Re: [asterisk-users] Fwd: cisco phone 7911
Make sure this is in your xml config: ntps ntp nameip.of.ntp.server/name ntpModeUnicast/ntpMode /ntp /ntps On Thu, Jul 2, 2009 at 10:27 AM, mahboob zamanmahboob.za...@ssl.com.bd wrote: -- Forwarded message -- From: mahboob zaman mahboob.za...@ssl.com.bd Date: Tue, Jun 30, 2009 at 8:42 AM Subject: cisco phone 7911 To: asterisk-users@lists.digium.com Hellow, I have cisco 7911 and 7906 worked with asterisk server. But i can not set the time and date for these phones. can any one tell me how can i set the time and date for these phone. Thanks mahboob -- Mahboob Zaman System Engr Systems Services Limited Cell: +8801712280308 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
I think this is what you want: http://bugs.digium.com/view.php?id=8824
On Fri, Oct 3, 2008 at 4:21 AM, Olivier [EMAIL PROTECTED] wrote:
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include this kind of
line :
To: sip [EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like this
instead ?
To: John Doe sip [EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED];user=phone
Regards
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[asterisk-users] Remote Support
Does anyone have any suggestions on what to use to monitor a vendor doing remote support? On the windows side things are typically done via screen sharing ( gotoassist.com, bomgar or similar) so at least you can see what the other end is doing. In working with linux (especially hardware vendors for asterisk) they want ssh root access, but I'm nervous about giving someone free range to a box without any type of monitoring. What does everyone else do? (besides not give them access). Looking at something like screen sharing or recording, perhaps keystroke logging. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ?
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[asterisk-users] Echo on PRI even with H/W echo cancel
Hello, I have a PRI coming into a Digium TE122B with hardware echo cancel, but we are still experiencing echo on the first 10 seconds of a call. Is there anything that can be done about this? I have tried contacting digium support, but have not heard back from them (placed a support incident about a week ago). I see on digiums website that some of their card have a VPMOCT128 Octasic echo cancel, but the TE122B comes with digiums VPMADT032 echo canceler. Is the octastic echo cancel better? Should I look into a card with Octastic echo canceler? I see Sangoma has a single port with T1 with Octastic echo canceler or would have to move up to the dual span card to get the Octastic echo cancel on the digium card. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error Counters on PRI Circuit
Is there a way to see error counts on the T1 of a PRI? Hooked up to asterisk via a digium TE122. Looking for something to make sure I'm not getting any CRC, framing or other errors on the T1. Using asterisk 1.4.19 and zaptel 1.4.10 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rebooting newer cisco phones
Does anyone have a solution for remotely getting the newer cisco phones (7941, 7961, 7970, etc ) to reread their configs (or even rebooting). I am running SIP firmware connected to asterisk. Check-sync doesn't seem to work anymore, I can't login to the phones as root because I am given a challenge: random digitspassword: prompt. occasionally I need to make changes to phones at remote locations, only solution I have now is rebooting the POE switch. Kinda an overkill. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog DID
It seems the standard for Analog DID (at least around here) is wink start, does the Rhino cards work with this or do I need to have the telco immediately send the DTMF tones? On Wed, Feb 13, 2008 at 12:33 PM, James Finstrom [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rhino's Analog cards support analog DID. no need for all the extra stuff You will want to get an R8FXX with fxs modules that will give you channels in sets of 2. ADID has not really taken off in the OS telephony market I think due to a lack of understanding people stay with the proprietary phone systems that pimp this feature. Okay so I will take the lead and pimp it for asterisk. With Rhino Analog cards you CAN do ADID with no extra equipment. However if you want to spend the money we can go the other route :) darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price. Darren Wiebe [EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DID On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks? What is an analog DID trunk? You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 This is for providing plenty of analog extensions (phones). Is that what you're after? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b327c1163231152562594! -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b327c1163231152562594! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFHsynrdloC7YyaIOoRAuKhAJiCRxUX+E7rzt6/A5nyAjXdO5yaAJ4/HoKB Gxd6H7YOdzXfygVuBygzAw== =51QY -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft Office Communications Server
We are interested in getting something working also, let me know how it goes. We are currently using LCS 2005 for IM, the only thing we want to add is the ability to update the On the Phone status in communicator. I have a test system on 1.6, but so far have been unable to update the presence information, would be interested if anyone has been able to do it will Office communications server. On Tue, Mar 11, 2008 at 10:37 AM, Razza [EMAIL PROTECTED] wrote: On 10/03/2008, Matt Riddell [EMAIL PROTECTED] wrote: Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Analog DID
Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
Looks like it is part of the 1.6 Beta. From the Change Log: 2008-01-18 22:04 + [r99080-99085] Russell Bryant [EMAIL PROTECTED] * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) On Feb 13, 2008 4:21 PM, Razza [EMAIL PROTECTED] wrote: I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk voice quality tuning
I will try to answer it this way: G.711 is toll quality voice, if everything is functioning properly should be almost identical to a regular phone call. You will need to do trouble shooting to (in the words drilled into me by an old boss): isolate, identify and quantify the issue. I would start by setting up a record/playback extension, call it from PSTN and call it from the SIP phones, see where the noise is being introduced, from there could be hundreds of different things (LAN congestion, interrupt sharing on the PSTN card, bad wiring, faulty switch, etc). So to answer your question there isn't a parameter that says Noise=Yes/No, you need to: isolate, identify and quantify the noise. On 9/14/07, satish patel [EMAIL PROTECTED] wrote: I have both type of call sip-2-pstn and pstn-2 -sip but quality is not good so how to check asterisk voice quality and codec quality i am useing G.711 alaw and ulaw and it is my LAN network so is there any specific perameter or option to improve quality of voice ??? *Adrian Marsh [EMAIL PROTECTED]* wrote: Satish, Whats your network setup? Do you get bad quality on internal-asterisk calls, or only on external calls? Are you making pure IP calls (sip2sip), or are there E1/T1 cards involved? What codecs are you currently using? What devices are you using? Adrian Marsh From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of satish patel Sent: 14 September 2007 06:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk voice quality tuning Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Catch up on fall's hot new showshttp://us.rd.yahoo.com/tv/mail/tagline/falltv/evt=47093/*http://tv.yahoo.com/collections/3658+%0Aon Yahoo! TV. Watch previews, get listings, and more! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Softkey config example for Cisco 7941/7961
Anybody have this file or find any documentation on it? Thanks. On 5/10/07, Joe Pukepail [EMAIL PROTECTED] wrote: I found on the web that there is way to customize the softkeys for the 7941/7961 phones. In the SEPmac.xml there is a section called softKeyFile where you can specify an xml file for the softkeys. I couldn't find any examples of this softkey file or the format to this file. Does anyone have a call manager enviroment where they can look at this file and send me an example. Looks like it is typically it is called SKuniquie-id.xml ___ --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] Softkey config example for Cisco 7941/7961
I found on the web that there is way to customize the softkeys for the 7941/7961 phones. In the SEPmac.xml there is a section called softKeyFile where you can specify an xml file for the softkeys. I couldn't find any examples of this softkey file or the format to this file. Does anyone have a call manager enviroment where they can look at this file and send me an example. Looks like it is typically it is called SKuniquie-id.xml ___ --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] snom 360: how to make record button working ?
There was a patch to get this working, looks like it has been abandoned, though. Should give you a starting point to get it working, or perhaps a bounty would get someone interested in getting it usable and committed. http://bugs.digium.com/view.php?id=4845 On 10/4/06, Joel Hill [EMAIL PROTECTED] wrote: Hi Noro,Depending on what firmware you have this is the way to go.Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asteriskrecording function.Hope this helpsCheers,JoelAsterisk ITwww.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output,the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dual core
I believe asterisk for the most part is single threaded, you will get some advantages by having other system processes use the extra Processor/Core, but I don't think asterisk will use alot of the other CPU. On 9/25/06, Tomislav Parčina [EMAIL PROTECTED] wrote: Asterisk is very happy on dual core. It greatly reduces load. We just put a Pentium-D in poduction last week and it is working verry well. We have a Core 2 Duo on order that we should be putting in production next week. MATT---Hi Matt!Thank you for this information. Can you please tell me if you weight Asterisk, does it divide that job on both processors or it's only one that does the job? --Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148Mob.: +385(91)1212148SIP: [EMAIL PROTECTED]e-mail: tparcina#lama.hr http://www.lama.hr___--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] new in 1.4?
I seen something in the bug tracker and svn about SMDI. Not sure if it was included it 1.4 though. Would be interested if anyone knows if this will work with nortel system (option 11 in particular). On 9/22/06, Bruce Reeves [EMAIL PROTECTED] wrote: There are a couple more that I have run across.- Shared line Apperance support- Users.conf file for simple config of users and devices - follow me application and conf file- Asterisk Builtin mini-HTTP server On 9/22/06, Zoa [EMAIL PROTECTED] wrote: I was thinking the same thing when reading the press release on sineappsand writing a news article for asteriskguru. I think this covers most of it:- Generic Jitter Buffer- t.38 passthrough- Dial plan programming language (AEL v2) - Asterisk can talk to googletalk and Jabber networks- imap storage for voicemail - whisper paging- Autoconf configuration- menuselect (graphical module select tool similar to the kernel configsystem) - higher quality prompts (in English, French and Spanish). - watch outthey are restructured a little Zoa.Roy Sigurd Karlsbakk wrote: Hi all I've read through the UPGRADE.txt file, but AFAIK it does not quite discuss all the new stuff with 1.4. Neither the jitterbuffer nor the packetization patch (#5162, if that ever made it into 1.4) are mentioned. So, is there a document somewhere describing what's new in asterisk? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people- Terry Pratchett --- Roy Sigurd Karlsbakk [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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] long distance ethernet Asterisk
Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Two questions: We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA ---Brian VincentCopper Mountain Telecom [EMAIL PROTECTED] __ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview,retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethismessageandanyattachmentsfromyoursystem.Thankyou.__ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] PRI vs Digital Trunk
I don't know about what our LEC is calling a digital trunk but verizon tried to offer me something like this for a location that they couldn't offer a PRI, basically it was a just a voice T1 (24 channels),didn't havefeatures like Caller ID, setting outbound caller ID, ANI, etc. YMMV. On 7/25/06, Barry D. Hassler [EMAIL PROTECTED] wrote: Hi, can someone enlighten me as to the difference between a PRI and aDigital Trunk (other than cost)? I do understand PRI (B-channel signaling, incoming/outgoing call setup,D channel for voice/data, etc), but I'm not quite sure how that compareswith what my vendor is calling a Digital Trunk (specifically in contrast to a PRI). The PRI is about twice the cost.If this is just a channelized T1 (24 64k voice/data channels'), wouldthey each be assigned a specific phone number, or is there furtherflexibility in sending/receiving calls, callerid (receive or send), etc? Feeling ignorant hereThanks!___--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] Urgent source code changes needed
Why don't you detail what you are trying to accomplish on the list, perhaps someone will do it for free. If it is a legitimate bug you could add an incident to the bug tracker. On 7/24/06, Bart Fisher [EMAIL PROTECTED] wrote: I need someone to patch what I believe to be a simple change tochan_zap.c - I know if I attempt I'll screw it up :) Whom would you approach for doing this? - My requests have received a'blank stare' from Free Lance sites and I'm running out of time on thisinstall.If you know someone or could handle this yourself, please contact me at [EMAIL PROTECTED]ThanksBart___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Invite someone to Conference
Or you could use web-meetme, it has this feature.
On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote:
I would like to walk you through it but I have much on my plate right now that requires my attention.
I will point you in the right direction.
Look at the menu options in MeetMe, the exit menus in particular. These would allow you (admin) to press a key and exit into the dialplan.
Once in the dial plan you can decide what to do, You can manually call the invitee or enter the extension and use an AGI to create the call files and play an automated 'invite' the that person. You could also parse the
voicemail.conf file via the AGI and send them an email. Once you start with your AGI you will be able to do anything you want.
GOTCHAs:
Make sure that you do not have the MeetMe end when you 'marked' exit the conference, This would drop all the members when you leave to go 'get someone else'. These are options to MeetMe.
I am sure there is more to it but I gave you a brief overview.
If you are willing to put whatever you do back into project, I will be happy to help you along a little more.
Alex
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Rizwan HishamSent: Thursday, July 06, 2006 9:27 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Invite someone to Conference
how do i do that.let me tell u that im new to asterisk technology, so ur gona have to walk me thru the solution. i do have a background in programming so i can do whatever configuration for asterisk u want me to do. tell me if there is any help on the internet about this.
On 7/6/06, Alexander Lopez
[EMAIL PROTECTED] wrote:
Define invite.
Yelling across the office saying, Yo, Dude! Dial ${CON}, works as an invite for me!.
If what you want is an automated invite look at callout files and using creative dialplan options to Meetme App.
Snip
Hi,recently im working on using meetme application in asterisk. i have explored all the options for meetme application and i believe there is no option for inviting a person to the conference while the conference is on. is there any other way to do that?
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Re: [Asterisk-Users] Dropping incompatible voice frame
This is known issue, we fixed it by putting an answer() in the dial plan before it gets forwarded,thefix transcode_via_sln=no (detailed in the bug tracker)didn't work for me. YMMV. http://bugs.digium.com/view.php?id=4101 On 6/28/06, Kevin Savoy [EMAIL PROTECTED] wrote: Sorry if this has been posted before but I'm having an issue where I get the following on my CLI. ast_read: Dropping incompatible voice frame on Local/XX of format ulaw since our native form has changed to slin A call comes in on our main to toll free number on an ATT T1 line and is sent to phone 4000. This is our secretary's desk. If she leaves the desk she forwards the phone to one of our sister companies so that they would answer the call. This call is sent back out the ATT T1. If she answers the call and then forwards outside the building it works fine but if she forwards her phone outside the building to auto forward the call when she is away from her desk we get the above error. I have recreated this on my own phone (both hers and mine are Polycom 501's) and with a Cisco 7960. I also tried a different toll free number with the same results. I searched the internet and found four people having the same issue but none have gotten responses on how to fix it. Each time it was something similar where the call was redirected. I know the T1's are configured correctly because all other incoming and outgoing calls work fine until this error occurs. Then nothing works. I am using Asterisk 1.2.7.1 with Zaptel 1.2.5 and Libpri 1.2.2 . I have tried using both a digium Wctxxp 4 port and RedFone's Fonebridges and have gotten the same result both ways so the problem is within Asterisk itself. I also tried allow=all in sip.conf as well as specifically listing allow= slin and all other formats to no avail. Also when this happens the channel is no longer usable even though Asterisk thinks it is available. When the next call is placed it times out because that channel has been locked by the above error. The only way out is a complete reboot and reset of all systems. Not good. Any help would be greatly appreciated. If I had hair left I'd be pulling it out about now. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Single T1 card with Echo Cancellation toworkwithDell?
I am in the same situation, I have heard the hw echo can is much better, easier to configure, etc. But it seems like an overkill to use a quad span card when we will only be using 1. Anyone know if digium or sangoma will release a dual span card with hw echo can? On 6/14/06, Cory Andrews [EMAIL PROTECTED] wrote: Sangoma is NOT releasing a single T1 with echo cancellation.Cory AndrewsExecutive Vice President ++VoIPSupply.comPBXSelect.com++454 Sonwil DriveBuffalo, NY 14225voice - 800.398.VoIP X3402fax - 716.630.1548e - [EMAIL PROTECTED] m - 716.907.4059aim - B2Cory-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of shadowymSent: Wednesday, June 14, 2006 3:56 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Single T1 card with Echo Cancellation toworkwithDell?I also heard that Sangoma was planning to release a Single T1 card with HWecho can but I don't know when.My source was a VERY reliable one. -Original Message- From: Shane Burrell [mailto: [EMAIL PROTECTED]] Sent: Wednesday, June 14, 2006 11:46 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Single T1 card with Echo Cancellation to workwithDell? Don't know about the single T1 but the a104d works flawlessly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Warren Sent: Wednesday, June 14, 2006 1:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Single T1 card with Echo Cancellation to work withDell? HI all, I was on this list back in Dec-Jan but the asterisk server got pushed back in the project queue and it seems to have finally risen to the top. I am looking to deploy * running on Centos 4 on a Dell 2850. I need a single T-1 (PRI) card with HW echo cancellation.I had been told that the digium cards were having problems with Dell servers back in January and also told that Sangoma was due to have a single T1 with echo cancellation out by March at the latest (by an email from Digium themselves). I was wondering if anyone could give me a heads-up on the state of single T-1 with HW EC cards. Thanks, W ___ --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 listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Seize phone line
I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is down and someone calls 911. Is there a way to use asterisk to seize a phone line from the fax machine? I don't want to have to have an analog line that only gets used in the very rare situation with the PRI being down and someone needed to dial 911 (other incoming and outgoing calls would be routed over a private T1 to another location), but I don't want to just tap into the fax line because there is a chance that someone could be sending or receiving a fax at the same time. I found this: http://www.twacomm.com/catalog/model_LSR-1.htm on an internet search, anyone have any experience with this (or something similiar)? Would it work with asterik? On a related issue, at locations where we have 3 or 4 phone lines connected to asterisk and they are all in useand someone dials 911 we want it to disconnect one of the active calls so the 911 call can be made. Does anyone know how to do this? Would I need to use a device like the above or is there a way in software to do this? ___ --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] Seize phone line
On 4/27/06, Rich Adamson [EMAIL PROTECTED] wrote: Joe Pukepail wrote: I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is down and someone calls 911.Is there a way to use asterisk to seize a phone line from the fax machine?Multiple ways to do that. Something like the SPA3000 provides both ananalog pstn interface and fxs interface (for the fax machine), and both of those interfaces are addressable via asterisk's dialplan. Or, use thesangoma A200D card with an fxo and fxs interface and you'll get the samefunctions (but with better quality). Aren't I asking for trouble by bridging fax traffic through asterisk? I have seen many reports on the mailing list that trying to fax through asterisk is problematic (at best) (until T.38 is implemented. ). ___ --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 integration with office PBX
Best way is to have a PRI interface to your PBX, I don't have any experience with NEC, but with our nortel system this is what we did. You program your PBX to send extension 123 out the PRI, asterisk sees the call and routes it accordingly. On 3/15/06, John Padovano [EMAIL PROTECTED] wrote: Forgive me if this question has been asked/answered in another post.And let me reiterate what other users have frequently said - Asterisk is great, and I really appreciate all the work you folks have put into it. How have some of you gone about integrating Asterisk with a legacy office PBX, such that the end-user can use a regular office (digital handset) and dialing is fairly seamless ?Our end-users are accustomed to picking up their office handset and just dialing a 4 digit extension to reach another staffperson in our office. I'd like to replicate that so they can reach staff in our other (international) offices (behind the scenes, the call would route over IP). For instance, we have regular NEC handsets talking to an NEC PBX, and an analog line from the PBX to the Asterisk FXO.I already had our NEC tech set up an access code/alias, such that an end-user just dials 6 and it goes to the analog line going into Asterisk. Asterisk picks up after about 2 rings, and then the end-user is prompted to enter the destination phone number (which would be an e.g. 3 digit number corresponding to a SIP destination in the dialplan).But this means the end-user has to dial 6 and then wait for Asterisk to pick up. I'd Is there a way to have Asterisk pick up sooner, e.g. without any rings ? Ultimately, I'd like to get it to the point where the end-user doesn't have to pause at all. In other words, they could dial e.g. 6123 and their call would be appropriately routed. I realize that probably involves configuring Least Cost Routing on the NEC PBX, but that still leaves the issue of having to wait for Asterisk to pick up the line. Any help is appreciated.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Send One Touch Record to mail
As far as I know, you will need to do this yourself with some creative scripting. There was some talk on the list awhile ago to move the recording tovoicemail, but I dont' know if anyone has made a patch to do it yet. On 3/7/06, Tomislav Parčina [EMAIL PROTECTED] wrote: How can I send recordings, that I have recorded with One Touch Record, to e-mail address that is defined in voicemail.conf?Thank you for your ideas.--Tomislav Parcinatparcina#lama.hr___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] snom 320 MWI light
I had the same problem, I just set the voicemail button on the phone to dial the voicemail extension, but you will still have the problem (at least on the 360) if the user uses the Soft buttons below the display to access the voicemail. On 3/3/06, Nabeel Jafferali [EMAIL PROTECTED] wrote: I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have mailbox= [EMAIL PROTECTED] and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail.FYI Received the following from a vendor:Currently there is not a way to keep the MWI light to stay on after hittingretrieve button on the Snom.The best option at this point is to set checkmwi=1 in the general section of your sip.conf file.This will causethe light to turn back on shortly if there are un-checked messages waiting.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] What business IP phone to use
I like the specs on this, the only thing that it seems to be missing is POE. Anyone know if POE is going to be supported on the 300? Looks nice and I could see it for low use areas, but would suck for wall mounting if it can't do POE. On 2/22/06, Cory Andrews [EMAIL PROTECTED] wrote: Clint - Looks like your wish has been granted, and your love affair with Snom can continue. They are soon releasing the new Snom 300, which has most of the features your are fond of in the 360 and 320 models, and should be quite near, if not at, your $100 price point. Read up on it here - http://www.snom.com/pressinformation_details.html?tx_ttnews[tt_news]=354tx_ttnews[backPid]=33cHash=1bb97caf5cL=1 Detailed specs here - http://www.snom.com/snom300_voip_phone.html?L=1 Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY - Original Message - From: Clint Sharp To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 22, 2006 1:03 AM Subject: Re: [Asterisk-Users] What business IP phone to use It's funny this thread has been coming up, because I've been testing out phones at my office, and I just did a fairly intensive quality test on them.1) Budgetones: Don't bother for a business setting. The speaker phone is basically useless (echo problems) and the handset is horrible. If you follow the suggestion on the Wiki to drill out the handset, it improves things marginally, but not much. Users talking to you will constantly complain about you sound muffled. It's think it's a frequency response thing and not a volume thing, I think it's just getting lower than a standard 8 khz sample out of the microphone, because it's so cheap. 2) GXP-2000: Not much better than the Budgetones, but at least the firmware is still in active development. Feature-wise it's pretty cool, but poor firmware and poor handset hardware again make this a real problem for us. We lost one handset to static electricity yesterday (which was fixed by adding in a microphone from an old business set, which actually improved that phone's quality). The speakerphone is useless due to echo issues. However, 4 line appearances is pretty cool for that price of phone, and passthrough Ethernet at 100 mbs is pretty cool too. Overall, I can't recommend them, because while they sound slightly better than the budgetones, I still get many complaints about muffled calls. 3) Polycom: Of the 4 phone brands we're actively using (not including the Wifi phone which rarely gets used), this was the best until I got the Snom in today. The handset is of good quality. I have an IP 301, but if the cheapest phone is this good, I'd definitely get a 501 or 601 (and am considering ordering some, although I may order Snom 320s instead). Their support policies do get on my nerves, I'd like to not have to worry about what reseller I'm using, but it's a solid phone with solid features, although the menus are cumbersome and I haven't gotten MWI to work on it yet. 4) Snom 320: This is an excellent phone based off one days testing. Minimal configuration, professional looking web interface, and the best sound quality of any of the phones I tested. THe speakerphone works great, and the handset quality is outstanding, and tested the best with my callers that were listening to me through the PSTN. I haven't upgraded firmware or anything on this yet, so can't tell you there, but I can't see a compelling reason to upgrade from whatever it shipped with that this point (i'm not feature crazy, I only upgrade the firmware if basic features don't seem to be working right). Overall, stay away from the Grandstream's IMHO. The audio quality issues will drive you insane. I'm hoping someone will come out with a sub-$100 phone that drops some features but fixes what should be the cheapest part of the phone to manufacture, since they've been the same for nearly 50 years, the handset. Clint ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] [Announce] Web-MeetMe v2.0.0
I'm getting the error on the bottom of pages, I'm running this in tandem with 1.4, so not sure if this is an issue, but 1.4 still works (using the same user, password and database as version 2). Warning: mysql_pconnect(): Access denied for user: '[EMAIL PROTECTED]' (Using password: YES) in /var/www/html/Web-MeetMe/lib/DB-modules/phplib_mysql.php on line 75Database error: pconnect(localhost, root, $Password) failed.MySQL Error: () On 1/12/06, Dan Austin [EMAIL PROTECTED] wrote: [New Features] 1.Added focus and tab-order to all input fields 2.Dynamic generation of date/month/year listboxes a.It is no longer possible to schedule an invaliddate. 3.Added 'Extend' and 'End Now' buttons to the monitor page. 4.Invite button on the monitor page.This greatly simplifies the process of adding callers to a conference. The ./lib/defines file includes definitions for the prefered channel and context*** 5.Call history report.Support for this feature requires the php script ./lib/cbEnd.php be running at all times.This also requires a new table in the meetme database if you're upgrading from an earlier release.***[Location] http://www.fitawi.com/Asterisk[Files] Web-MeetMe_v2.0.0.tgz (required) app_cbmysql.c (required) cbmysql.conf (required) cb-extensions.conf (suggested) README (suggested)[Installation] See the README[Features] 1. Schedule new conferences a. Control start and end timesb. Set conference pin # i. Generate one if the requester leaves it blank ii. Identify pin # conflicts (another conference with the same pin is scheduled at the same time)c. Set Admin and User passwords i. Generate a user password if an Admin pw is set but the User pw is blankd. Weekly recurring conferences with the same settings e. Select MeetMe flags per conference for Admins and Users 2. Email the details for a successfully scheduled conference 3. Separate views for Current, Past and Future conferences 4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activitya. Mute/Kick participants 6. Optional authenticationa. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUSsupport could be easily added 7. Users can only monitor, update or delete their conferences 8. Verified administrators can monitor, update or delete any conferences. 9. Updated to Asterisk 1.2.0a. Changes to the Manager interface may have causedsupport for 1.0.X to slip, I cannot test that)Thanks and enjoy,Dan ***Beta testers and anyone who downloaded v2.0.0 before todayThe only changes from the beta was a cosmetic change to work withnon-IE browsers and a couple of installation hints.I onlyreceived feedback from one tester, so it appears the package is ready to go.***Developer help/guidence request***The PHP script to monitor conference endtime andup date the CDR is fragile.If Asterisk is shutdown for more than 30 seconds, the script exits. I'd like to make it more resilent.If any PHPexperts can make suggests on how to improve thescript it would be appreciated___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] [Announce] Web-MeetMe v2.0.0
Ok, I got through that error, after recompiling with app_cbmysql asterisk doesn't want to start up. I renamed the app_cbmysql.so file and it came up ok.. Anyone have any advise? [app_cbmysql.so]Feb 16 13:08:17 WARNING[21558]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_store_resultFeb 16 13:08:17 WARNING[21558]: loader.c:554 load_modules: Loading module app_cbmysql.so failed! On 2/16/06, Sean Cook [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1This shouldn't make any difference... check your defines.php and makesure you have the correct username/password...define (USER, root);define (PASS, some_really_strong_secret);SeanJoe Pukepail wrote: I'm getting the error on the bottom of pages, I'm running this in tandem with 1.4, so not sure if this is an issue, but 1.4 still works (using the same user, password and database as version 2). *Warning*: mysql_pconnect(): Access denied for user: '[EMAIL PROTECTED]' (Using password: YES) in */var/www/html/Web-MeetMe/lib/DB-modules/phplib_mysql.php* on line *75* *Database error:* pconnect(localhost, root, $Password) failed. *MySQL Error*: () On 1/12/06, *Dan Austin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: [New Features] 1. Added focus and tab-order to all input fields 2. Dynamic generation of date/month/year listboxes a. It is no longer possible to schedule an invalid date. 3. Added 'Extend' and 'End Now' buttons to the monitor page. 4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference. The ./lib/defines file includes definitions for the prefered channel and context *** 5. Call history report. Support for this feature requires the php script ./lib/cbEnd.php be running at all times. This also requires a new table in the meetme database if you're upgrading from an earlier release. *** [Location] http://www.fitawi.com/Asterisk [Files] Web-MeetMe_v2.0.0.tgz (required) app_cbmysql.c (required) cbmysql.conf (required) cb-extensions.conf (suggested) README (suggested) [Installation] See the README [Features] 1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blank ii. Identify pin # conflicts (another conference with the same pin is scheduled at the same time) c. Set Admin and User passwords i. Generate a user password if an Admin pw is set but the User pw is blank d. Weekly recurring conferences with the same settings e. Select MeetMe flags per conference for Admins and Users 2. Email the details for a successfully scheduled conference 3. Separate views for Current, Past and Future conferences 4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activity a. Mute/Kick participants 6. Optional authentication a. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUS support could be easily added 7. Users can only monitor, update or delete their conferences 8. Verified administrators can monitor, update or delete any conferences. 9. Updated to Asterisk 1.2.0 a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that) Thanks and enjoy, Dan ***Beta testers and anyone who downloaded v2.0.0 before today The only changes from the beta was a cosmetic change to work with non-IE browsers and a couple of installation hints. I only received feedback from one tester, so it appears the package is ready to go. ***Developer help/guidence request*** The PHP script to monitor conference endtime and up date the CDR is fragile. If Asterisk is shut down for more than 30 seconds, the script exits. I'd like to make it more resilent. If any PHP experts can make suggests on how to improve the script it would be appreciated ___ --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 -- ___ --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 -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (MingW32)Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.orgiD8DBQFD9LIEy9wPyZpnL2URAhj6AJ9wMOKeTfiSNEGnI5x9f/MDIXbMsgCfYvot br7MFSRGc/mw981c1bLwrnA==jtHa-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options
Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP
What you are doing is changing the priority of packets that you are sending to the internet, you'll have to throttle the bandwidth for incoming packets (or better yet, have sprint do it on their router). What you are doing will help if you are getting bad calls when someine is uploading something to the internet, but not downloading. On 2/14/06, Philip Edelbrock [EMAIL PROTECTED] wrote: David Choo wrote: Hi, Consider doing rate limiting / bandwidth reservation. It worked heaps of wonders for mine!That's good to hear.BTW- Am I doing this right?Here are the releventchunks of my config on my router:!!class-map Platinummatch access-group 101! !policy-map IPCOSclass Platinum bandwidth percent 35!access-list 101 permit udp any any range 16384 32768access-list 101 permit udp any any range 6050 6060!interface Serial0/0service-policy output IPCOS service-module t1 timeslots 1-24!Phil___--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] 911 and ISDN PRI
I've talked to the carrier (verizon), what they said is that the call is not leavingmy phone equipment. I tried to tell him that I'm getting an error back from his system, but he insists that the channel never comes up. Their answers was talk to your telco vendor, its on their end. So I guess I'm pretty much SOL when it comes to using 911 with the PRI. Below is the debug, they wanted me to try all the DID numbers to see if it worked on any of them (40 numbers) and the billing number, wouldn't work with any of them. -- Executing SetCallerID(IAX2/sycam-16384, 8157548823) in new stack -- Executing Dial(IAX2/sycam-16384, Zap/g2/911) in new stack-- Making new call for cr 33385 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 617/0x269) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 41 80 38 31 35 37 35 34 38 38 32 33] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '8157548823' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 617/0x269) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ]-- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/25-1' On 2/8/06, Watkins, Bradley [EMAIL PROTECTED] wrote: It looks like the outbound caller ID is not being set properly. Most of the carriers that I've dealt with will act exactly as you said if you do not set it to what is expected at the 911 center. In particular: Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] Your user number being sent is just the caller ID of the SIP channel. Regards, - Brad -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 3:26 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 911 and ISDN PRI I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of Invalid Number format, I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that invalid number format is the calling number or the number I'm calling. I'll let the list know the result. I would encourage everyone to test their 911 functionality (especially if you have a PRI), I almost didn't check it.The PRI is up and 411 works, so I almost assumed that 911 would work. Make sure you call the police station first to make sure they are not swamped by real emergency calls and let them know you are testing . -- Executing NoOp(SIP/3251-7316, 3251) in new stack -- Executing Dial(SIP/3251-7316, Zap/g2/911) in new stack-- Making new call for cr 33144 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 376/0x178) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 09 b1 52 65 63 70 74 69 6f 6e] Display (len= 9) Charset: 31 [ Recption ] [6c 06 41 80 33 32 35 31] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user
[Asterisk-Users] 911 and ISDN PRI
Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? ___ --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] 911 and ISDN PRI
I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of Invalid Number format, I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that invalid number format is the calling number or the number I'm calling. I'll let the list know the result. I would encourage everyone to test their 911 functionality (especially if you have a PRI), I almost didn't check it.The PRI is up and 411 works, so I almost assumed that 911 would work. Make sure you call the police station first to make sure they are not swamped by real emergency calls and let them know you are testing . -- Executing NoOp(SIP/3251-7316, 3251) in new stack -- Executing Dial(SIP/3251-7316, Zap/g2/911) in new stack-- Making new call for cr 33144 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=44 Call Ref: len= 2 (reference 376/0x178) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 09 b1 52 65 63 70 74 69 6f 6e] Display (len= 9) Charset: 31 [ Recption ] [6c 06 41 80 33 32 35 31] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3251' ] [70 04 a1 39 31 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ] -- Called g2/911 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 376/0x178) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 9c] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ]-- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote: I dunno about your provider but I know that 2 of my 3 MCI PRI circuitshave no 911 abilities. MCI tells me this is becasue I have no local dialing plan on them.Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.comMichael Collins wrote: 911 **should** work on a PRI.If you are getting a hangup and you don't see a valid hangupcause, it might be best to get your carrier on the line and have them monitor the circuit while you dial 911.They might be able to tell you what the problem is. -MC *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] *On Behalf Of *Joe Pukepail *Sent:* Tuesday, February 07, 2006 10:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] 911 and ISDN PRI Does asterisk support this?I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup.Does 911 normally work over a PRI line?Anything special I have to setup in asterisk? ___ --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 listTo 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] Blocked Callerid
Do they have an 800 number? If so perhaps their 800 number provider is doing it via DTMF. Search around on the internet, I believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps reversed). On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code wasunproductive... Any ideas? PaulH ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] New version of snom soft phone
I see the announcement for the snom 300 on the website, any idea of the street price for that phone? On 2/1/06, Christian Stredicke [EMAIL PROTECTED] wrote: Hey we have made a new version of our soft phone which fixes animportant bug in the SRTP SSRC part... It is compatible with our latest version 5.3 of the hard phones.http://www.snom.com/download/snom360-5.3.exeEnjoy, Christian___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Networking voicemail
Is there a way to network the asterisk voicemail system between offices? We would like the ability to forward a voicemail to another user at a branch office (each office would have their own asterisk server connected via iax), I guess I would prefer not to use one central server for voicemail for redudancy and disaster recovery, but I guess I'll have toif others have gone this way and I don't have any other choice. ___ --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] Snom 360 Message Waiting indicator
I'm having problems with the Message waiting indicator on my Snom 360 that I'm using for testing. I got the button and message waiting indicator working, the problem is : when I hit the voicemail button (or use the menu on the display to access voicemail) it seems to clear the message waiting indicator on the phone. So even if I don't go in and delete any messages it clears the light and seems like the light isn't updated until I get another voicemail. Anyone else run into this? Anyone else get it working properly? ___ --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] Sip phone with Bluetooth - does it exist?
Anyone know if a Sip phone with bluetooth for a wireless headset exists? If so does anyone have any recommendations? Or maybe a Wifi/Sip headset? ___ --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 1.2.2 Released!
Perhaps I'm an idiot, but I looked through the readme and changelog but can't figure out what asterisk-netsec is all about? Anybody figure it out? On 1/18/06, Mr. James W. Laferriere [EMAIL PROTECTED] wrote: Hello Announce All ,On Wed, 18 Jan 2006, Asterisk Development Team wrote: Greetings everyone! The 1.2.2 versions of Asterisk, Zaptel, and Libpri have now been released. The source tarballs are available for download on ftp.digium.com. For details about what has changed, see the ChangeLog for Asterisk, Zaptel, or Libpri. We are also excited to announce the release of a special version of Asterisk 1.2.2, called Asterisk-NetSec. It includes some very exciting features not available in any other version of Asterisk, or even any other related product! Please view the appropriate README and ChangeLog for more details. Asterisk-addons and Asterisk-sounds will remain at version 1.2.1. Previously, all packages were updated to reflect a matching version number, even if no changes have been made. From now on, releases will only be made when changes have actually been made. Even if version numbers do not match, it is safe to use all of these releases together, as long as all of them are the latest version available. Thank you! The one thing that annoys me most is a announcment with out a url: to what it is announcing .Can we please correct this ?Tia ,JimLps: Not that I can't find it , but ... is just courtisy to others .--+--+| James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. |Give me Linux|| [EMAIL PROTECTED] | Billings , MT. 59105 | onlyonAXP || http://www.asteriskhelpdesk.com/cgi-bin/astlance/r.cgi?babydr |+--+___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] New Mail Message Waiting
My snom phone shows how many messages, it is a snom 360, there is the MWI light and on the lcd display it will say 5 New and 5 old messages. On 1/4/06, Aaron Daniel [EMAIL PROTECTED] wrote: If the voicemail is stored locally on the server that the phone isregistering to, the phone should automatically turn MWI on. AaronForrest Beck wrote: I am looking for a way to notify my users that there is a message waiting in voicemail.Just a simple text on the phone that says there is a new message in the mailbox.Any ideas???I sniffed around VoiceMail.conf samples and didn't see anything. BTW.This is a SIP 7912G Phone. 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 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] snom Firmware 5.0.
I agree, I liked the old ringtone 2 also (just abeep), I use it at my desk, If I'm there I can pick it up and it wasn't obnoxious enough to disturb others. Please email it to me if you get it in the format needed. On 1/2/06, Remco Barende [EMAIL PROTECTED] wrote: Hi Usman,Thanks for the explanation.Could you make the old Ringer 2 available in some form, preferable already in the format the phone understands?That would solve the problem too :)Thanks!!RemcoOn Mon, 2 Jan 2006, Usman Tahir wrote: Hi Remco, Old Ringer 2 is not there on the phone anymore, perhaps you can use another ring melody or a suitable custom melody: The wav file itself should be a PCM encoded 8 KHz file at 16bit mono. The time for loading the file should not be longer then 3 seconds ! And the size should be below 250KB. To create this format from mp3: /usr/bin/mpg123 -m -r 8000 -w - -n 190 -q test.mp3 test.wav To convert an existing WAV file: sox GENERIC.wav -c 1 -r 8000 -w SNOM.wav * The -c 1 flag makes the output mono. * The -r 8000 flag makes the output a 8kHz sample. * The -w flag uses 16 bits (word) per sample. Regards, Usman. - Usman Tahir snom technology AG Gradestraße 46 D-12347 Berlin. Tel: +49 30 398330 Fax: +49 30 39833111 [EMAIL PROTECTED] www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet. - -Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED] ] Sent: Monday, January 02, 2006 2:29 PM To: Usman Tahir Cc: Asterisk Users List Subject: Re: [Asterisk-Users] snom Firmware 5.0. Thanks for the new firmware, finally some of the features are becoming available that make the phone more usable with Asterisk. One question though, ringer tone #2 on the Snom 360 firmware has been replaced? How can I get the old ringtone back? I was using the ringtone on phones in locations like meeting rooms. The ringtone wasn't intrusive at all, yet well audible. Now when a phone rings everybody is disturbed with a loud noise. Thanks! Remco On Thu, 22 Dec 2005, Usman Tahir wrote: Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, - Usman Tahir snom technology AG www.snom.com -___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] PRI: This number has been disconnected
I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of Do-De-Dah The number of have reached is notin service fastbusy. PRI Debug below. -- Executing Dial(IAX2/sycam-16385, Zap/g2/8157872800) in new stack-- Making new call for cr 32816 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=46 Call Ref: len= 2 (reference 48/0x30) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 41 80 38 31 35 37 35 34 38 38 32 33] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '8157548823' ] [70 0b a1 38 31 35 37 38 37 32 38 30 30] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) '8157872800' ] -- Called g2/8157872800 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 48/0x30) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ]-- Processing IE 24 (cs0, Channel Identification) -- Zap/25-1 is proceeding passing it to IAX2/sycam-16385 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 48/0x30) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]-- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup requestNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=18 Call Ref: len= 2 (reference 48/0x30) (Originator) Message type: RELEASE (77) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [7e 07 04 58 0b 2d 08 31 35] User-User Information (len= 9) [ 04 58 0b 2d 08 31 35 ] -- Hungup 'Zap/25-1' == No one is available to answer at this time (1:0/0/0) -- Executing PlayTones(IAX2/sycam-16385, congestion) in new stack -- Executing Congestion(IAX2/sycam-16385, ) in new stack == Spawn extension (pri, 7872800, 8) exited non-zero on 'IAX2/sycam-16385' -- Hungup 'IAX2/sycam-16385' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 48/0x30) (Terminator) Message type: RELEASE COMPLETE (90)NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null On 12/29/05, Adam Goryachev [EMAIL PROTECTED] wrote: On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote: I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think the problem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of the call, but Asterisk is hanging up the channel instead of connecting it to let hear the audio message. There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html Is anyone experiencing the same behavior?Sounds like the difference between doing inband signalling or out ofband signalling. I think by default, a PRI uses out of band signalling, ie, it just sends a message saying this number if un reachable soasterisk just hangs up and plays the local congestion dialplan.What you need to do is use inband signalling, so that asterisk won't hangup, and instead will pass the audio from the telco through.See /etc/asterisk/zapata.conf:; PRI Out of band indications.; Enable this to report Busy and Congestion on a PRI using out-of-band; notification. Inband indication, as used by Asterisk doesn't seem to work;
[Asterisk-Users] Allison on Free 411
I heard on the radio about 1-800-FREE411andtried it out, Iwas very suprised to hear allisons' voicefor the digits. Not sure if theyare using asterisk for the backend on this or not. Try it out its Free! http://www.snopes.com/inboxer/nothing/free411.asp (not afflicated with it in any way). ___ --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] PRI: This number has been disconnected
I am using T1 Signaling and seeing the same problems (I think), so I don't think its just E1. On 12/29/05, Javier Ergas [EMAIL PROTECTED] wrote: I have tried both inband and outofband too unsuccessfully. I think the priindication parameter says how Asterisk reports Busy and Congestion to the PSTN, not the other way around. In the Asterisk config sirrix.conf ( http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf) there is a providetones parameter, witch I think handles the way that interface receives the signalization from the PSTN, but I think it won't work for zaptel/Zapata. Today I tried Asterisk 1.2 in another Telco and I experienced the same behavior. I'm starting to think this is a bug in the Asterisk E1 signalization. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] En nombre de Joe PukepailEnviado el: Jueves, 29 de Diciembre de 2005 15:22 Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] PRI: This number has been disconnected I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of Do-De-Dah The number of have reached is notin service fastbusy. PRI Debug below. … … On 12/29/05, Adam Goryachev [EMAIL PROTECTED] wrote: On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote: I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think the problem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of the call, but Asterisk is hanging up the channel instead of connecting it to let hear the audio message. There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html Is anyone experiencing the same behavior?Sounds like the difference between doing inband signalling or out of band signalling. I think by default, a PRI uses out of band signalling, ie, it just sends a message saying this number if un reachable soasterisk just hangs up and plays the local congestion dialplan. What you need to do is use inband signalling, so that asterisk won't hangup, and instead will pass the audio from the telco through.See /etc/asterisk/zapata.conf:; PRI Out of band indications.; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work; outofband:Signal Busy/Congestion out of band withRELEASE/DISCONNECT; inband: Signal Busy/Congestion using in-band tones priindication = outofbandRegards,Adam___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] PRI: This number has been disconnected
I am having a similar problem, I thought it was because the PRI card is in another server that I connect to via IAX from my server, but we are seeing the same problem, ie getting a hangup instead of unavailable when calling a number that is not in service. I'm using T1 and Asterisk 1.21 On 12/28/05, Javier Ergas [EMAIL PROTECTED] wrote: I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think theproblem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1'== No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of thecall, but Asterisk is hanging up the channel instead of connecting it to lethear the audio message.There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.htmlIs anyone experiencing the same behavior?-Mensaje original- De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] En nombre de Francesco Peeters (Asterisk)Enviado el: Martes, 27 de Diciembre de 2005 20:09Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] PRI: This number has been disconnectedOn Tue, December 27, 2005 23:37, Javier Ergas said: Hi, I'm running [EMAIL PROTECTED] 1.5 with TE110P E1 PRI in Chile. When calling an invalid number using, I expect to hear: We're sorry you have reached a number which has been disconnected ... And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone.When I dial that same number trough the T1 / PRI interface however, I only hear the allison7/all-circuits-busy-now message. There was another issue like this in an old post (http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html ) but I think it isn't the same.SNIPI believe this has to do with the AMP macro's being used in [EMAIL PROTECTED] I amseeing similar things.For instance: One issue I have is that when a route has multiple trunks, and the first trunk after a while returns with 'NOANSWER', it merrilycontinues to the next trunk, which is not quite the behavior I'd expect.Especially as the primary trunk (IAX/VoipBuster) is *much* cheaper (ie free) as compared to the second trunk (Zap/g1), but the switch is madewithout any message. This could mean that you might be talking to someoneon a different trunk, and instead of a free call, be paying normal fees. This could become expensive if you're calling the USA from Europe!...I am currently looking in to ways to enhance those macro's to respond morereliably, as well as return more useful information (busy tone on busy and no-answer, number disconnected info, etc.) when needed.If I do get to a satifactory set of macro's, I will put them up on theWiki and let the list know... (I'm just starting on doing manualconfiguring, so it will be a tough job to crack, but also a learning experience...)--F PeetersPIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Nortel Meridian Option81C to TE405P
You can't use a ethernet crossover cable, make sure you are using a T1 crossover cable. (you will definately need to use a T1 crossover cable). I'm running a Nortel Option 11 and Asterisk connected in this manner. On 12/8/05, Steve Totaro [EMAIL PROTECTED] wrote: He said that he is using a crossover but for some reason I think thecrossover may be the problem.Try making a new one.Cross pin one with four and two with five.Also try a straight through cable.Yourconfigs look fine on the asterisk side although I am not real cluefullon the Meridian.One question, was the Meridian ever hooked up to the PSTN? Thanks,Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX tomy Asterisk 1.20 server. We are using the TE405P card with onecrossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If wecannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks forany help in advance.zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K 7ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 8901 TCPP NO PII NO TARG 01 CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 1 73 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DTN WTA LPR APN THFD HKD P10 VNL TKID DATE 5 DEC 2005 Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Exit Voicemail
Is there a way to have control go back to the dialplan after a call gets tovoicemail? I'm looking to implement findme and campon, but I wantthe options to be hidden, so if someone calling got a voicemail they could key in *1 (or whatever) and it would go back to the dialplan so I can implement finemein the dial plan. The same with campon, if you got a busy voicemail you could key in *2 (or whatever) and it would take them to the piece of the dialplan where it would wait for person to get off the phone. I realize I could do this by having the user key in another option (Hit 1 to leave a voicemail, hit 2 to findme) but would prefer not to, users could record this as part of their voicemail message if they want the public to know about the findme and camping on a busy extension. ___ --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] DISA function
I tried to use DISA 1.2 with regular asterisk (not [EMAIL PROTECTED]), and had problems with it (losing the last digit or occasionally other digits), YMMV. On 12/4/05, Richard Smith [EMAIL PROTECTED] wrote: Hi all, I was wondering whether the DISA function on the latest asterisk 1.2 stable release actually works better than the other prior releases. Basically the [EMAIL PROTECTED] version 2.0 BETA 4 I'm using does not recognise the DTMF tones all the time and sometime when it does, it disconnects. Cheers, Richard.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Looking for advice on cell carrier's default Un avaliable message
Look into the findme feature, this will require the person receiving the callto push a buttonhit 1 to accept this call before a callgets transfered to a cell phone (or home phone for that matter), if nobody hits 1 it continuesin the dialplan, this will prevent calls from being transfered to cell phone voicemail or the caller getting the unavailable message from the cell phone carrier. On 12/5/05, Colin Anderson [EMAIL PROTECTED] wrote: In our dialplan, we use centralized voicemail for SIP, IAX and cell phones.This means, if a caller calls a user's DID, it tries his SIP/IAX extension, then if he doesn't answer there, it tries his cell, then it goes to ComedianMail.Everything works 100%, except when the user shuts his cell phone off. Whenthat happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played.Asterisk detects this as the call being answered and completes the call.However, this is undesirable behavior. We want it to go to Comedian mail instead. Note that this is contrary to what the carrier said would happen.The carrier indicated to us that it would just ring and ring and ringforever, which is what we want. Now they are saying: too bad, this is the way it works, deal with itIn order to have the desired behavior, there are three options:1. Carrier makes it ring forever (not gonna happen)2. I set the call forward/Unavailable on the cell to a DID that points to Comedian Mail and do some Caller ID stuff to make it go to the rightmailbox. This isn't practical from a management standpoint, it would betroublesome and error prone to maintain3. When the cell is off, the carrier's Unavailable message plays right away, within 2 seconds of the call being dialed. So, somehow magically modify thedialplan so that if a cell is answered within 2 seconds, go to ComedianMail.Of these options, 3) would provide the optimum workaround, but I don't think it's possible to express this in an Asterisk dialplan.Anyone have any advice or dialplan magic on how to do 3) ? ?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] echo canceling algorithm
I have been wondering about echo canceling, it seems to be one of the major problems people have with asterisk. I've gotten it to acceptable levels (using mark2 aggressive), but everything I've read indicates that the echo canceling software isn't very effective. My question would be, what do we need to get an effective echo canceler (in asterisk software)? Is it patent issues? No experience (I know I don't know anything about how to write a echo can algorithm) or just getting the right people interested in writing one ($)? With digium offering hardware echo can, I can only conclude that echo can can't be done effectively in software? If it is a matter of money perhaps a bunch of users can offer bounties for someone (or some company) to write an good echo canceler? With the amount of money that a hardware echo canceling card costs (+$1000 per T1/E1) if half of this were spent on a fund for software echo cancel it would seem we could do it (if it is even possible using todays technology??). I don't mean this as critical of the developers who have done so much, just an honest question what we (as users) can do to help improve the product. On 12/2/05, Patrick Fortin [EMAIL PROTECTED] wrote: HiJust wandering what solution worked to eliminate echo on your setup.I am trying every solutions I can find on the wiki and none is working perfectly.We have asterisk 1.2.03 x digium TDM400P30 Snom320 + 5 Snom360For now the best setup I have is using Mark2 Echo cancel.ThanksPatrick___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Call transfer with voicemail password
Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet. On 11/30/05, Benjamin Lenard [EMAIL PROTECTED] wrote: Hi,I'm trying to have an extension ring my SIP phone then try my cellphone.I can transfer the call fine to the cell but I want it to ask for a pin , voicemail pin, before transferring the call.This is so if my cell's voicemail answers , the call doesn't transferto it.Any ideas?Thanks,Ben___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Pros and Cons of T1/E1 cards
I haven't heard of this product before so I did some searches on the Internet, this card is $5,400 for a single span T1 card? ouch! http://www.eiconworks.com/DivaServerV-PRI_T1%20.asp On 11/25/05, David Waugh [EMAIL PROTECTED] wrote: Hi John,I'm going to have to disagree with some previous posts.The Eicon Diva Server PRI/E1/T1 cards support an E1 interface and reduce the load of the call handling, echo cancellation etc as this is all processed on board on the card, and not on the central CPU of the computer. You can use the CAPI interface of the card combined with chan_capi_cm with the card.I have not found any problems when using different kernels or different versions of asterisk.I have one setup in our test lab here at Eicon with Asterisk so it does work! You can have up to 8 Diva Server cards in once machine - including a mixture of the analog and BRI cards.The Diva Server cards in two variants - the V-Series if you only want to use them with Voice based applications and the normal All-in-one cards if you want to do fax and RAS too. If you need any more information let me know, and I will assist furtherDavid-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of JohnDaragonSent: 25 November 2005 00:46To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pros and Cons of T1/E1 cardsHi;We're looking to standardise on a single family of E1 PRI cards.I guess our options are :Digium / Zaptel / libpriSangoma/ Zaptel / Wanpipe AVM/ CAPIeIcon/ CAPIJunghanns/ BristuffCan anyone share any comparative experience of these, please ? Do theydiffer much in terms of interrupt requirement, CPU load c ? Any info gratefully received.jd--John Daragon[EMAIL PROTECTED]argv[0] limitedLambs Lawn Cottage,Staple Fitzpaine,Taunton,TA3 5SL,UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Command line
For an example here is what I setup to call out, we have a job that runs on our mainframe, when the job completes it ftps flag1.txt to ourasterisk server, the .bash program is run from the crontabat a certain time and notifiy staff if the job is not complete at that time. It will keep calling (using phoneme[1-3].call until someone acknowleges the alert by pressing 1.
There might bea betterway of doing this using AGI (or another way), but this is what I came up with.
--- extensions.conf ---
[phonehome]
exten = t,1,Playback(connection-timed-out)exten = t,2,Playback(goodbye)exten = t,3,Hangup()
exten = ,1,Playback(something-terribly-wrong)exten = ,2,Playback(to-confirm-wakeup)exten = ,3,Playback(press)exten = ,4,SayDigits(1)exten = ,5,Read(ACCEPTCALL|1-yes-2-no|1) ; (repeatoptions)
exten = ,6,GotoIf($[${ACCEPTCALL} = ] ?,9)exten = ,7,GotoIf($[${ACCEPTCALL} = 2] ?,9)exten = ,8,GotoIf($[${ACCEPTCALL} = 1] ?,96:,1)exten = ,9,Playback(auth-thankyou)
exten = ,10,Playback(goodbye)exten = ,11,Hangup()
exten = ,96,System(/bin/touch /phonehome/ack.flag)exten = ,97,Playback(auth-thankyou)exten = ,98,Playback(goodbye)exten = ,99,Hangup()
--- phoneme1.call ---
Channel: Zap/g2/5551212MaxRetries: 0RetryTime: 60WaitTime: 30Context: phonehomeExtension: Priority: 1
--- phonehome.bash ---
if [ -e /home/mainframe/flag1.txt ] then echo job completed exit;firm -f /phonehome/ack.flagwhile truedo cp -f /phonehome/phoneme1.call /var/spool/asterisk/outgoing while [ -e /var/spool/asterisk/outgoing/phoneme1.call ]
do sleep 5 done if [ -e /phonehome/ack.flag ] then echo ack!! exit; else echo no ack fi
cp -f /phonehome/phoneme2.call /var/spool/asterisk/outgoing while [ -e /var/spool/asterisk/outgoing/phoneme2.call ] do sleep 5 done if [ -e /phonehome/ack.flag ] then echo ack!!
exit; else echo no ack fi
cp -f /phonehome/phoneme3.call /var/spool/asterisk/outgoing while [ -e /var/spool/asterisk/outgoing/phoneme3.call ] do sleep 5 done if [ -e /phonehome/ack.flag ] then echo ack!!
exit; else echo no ack fidone
On 11/25/05, Tom Rymes [EMAIL PROTECTED] wrote:
On Nov 25, 2005, at 7:33 AM, Tony Spencer wrote: Hi I'm pretty new to using Asterisk and have searched to find an
answer to my question but have failed to. I was wondering if you can use Asterisk from the command line to make it make an outgoing call and issue other commands whilst it's in the call?
Sort of like when you use Minicom with a modem connected to a serial port and send it AT commands.Thanks TonyTony,If you have a sound card installed and properly configured in your
Asterisk server, then you can plug in a microphone and headset andmake calls from the CLI using the dial command.If you want to automate having the system make phone calls, googleand search
voip-info.org for info on .call files. Basically, youcreate a file that specifies to asterisk where to call, using whichchannel, and what to do once the call is connnected. You then copythe file to /var/spool/asterisk/outgoing and the call is executed as
defined.TomTom RymesCascade Link Systemswww.cascadelinksystems.com(603) 375-1414Intelligent technology solutions for small businesses.
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Re: [Asterisk-Users] Extension Ring on Multiple Phones
Here is what we use for our helpdesk, on saturday morning we have other people fill in on the helpdesk so we ring other extensions between 8am-3pm on Saturday, otherwise it rights 392 and 6001 when people call the helpdesk (x355 or x4357) exten = 4357,1,GotoIfTime(8:00-15:00|sat|*|?default,4357,10)exten = 4357,2,Dial(Sip/392Sip/6001,20,rt)exten = 4357,3,Voicemail,u4357exten = 4357,10,Dial(Sip/392Sip/6001Sip/249Sip/458Sip/394,20,rt) exten = 4357,11,Voicemail,u4357exten = 4357,103,Voicemail,b4357 exten = 355,1,GotoIfTime(8:00-15:00|sat|*|?default,4357,10)exten = 355,2,Dial(Sip/392Sip/6001,20,rt)exten = 355,3,Voicemail,u4357exten = 355,103,Voicemail,b4357 On 11/8/05, Dave Morrow [EMAIL PROTECTED] wrote: Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] server hardware
I was wondering what the concusses is for building a server for asterisk, we are looking at installing it in about 7 locations (all within an hour of each other). I prefer Dell servers, but have seen there is some incompatibility with digium hardware. ( http://www.digium.com/index.php?menu=compatibility), anyone have any results using Dell servers? Also what are the opinions as far as redundancy, should I go full bore with dual power supplies, hardwareraid, RHEL, etc? Looking at our existing phone system (nortel and norstar), they do not have redundant power supplies and the voicemail harddrive isn't raid'ed, would definately be cheaper to go with a regular PC and Fedora (and keep a spare one on the shelf), just curious what others have done. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank
Since it doesn't look like any of the FXS cards supported by asterisk support analog DID trunks, would it work if I used a T100P connected to an adtran channel bank (atlas 550?) with an FXS card installed? Anyone ever try this configuration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID Trunk
I'm working on putting together some Ideas about using Asterisk in our environment, one of the things I want to consider is DID trunks (analog), what hardware do I need to terminate these trunks? I'm looking at the voicetronix openswitch6 or openswitch12. On the openswitch, I'd like to use some of the lines for analog sets for the breakroom, kitchen, etc where they don't need all the cool features, and the other lines for POTS/DID trunks. Also how mature is this for production environment? I envision using mostly VOIP phones, cisco 7960 or Uniden UIP200 and using the voicetronix to bring in DID trunks/POTS lines. I've read reports about echo problems, is it still an issue with asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
