible? Yep sip.conf register =:[EMAIL PROTECTED]/111 register =mmm:[EMAIL PROTECTED]/222 [sipgate] type=friend host=sipgate.co.uk insecure=very context=sipgate-inbound extensions.conf [sipgate-inbound] exten = 11,1,Dial(SIP/2000SIP/2001SIP/2002) exten = 22,1,Dial(SIP/2003) Give me a shout if you want more help Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Streaming Music On Hold
On 22/02/06, Douglas Garstang [EMAIL PROTECTED] wrote: Thanks. I got it working. Yay. Now, it seems that Asterisk is very fussy with the streams. A lot don't work, especially when the URL ends in something.pls. Anyone know if that's true? Is Asterisk's support of this still pretty limited? something.pls isn't a stream, it's a playlist which (probably) lists streams within it. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] sipdiscount
On 08/02/06, Alejandro Vargas [EMAIL PROTECTED] wrote:
Sipdiscount has replaced their asterisk servers for another thing.
Then, no more iax. Ok, but I can't make calls using sip also... I'm
getting a forbidden error when using sip1.sipdiscount.com. Anybody
got it working?
A pretty simple setup works for me:
sip.conf:
[sipdiscount]
type=peer
host=sip1.sipdiscount.com
username=xxx
secret=yyy
canreinvite=no
dtmfmode=info
extensions.conf:
[sipdiscount-out]
exten = _6.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _6.,2,Hangup
(I use a prefix of '6' to reach sipdiscount)
Peter
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Re: [Asterisk-Users] Sendmail with exchange
On 10/02/06, Jordan Novak [EMAIL PROTECTED] wrote: I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much appreciated. Install MSMTP as your local MTA (replacing sendmail). Configure Asterisk to use the local MTA, and configure MSMTP to forward to the Exchange server with authentication. http://msmtp.sourceforge.net/ Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
YAC is a nice popup application (for Windows) to eat alerts just like
the one below.
http://sunflowerhead.com/software/yac/
Peter
On 06/02/06, Facundo Ameal [EMAIL PROTECTED] wrote:
If you wnt to do it quick, I've seen this in another post of this
list, and I think is good:
exten = s,1,System(/bin/echo -n -e '${CALLERIDNAME}
${CALLERIDNUM}'| nc -w 1 192.168.1.16 10629)
then you have tyo be monitoring that port and capture the information,
you can do that in VB.
2006/2/6, Facundo Ameal [EMAIL PROTECTED]:
First, about the Jabber library: I'm using Asterisk Perl and the
Jabber module for Perl.
About dinmically loading the jabberid list, welll that's the problem I
had and now I'm developing that. I thought about (and it's what I'm
doing) generate a little database in XML in which you would put
jabberid and extension so if you know the extension, you know the
jabberid... what do you think about that?
2006/2/3, Andrew Kohlsmith [EMAIL PROTECTED]:
On Friday 03 February 2006 10:21, Facundo Ameal wrote:
I 'm developing something similar. It a perl script which tells you
who is calling but it do it by sendind a jabber message.
it's my first perl script so it's not finished yet.
i'll share it so you can contribute if you want...
http://www.mixdown.ca/~andrew/astbot
-A.
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Facundo Ameal.
famealatgmaildotcom
Linux User #395088
FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793
Open your mind, use open source.
--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793
Open your mind, use open source.
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VoIP: [EMAIL PROTECTED]
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Re: [Asterisk-Users] UK Provider
You can do without registration these days - the Voiptalk routing for DIDs allows you to specify a SIP URI directly. (Route to an external provider). You still need to register if you want to receive calls to your @voiptalk.org number, though. Peter On 25/01/06, Morgan Gilroy [EMAIL PROTECTED] wrote: Yeah, http://www.voiptalk.org with one registration you can receive as many numbers as you like. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of scott Sent: 24 January 2006 09:14 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] UK Provider Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] UK Provider
On 24/01/06, scott [EMAIL PROTECTED] wrote: Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Which part of Asterisk? register = nn:[EMAIL PROTECTED]/m register = oo:[EMAIL PROTECTED]/ Works fine for me Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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 do you deal with subprefixes with LCR?
On 17/01/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX0.15 You're stuck, because you cannot decide if provider B's XXX prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste to try and contact it. Or maybe worse, you might be dialing a destination which /does/ work but is not displayed in the rates list and could be billed a lot more. I guess you need to determine each provider's rate for the route in question separately using the 'longest first' algorithm, then compare the rates you've found. Deal with the 'doesn't cover routes not specifically listed' issue as an attribute of the provider. Or a dummy catch-all at $99.99/min which your code knows never to select. In the end, if the provider doesn't give sufficient information about their charges for routes not specifically listed, there's not much you can do... Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll
On 17/01/06, Colin Anderson [EMAIL PROTECTED] wrote: Disclaimer: Not trolling. Cross-posting to -users to gague support. -users : Straw poll - if an XML based Manager Interface was avaliable as an option in asterisk.conf, would that be a good thing, or a stupid thing? Good thing. Make it loadable (or do I mean noloadable) so those who don't like the idea don't suffer any overhead. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Nested MySQL Commands
On 11/01/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. When things start to get that complicated, I reckon it's time for AGI Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Nested MySQL Commands
On 11/01/06, Douglas Garstang [EMAIL PROTECTED] wrote: Peter, Too slow! We're going to potentially be doing several MySQL lookups for routing even the most basic of calls, and if every one of those queries has to make a call out to an AGI script, it would become a performance problem. I mean, an AGI to do the routing. A single call which does all the MySQLing it needs to do in a manner efficient for the environment it's written in, and makes all the call routing desicions as well. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Presence support on GrandStream GXP-2000
On 10/01/06, Richard Smith [EMAIL PROTECTED] wrote: Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Yes - with the latest beta firmware (1.0.1.13). Working well for me in a SOHO environment. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] pattern matching in dialplan problems matching _NNN
On 07/01/06, Thomas [EMAIL PROTECTED] wrote: Hi, I have a problem with pattern matching N what should digit 2 to 9 in Asterisk 1.2.1. If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the context dialout and find there an matching _2. and is using this. If I change _NNN to _XXX everything works fine. If I dial 220 I hear playtones invalid. It seemed to be that pattern matching with N is not working as designed. 220 isn't supposed to match _NNN - N is digits 2-9, not 0. http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] bayhamsystems.com experience
On 06/01/06, Michiel van Baak [EMAIL PROTECTED] wrote: Anyone using their services ? I'm thinking of setting up my servers with their service. But before starting to mess with my extensions.conf I thought let's check the community for their experience. I use them - the service works exactly as advertised. Recommended. I use the perl version of their AGI (so I could hack it easily) - actually I really only used it as a building block for a more extensive MWI management system. The samples they provide are not foolproof, there's more logic needed to do the job properly. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Looping Problem With Call Forwards - Do you have comments on my solution?
On 03/01/06, Brent Torrenga [EMAIL PROTECTED] wrote:
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my extension at work, and my cell phone via NuFone.
Problem: A loop can be created if my cell phone is not on. Say a call comes
into my * box, it uses NuFone to call my cell, my cell forwards the call to
IP Kall, IP Kall to my * box.. You see.
My solution, and I post this here because I am looking for
comments/improvements to it: When a call comes into my * box, and my * box
dials my cell via NuFone, I will SetGlobalVar ZAPCALLEDTIME to the ${EPOCH}.
Then, whenever a call comes into my * box from IP Kall (aka, any call
forwarded from my cell to my IP Kall), I will take the difference between
the current ${EPOCH} and ${ZAPCALLEDTIME}, compare it to the value 10
(thinking that if it takes less than 10 seconds from the time I forward a
call to my cell, and a forwarded cell call comes into my * box, then it must
be the beginnings of a loop), and if less than 10 to send the call on to be
hungup, or else process it normally.
A simpler solution would be to have the IPKall number forward to a
different extension on your Asterisk server which doesn't include a
call out to the cellphone - and if you use the IPKall number for other
purposes, register yourself another one which you only use for the
cellphone forward.
Peter
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Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
On 04/01/06, Alistair Cunningham [EMAIL PROTECTED] wrote: Tijmen, We use SER for this to load balance across multiple Asterisks. We then use a custom program to monitor the health of the Asterisks and update SER's configuration should one go down. 2 SERs share a single IP address for users to contact using heartbeat. I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
On 04/01/06, Alistair Cunningham [EMAIL PROTECTED] wrote: Peter Bowyer wrote: I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though One being that it must be the device that NAT phones register with that delivers calls to them. Otherwise, the NAT device sees a packet coming from an unknown IP address and drops it (for common types of NAT such as restricted cone). Yes, that's the sort of reason I was thinking of :-) I guess you could NAT the whole cluster behind a single IP with some fancy firewall/router rules Since SER needs to deliver calls, it really needs to be SER that accepts REGISTERs and holds the registration information. The Asterisks then send calls from phones to the SER heartbeat address for delivery. And if a lot of the calls are SIP-SIP, I guess - why bother Asterisk with them at all... This is what we do in our ITSP in a box product. It gives us full redundancy and failover with the registration capacity of SER and the features of Asterisk. Sounds good. For very large systems, it's possible to have SER redirect (with load balancing) REGISTERs to a set of SERs so that NAT devices know about the machines their phones are registered on, but this takes great care to get right in all cases. Yeah - I knew this was harder than it looked :-) Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] GXP-2000 fw 1.0.1.13 and NTP
Hi all Slightly OT but I know a lot of GS experts hang out here - I just upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk (which so far works as expected), but as a side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ? (Hmmm.. just regressed to 1.0.1.12 and it's still not working - curiouser and curiouser said Alice...) Thanks Peter ___ --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] call test
Your SER is requesting authentication from my SIP client. On 28/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, I need to test my configuration please to dial sip:[EMAIL PROTECTED] . Your call will be sent to a queue . Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Creating conf files from db
On 23/12/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Dec 23, 2005 at 07:21:54AM +, Peter Bowyer wrote: On 22/12/05, Douglas Garstang [EMAIL PROTECTED] wrote: Just wondering if anyone here has tried the approach, where all config files are stored in a database, maybe using the ast_static table structure. Rather than using realtime to access the database live, you have scripts that read the contents of the db, and generate the .conf files from that., and then do a 'reload'. Anyone tried that? How'd it work for you? http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database Specifically, option 4b. You have scripts to do the bulk of this in your /contrib directory. And they are badly written. E.g.: the ammount of code replication between them. I guess they're written to be used independently - although I'm sure if you cared to contribute a 'better' version it would be well received -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Creating conf files from db
On 22/12/05, Douglas Garstang [EMAIL PROTECTED] wrote: Just wondering if anyone here has tried the approach, where all config files are stored in a database, maybe using the ast_static table structure. Rather than using realtime to access the database live, you have scripts that read the contents of the db, and generate the .conf files from that., and then do a 'reload'. Anyone tried that? How'd it work for you? http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database Specifically, option 4b. You have scripts to do the bulk of this in your /contrib directory. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SIP Subscriptions
On 21/12/05, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Oh, and no... I can't switch to another solution. The decision was made above my head to go with Asterisk. It's my job to make it do all that 'enterprise-grade' stuff. Aha, so *this* is where the big chip on the shoulder comes from. Yes, interesting logic - someone made a decision you don't agree with, so you decide the best course of action is to yell at the people who developed the product you've been made to use (who are the only people with the ability to rescue you from the hole you're in), rather than at the person who made the decision... Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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
