[asterisk-users] E-911 and Asterisk
Can someone point in the right direction to learn how to implement e911 while using asterisk? I am looking for a how to or general documentation on how to implement e911. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 67
Hi, What version of asterisk and zaptel are you using? I've used digium TE110P card with mfcr2. I had same T1 timed out problem too. I've asked from support and got answer to install latest zaptel. I've installed zaptel 1.2.13 and it solved. In short try latest asterisk v1.2.x packages with unicall. Hi List, I'm trying to connect Asterisk with a Nortel Meridian using an E1 with MRFR2 signaling. I've connected both cards, and compiled all the required software... The problem is every call (outgoing or incoming) got dropped, complaining about some T1 timed out Only for testing purposes I'm using an application called testcall included on the lib-unicall package, and this are the logs: Chan 1, class 'mfcr2', variant 'mx,0,4', end 335544324, caller 1, from '7000' to '6640' Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 1: Call control(8) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/4000/Idle /Idle ] MFC/R2 Chan 1: far_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Far end unblocked! :-) Chan 1: -- Far end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: Initiating call MFC/R2 Chan 1: Call control(1) MFC/R2 Chan 1: Make call MFC/R2 Chan 1: Making a new call with CRN 32769 MFC/R2 Chan 1: 0001 - [1/ 1/Idle /Idle ] Chan 1: -- Dialing on channel 0 Chan 1: -- Dialing on channel 0 MFC/R2 Chan 1: - 1101 [1/ 40/Seize /Idle ] MFC/R2 Chan 1: 6 on - [2/ 40/Group I /Idle ] Main thread MFC/R2 Chan 1: R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out MFC/R2 Chan 1: 6 off - [1/ 1/Idle /Idle ] MFC/R2 Chan 1: 1001 - [1/ 1/Idle /Idle ] Chan 1: -- Protocol failure on channel 0, cause (32769) T1 timed out Chan 1: -- Protocol failure on channel 0, cause (32769) T1 timed out MFC/R2 Chan 1: - 1001 [1/ 1/Idle /Idle ] MFC/R2 Chan 1: 1001 - [1/ 1/Idle /Idle ] Have anyone know something about this?? -- Ing. Arturo Ochoa N Network Administrator Electrosystems, ___ --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 -- Best regards, Yeruultmailto:[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] SMS Integration and SMS commands
Hi i would like to preform the folowing integration 1 send SMS to my asterisk Systen, i hope to know a way to connect a simple GSM or CDMA cell phone to asterisk (how can i connect a cell phone to asterisk) 2 be able to put in the SMS string a command to generate a call (can i pass thru instructions to my server via SMS ? ) 3 after the the SMS is sent my Asterisk system will see the incoming Caller ID, it will call to the number i gave in the SMS and call me back to my cell phone with the other call allready connected, using my caller ID so with this i looking forward just only with a SMS tell my asterisk to call me back with other number connected, and with this i hope to save my cell phone min. so if anyone has done this please let me know how Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile
On Fri, Mar 16, 2007 at 11:43:21PM -0400, John Novack wrote: Tzafrir Cohen wrote: On Fri, Mar 16, 2007 at 03:55:56PM -0400, John Novack wrote: Tzafrir Cohen wrote: On Fri, Mar 16, 2007 at 11:28:43AM -0400, Sean Bright wrote: Whats the non-workaround solution? Is there one? http://bugs.digium.com/view.php?id=9303 Please test. Wasn't there an existing issue on this one? The bug traker says, to me at least, that this problem doesn't exist in the tarballs, but it does. Zaptel 1.2.15 installed, and attempts to make asterisk- 1.2.16, and asterisk-1.2.15 fail/ Asterisk-1.2.14 compiles with no errors This is on a CentOS 3.8 system Asterisk 1.2.X fails with zaptel 1.2.15? This is something new. What error do you get? The same string of errors mentioned earlier in the thread The issue we had here was asterisk 1.2.16 (and maybe slightly older versions) with zaptel 1.2.13 . Asterisk 1.2.14 compiles, 15 and 16 do not I am using the slightly modified version supplied by govarion for the tor3 4 port T1 card. I suppose it is possible that govarion slipped a file in somewhere that doesn't belong, but I am not smart enough to figure that one out. Here are the modified zaptels supplied by them http://govarion.com/tor3/ I can try the Digium distribution over the weekend and see if the errors persist, but will have to revert back to keep the T1 card in service. Interesting. Could you kindly post the actual errors you get? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] E-911 and Asterisk
www.intrado.com :) There is no way to do it yourself. You will need to contract with a PS/ALI vendor. On 3/17/07, Davis Sylvester III [EMAIL PROTECTED] wrote: Can someone point in the right direction to learn how to implement e911 while using asterisk? I am looking for a how to or general documentation on how to implement e911. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any asterisk scripter around?
We are looking for someone who can help us to write script for asterisk for doing virtual switchboard at the moment. If you are interested please email me back on sam__tam AT hotmail Dot com Sam ___ --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: Zaptel version for asterisk 1.2.16
I discussed this with Digium techs, who recommended using the 1.4 version of Zaptel with the 1.2 version of Asterisk, at least with my hardware Hi Brent, It seems like zaptel 1.2.15 and asterisk .16 are working now, thanks to a quick hint from Tzafrir about changing a Makefile. Commenting out codec_zap in the codecs makefile allowed me to make and install asterisk. There was a small change in the behavior of Voicemail and it requires an explicit context which it apparently didn't before. The only problems I have now will be looking at RxFax to see if I want to try a later version and finding out whether g729 module is still good. And perhaps a few unknown gotchas I'll find out about while testing. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call counter for sip misbehaving
Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10] ERROR[7664]: chan_sip.c:3030 update_call_counter: Call to peer 'rehmat' rejected due to usage limit of 2 -- Couldn't call rehmat == Everyone is busy/congested at this time (0:0/0/0) Im using asterisk1.4.0 . declaring type=peer solves the problem. but if anybody knows why its not working for type=friend, plz share. -- Regards Rizwan Hisham Software Engineer ___ --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] Any asterisk scripter around?
Sam Tam wrote: We are looking for someone who can help us to write script for asterisk for doing virtual switchboard at the moment. If you are interested please email me back on sam__tam AT hotmail Dot com Sam You may want to post this to the biz list, assuming you are looking to pay someone to do this for you. You may want to take a look at Flash Operator Panel, Nico has a very nice virtual switchboard already. Thanks, Steve Totaro ___ --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] Dell poweredge 860 acceptable for office environment ?
On Fri, 16 Mar 2007, Olivier wrote: I'm really after 1U-2U silent servers as I've got the feeling most of them are too noisy for offices and most of our clients don't have server rooms. How about a mini tower type unit? I've just bought one of these: http://www.asus.com/products.aspx?l1=9l2=40l3=121model=1017modelmenu=2 and while the fans can crank themselves up to full turbo mode, when running all the usual stuff I do to soaktest my servers, they've stayed at the lowest possible speed and have been very quiet. (and you can stuff GB of RAM and a dual-core processor in it which would satisfy the asterisk needs of a small office trivially!) Or if you really need to put it in a rack, a fanless 1GHz Via processor in a 1U rack fitted with a 2.5 laptop drive will more than satisfy your needs for 10 users. Gordon [I notice the list, or probably a subscriber is duplicating posts again )-: ] ___ --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 branch revision 53132 failed to compile
I would check to see if you had the latest version of the zaptel library for the version of asterisk you are trying to compile. Has this issue been resolved? I'm having the problem now with the code base downloaded yesterday. I should have said the code base for zaptel and asterisk I downloaded yesterday which was 1.2.15 and 1.2.16 respectively. Problem solved, thanks Tzafrir I'm not that comfortable guessing about what to not make and don't have time for trial and error. The fact that someone immediately told me what not to compile in asterisk showed me where to go to fix it. When you only complie once or year, you tend to forget all about modules (other than chan_sip and iax2 etc) and what they do :) ___ --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] detecting missing end-of-queue records
Hello list, we noticed that in some conditions Asterisk would not log queue exit records, thus producing a queue_log lacking some end-of-queue events. I have then prepared a small tutorial to get you started on tracking queue exit conditions that cause the problem. http://astrecipes.net/index.php?n=256 Comments/suggestions are welcome. l. -- Loway Research - Home of QueueMetrics http://queuemetrics.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
Re: [asterisk-users] E-911 and Asterisk
Matt wrote: www.intrado.com http://www.intrado.com :) There is no way to do it yourself. You will need to contract with a PS/ALI vendor. On 3/17/07, *Davis Sylvester III * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Can someone point in the right direction to learn how to implement e911 while using asterisk? I am looking for a how to or general documentation on how to implement e911. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users What do you mean there is no way to do. Others have done this so there is a way. I thought the taught the purpose of this list was to HELP THANKS, for the HELP ___ --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] E-911 and Asterisk
Davis Sylvester III wrote: Matt wrote: www.intrado.com http://www.intrado.com :) There is no way to do it yourself. You will need to contract with a PS/ALI vendor. On 3/17/07, *Davis Sylvester III * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Can someone point in the right direction to learn how to implement e911 while using asterisk? I am looking for a how to or general documentation on how to implement e911. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users What do you mean there is no way to do. Others have done this so there is a way. I thought the taught the purpose of this list was to HELP THANKS, for the HELP Yes, there is a way to do it yourself. It is very complex so the audience is way too small for anybody to offer a free howto or docs on the web. The total costs will not be low, so you might as well start out by hiring a consultant with the right experience. Begin by posting your need for such a person on http://www.asteriskhelpdesk.com/, the -biz mailing list, the jobs forum on http://forums.digium.com/ and similiar places. ___ --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] E-911 and Asterisk
On Mar 17, 2007, at 11:53 AM, Davis Sylvester III wrote: What do you mean there is no way to do. Others have done this so there is a way. I thought the taught the purpose of this list was to HELP THANKS, for the HELP Wow, kinda rude don't you think? Matt WAS Helping, why aren't you listening? Contact your state SETB and ask about connecting to their selective routers so you can send calls direct to the PSAPs. You'll also need to update the location of your DIDs and validate them against the MSAG.Normally you'll need to use IMTs to connect to the selective router and IMTs require SS7 with ISUP routes to your LECs STP. You do have SS7 on your Asterisk box right? If you want to provide cross LATA E-911 then you'll need to connect to each pair of Selective Routers in each LATA. You'll also need ISUP routes to each Selective Router.Go to www.nena.org and read about your E-911 responsibilities. Make sure you setup your Asterisk box so once a customer dials 911 they CANNOT hang up the phone. You also need to make sure you have redundant routes to each Selective Router and provide a failover to a live operator in case the call can't complete (in short, a 911 call must ALWAYS complete to a live person). Only a 911 operator can hang up a 911 call... Or, you could call Intrado.com and save a ton of money and headaches. Oh, an don't forget about CALEA, good luck with that one. But hey, what do I know about VoIP, SS7 and building a CLEC? I run a full CLEC, with full SS7 connectivity to Verizon, redundant routes to the Selective Routers for my LATA. Local number portability database access, all that fun stuff. I still outsource my 911 traffic to Intrado. -- Matthew S. Crocker President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.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
Re: [asterisk-users] Refund from SellVoip?
At 02:32 PM 3/16/2007, you wrote: You were able to cancel service with Sellvoip? That's impressive, that Actually, it's Voxee I tried to cancel and failed. I still use SellVOIP and it mostly works but support is a problem. I'm starting to use using Telasip more though as they work and have a POP only 19ms from here, a big advantage. ___ --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] E-911 and Asterisk
Matthew Crocker wrote: On Mar 17, 2007, at 11:53 AM, Davis Sylvester III wrote: What do you mean there is no way to do. Others have done this so there is a way. I thought the taught the purpose of this list was to HELP THANKS, for the HELP Wow, kinda rude don't you think? Matt WAS Helping, why aren't you listening? Contact your state SETB and ask about connecting to their selective routers so you can send calls direct to the PSAPs. You'll also need to update the location of your DIDs and validate them against the MSAG.Normally you'll need to use IMTs to connect to the selective router and IMTs require SS7 with ISUP routes to your LECs STP. You do have SS7 on your Asterisk box right? If you want to provide cross LATA E-911 then you'll need to connect to each pair of Selective Routers in each LATA. You'll also need ISUP routes to each Selective Router.Go to www.nena.org and read about your E-911 responsibilities. Make sure you setup your Asterisk box so once a customer dials 911 they CANNOT hang up the phone. You also need to make sure you have redundant routes to each Selective Router and provide a failover to a live operator in case the call can't complete (in short, a 911 call must ALWAYS complete to a live person). Only a 911 operator can hang up a 911 call... Or, you could call Intrado.com and save a ton of money and headaches. Oh, an don't forget about CALEA, good luck with that one. But hey, what do I know about VoIP, SS7 and building a CLEC? I run a full CLEC, with full SS7 connectivity to Verizon, redundant routes to the Selective Routers for my LATA. Local number portability database access, all that fun stuff. I still outsource my 911 traffic to Intrado. -- Matthew S. Crocker President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.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 Thanks, that's the type of information I was looking for. I want to read documentation myself and make a decision based on my needs and how much time I can devote to providing e911 services ourselves. I wasn't being rude, but I requested assistance, and was told it couldn't be done, when there are 30+ companies listed on the FCC sites as providing e911 services. So I knew it could be done, I was wondering what the request of your were doing to meet e911 services. I'll read up on it and if we choose to move ahead with it I will document the process and place on a e911 wiki.But nevertheless thanks for all the replies, it helps in getting me pointed in the right direction. ___ --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] Queues
A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
Re: [asterisk-users] Queues
Yes. On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Queues
On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote: Yes. to which bit? auto-agent (as per resource) or voicemail to an agent? Steve On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
[asterisk-users] 1.4 sample postgresql configs
Here are some 1.4.x sample basic postgresql configs w/sh1tl1sting, cid rewriting and 8xx # blocking http://www.kfuq.net/asterisk/cfgs/ I forgot to add these configs also have pgsql voicemail storage and uses ogg vorbis for emailed voicemail messages :-D signature.asc Description: PGP signature signature.asc Description: OpenPGP digital signature ___ --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] Queues
If you make a SIP device a queue member, that member will be rung so long as the device state of the SIP interface shows as not in use. With regard to voicemail, are you trying to get a queue call answered by voicemail or is that not your intent? On 3/17/07, Steve Kennedy [EMAIL PROTECTED] wrote: On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote: Yes. to which bit? auto-agent (as per resource) or voicemail to an agent? Steve On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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 -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [asterisk-users] shutdown
The stop now command tells asterisk to stop right away. what you are seeing is asterisk shutting down. - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 15, 2007 9:54 PM Subject: [asterisk-users] shutdown somebody can help me with this message I don´t understand *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Connection reset by peer Asterisk cleanly ending (0). thanks _ José -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4 Follow-Me Application
Ok, bug report submitted. 0009307 -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Friday, March 16, 2007 6:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Follow-Me Application On 3/16/07, Kevin Kiely [EMAIL PROTECTED] wrote: I am having an issue with the follow me application in 1.4 The application description (below) indicates that if the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. That's not happening for me and the execution terminates not continuing to the next priority in the dialplan. Can anyone confirm this? Thanks, Kevin exten = 502,1,Followme(cell|s) exten = 502,2,Playback(goodbye) exten = 502,3,Hangup -- Executing [EMAIL PROTECTED]:1] FollowMe(SIP/2101-b6e17f60, cell|s) in new stack [Mar 16 23:29:34] WARNING[10814]: app_followme.c:954 app_exec: Profile requested, cell, not found in the configuration. == Spawn extension (from-sip, 502, 1) exited non-zero on 'SIP/2101-b6e17f60' [Description] FollowMe(followmeid|options): This application performs Find-Me/Follow-Me functionality for the caller as defined in the profile matching the followmeid parameter in followme.conf. If the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. Options: s- Playback the incoming status message prior to starting the follow-me step(s) a- Record the caller's name so it can be announced to the callee on each step n- Playback the unreachable status message if we've run out of steps to reach the or the callee has elected not to be reachable. Returns -1 on hangup I can't confirm it just now but I can certainly fix it if you post a bug on bugs.digium.com about it. :) Thanks! -- Bird's The Word Technologies, Inc. http://www.btwtech.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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.11/723 - Release Date: 3/15/2007 11:27 AM ___ --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] Cepstral voices
Lee Jenkins wrote:
Funny you should mention FastAGI. I am implementing a variation of my
DTSwift app through an Object Pascal based FastAGI scripting server now.
http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm
The newer version just uses the System() AGI command to build the file
to play through the shell. I'd be open to any suggestions for a more
efficient way of doing it.
Sorry for not answering faster - was busy shoveling snow.
Once you've got app_swift installed, saying something
should be pretty close to:
AGI.Exec('Swift','This is something to say.');
And since it doesn't render to a file first you'll probably experience
less of a delay to say something.
Steve
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Re: [asterisk-users] Cepstral voices
Best code comment ever, by the way: here's a for loop for i := 1 to iLen do :-) On 3/16/07, Lee Jenkins [EMAIL PROTECTED] wrote: Steve Prior wrote: Julian Lyndon-Smith wrote: what input text ? To what application ? I agree completely with the app_swift suggestion from loopfree as Kai suggested. It provides the app_Swift which you can use from within a dialplan. In fact, if you're getting fancy by using a fastAGI bound language(as I'm doing with asterisk-java), app_swift becomes the only good option. slight rantI think Cepstral should be providing an app_swift like binding themselves because if you're writing an application which is going to use information from a back end business model in creating the speech (and this is something they seem to think is their future and I agree), then a high level language through fastAGI seems by far the best way to control the call. /slight rant Funny you should mention FastAGI. I am implementing a variation of my DTSwift app through an Object Pascal based FastAGI scripting server now. http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm The newer version just uses the System() AGI command to build the file to play through the shell. I'd be open to any suggestions for a more efficient way of doing it. -- Warm Regards, Lee ___ --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 -- sean ___ --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] Jajah.com like script?
you can also have a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+click+to+call I implemented the PHP script mentioned in the comments and it works like a charm. It will require some modification to achieve what you want to do, but that shouldn't be too difficult. Steve Edwards wrote: Or supply the second channel as the data to the dial application. Ritesh, please read the documentation and experiment a bit so you can ask more challenging questions :) On Fri, 16 Mar 2007, mitcheloc wrote: You are missing something. Initiate the call to channel for the first user (i.e. ZAP/g1/phonenumber), and have their destination extension the second phone number. On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: So does this mean that we have to dump the two callers into a Meetme context??? The problem there is what if one of the callee's doesn't accept the call (call screening). There is no easy way to kick the other user out of meetme and dump him to a vmail context. Am I missing something? R On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Thanks Steve! I will give it a shot. R On 3/16/07, Steve Edwards [EMAIL PROTECTED] wrote: Search on voip-info.org for call files. On Fri, 16 Mar 2007, Ritesh Agrawal wrote: Hi Folks, I am planning to create an internal portal where the users can enter two phone numbers (theirs and the party they are trying to reach) and connect the two of them by initiating two calls from Asterisk. Any pointers on how to initiate two calls and then bridge them (without using meetme?). Ideally, I would like to do a call screening as well. Thanks for your help. R Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- Nicolas Brisac Solutions First 9476 0076 0404 849 629 http://www.solutionsfirst.com.au ___ --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] Cepstral voices
Sean Bright wrote: Best code comment ever, by the way: here's a for loop for i := 1 to iLen do :-) LOL, I know. The script was originally to show some of the standard language features supported by the scripting engine. But then I started writing all the db access and cepstral examples and got off on a bit of a tangent... -- Warm Regards, Lee ___ --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] Cepstral voices
Steve Prior wrote:
Lee Jenkins wrote:
Funny you should mention FastAGI. I am implementing a variation of my
DTSwift app through an Object Pascal based FastAGI scripting server now.
http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm
The newer version just uses the System() AGI command to build the file
to play through the shell. I'd be open to any suggestions for a more
efficient way of doing it.
Sorry for not answering faster - was busy shoveling snow.
Once you've got app_swift installed, saying something
should be pretty close to:
AGI.Exec('Swift','This is something to say.');
And since it doesn't render to a file first you'll probably experience
less of a delay to say something.
Yeah, that is what mine does. I'll take a look at loopfree code and see
how it's streamed.
--
Warm Regards,
Lee
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