[asterisk-users] CLI not showing DTMF
Hi, after I upgraded from 1.2 to 1.4.13 the CLI does not show DTMF anymore, even at high debug level. Do I need to activate that? Regards Volker -- Volker Sauer * Poststrasse 1/601 * 64293 Darmstadt * Germany E-Mail/Jabber: volker(at)volker-sauer.de * http://www.volker-sauer.de PGPKey-Fingerprint: DB26 11C7 B12E 0B27 3999 2E4F 7E35 4E4D 5DD5 D0E0 http://wwwkeys.de.pgp.net/pks/lookup?op=getsearch=0x7E354E4D5DD5D0E0 signature.asc Description: Digital 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
Re: [asterisk-users] Refrigerator Alarms
Balu Raman wrote: Omar, I am hoping that there may be some temp sensor interface that can be routed to a pc and if the temp falls out of a range, I can have this event call someone. I know what to do in asterisk to make a call. I have to do some research. may be, someone has already done a similar thing. Has to be event driven. Here's what we do - even it's not asterisk-related - temperatures are monitored/polled using Maxim/Dallas DS1820s devices. These are cheap and the size of a transistor. When/if certain thresholds are exceeded, an email is sent to our central mail-server where it is turned into an SMS. The same email could just as easily be turned into a call file and dropped into to the appropriate asterisk directory. We expect to start using SNMP traps instead of the email, but the principle is pretty much the same. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] Issues with Zaptel 1.4.5.1
Tzafrir Cohen wrote: snip /... is required to properly start zaptel. It will also run ztcfg. Otherwise users run into issues where misconfigured zaptel.conf fails loading of a module. That is a buggy behaviour. If your card is an analog one, take a look at http://bugs.digium.com/7613 and tell me what you think. Something similar for digital spans would require more information in sysfs. So do I understand this correctly? I can use this patch to alleviate the need for a userspace init.d start-up script? I just patch the zaptel module source and rebuild? Does it read /etc/zaptel.conf and thus load the zone data? I only have an x100p so this could be a useful patch for me. Alan -- The way out is open! http://www.theopensourcerer.com ___ --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] Relaying calls to another SIP extension
Hi, I am learning Asterisk for a small project. At this stage I have an AsteriskNOW system running locally. I can call SIP phone to SIP phone fine, the operator and voice mail work fine, except some stuttering (probably caused by it running in MS-VPC) What I need to figure out is... I have an automated response telephony server (Voice Media Server) available via a SIP URI like: [EMAIL PROTECTED] If I place this into the SIP phone, X-Lite the call goes through fine... but not through my Asterisk. What I need to do is to add this service 'through' my Asterisk. So when a user calls a certain extension through Asterisk it 'rings' the extension [EMAIL PROTECTED] as if it were a standard SIP phone. That is stage 1. Stage 2 is for the media server handling [EMAIL PROTECTED] to be able to forward the call onto another SIP phone, allowing it to drop the call completely and the call tromboning or bridging to happen in the Asterisk PBX and not take up 'lines' on the media server. (Eventually both SIP ends may become PSTN's). I'm having trouble deducing how to do stage 1. In asteriskNow I don't see a way to add a calling rule for this. I tried adding a service provider foo.bar.com and a calling rule to send all calls for extension 6002 to that provider, but all I get is Service unavailable. With the Asterisk-docs site down I'm finding it tough going. Thanks for any pointers you can give me. Paul Campbell This e-mail is intended solely for the addressee and is strictly confidential; if you are not the addressee please destroy the message and all copies. Any opinion or information contained in this email or its attachments that does not relate to the business of Kainos is personal to the sender and is not given by or endorsed by Kainos. Kainos is the trading name of Kainos Software Limited, registered in Northern Ireland under company number: NI19370, having its registered offices at: Kainos House, 4-6 Upper Crescent, Belfast, BT7 1NT, Northern Ireland. Registered in the UK for VAT under number: 454598802 and registered in Ireland for VAT under number: 9950340E. This email has been scanned for all known viruses by MessageLabs but is not guaranteed to be virus free; further terms and conditions may be found on our website - www.kainos.com ___ --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 incomming call huntgrroup
Try this: http://astrecipes.net/index.php?n=42 l. On Thu, 18 Oct 2007 06:43:11 +0200, satish patel [EMAIL PROTECTED] wrote: I am new in asterisk world can u shortly explian how to create queue and how to work this ? David Gomillion [EMAIL PROTECTED] wrote: On 10/17/07, satish patel [EMAIL PROTECTED] wrote: Dear all I want to configure Huntgroup for my company like i call on 1100 extention i will transfer to avalible group extention i got some document on voip-info website but this is not working for me http://www.voip-info.org/wiki/view/Asterisk+Hunting+Groups+for+incoming+calls Why not just use a queue? ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] Cisco phones with Asterisk
This have been discussed a couple of weeks ago in this list. You should find useful and detailed answers in archives. Regards ___ --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] Is anyone successfully using IMAP storage
Hi, From personal experience, Asterisk 1.4 IMAP storage seems broken and unusable. Is anyone using it successfully ? This kind of poll would be very useful to estimate is a code rewrite has a chance to disturb a running system. I we get no successful report, it would help developers to consider a code rewrite as patch instead of a new feature. So please, do not hesitate to report successful or unsuccessful or never tried use of IMAP storage, it will increase our chances to get this feature in 1.4 branch and not wait for 1.6. Cheers ___ --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] Crash related to asterisk -rx ?
On Thursday 18 October 2007 04:47:14 Jean-Denis Girard wrote: Hi list, Last Friday, an Asterisk server became unresponsive after ~8,5 months of smooth operation (~32 calls). Server did reply to pings, but no ssh, no more console login. Also Asterisk no longer took calls, but ISDNguard watchdog was still alive. Looking at the logs after reboot, I could not find anything significant, except in a file created by the following command via a cron job: date /var/log/asterisk/calls.log ; asterisk -rx show channels concise /var/log/asterisk/calls.log Two days before the crash, the calls.log file started to be filled with the Asterisk console messages. I suspect this is what caused the server crash. Anybody seen this before, is this a known problem with asterisk -rx commands? Yup, it's also a problem for me, but it haven't ever crashed server. It just makes specific remote process unresponsive. There's a patch for 1.4, but i guess it wouldn't be hard to backport it for 1.2 http://bugs.digium.com/view.php?id=10847 you might also want the one mentioned in comments: http://bugs.digium.com/view.php?id=10888 Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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] Softphone that emulates Skype API ?
There's a large number of gadgets one can buy that work with Skype through the API. One of the things I'm interested right now is the ability to properly use a mobile phone headset with a SIP/IAX softphone. Is there an softphone that emulates the Skype API? Are there legal implications in writing an softphone that emulates the Skype API? Should I just give up and buy a Siemens DECT phone that supports a bluetooth headset? -- Thanks, Cosmin Prund ___ --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] phone as control interface(was99bottlesofbeer)
Hi All, sorry if I post again this e-mail but I think the first one was lost. I don't know if this is OT but I'm working in my spare time at a small hardware project that match to what's requested below. It's a board with Input/Output capabilities and 10Mbps ethernet interface. It has Microchip software TCP/IP stack on it. Being at a very beginning stage, you can see a little preview (and hopefully play with an online prototype) at this address: http://www.auto-matica.com/index.php?id=16 I've used it in the past to open a garage door when calling a private extension with Asterisk PBX and a little perl-AGI glue. I hope could interest to someone and I'm open to any suggestion/collaborations. Thank you and bye, Marco. - Original Message - From: John Faubion [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 16, 2007 9:46 AM Subject: Re: [asterisk-users] phone as control interface(was99bottlesof beer) has anyone actually been satisfied with the performance of these powerline signalling devices ? Universal Powerline Bus is a vast improvement over the original X10. I believe the X10 devices used a 4V signal during the zero crossing point of the AC voltage to transmit 1 bit. Needless to say this made X10 slow, susceptible to line noise and not very reliable. IIRC X10 is only 75% reliable. By contrast, UPB is 20-40 times faster, it uses a higher signaling voltage so line noise isn't a big factor any more and has the advantage of controlling 250 times more devices than X10. This will help to prevent stray signals from the neighbors controller from accidentally controlling your devices. UPB is supposed to be 99.9% reliable with a latency of less than 100 milliseconds. Granted that still leaves a tenth of a percent of uncertainly. However that is without resorting to filters, couplers and the like. Granted in an existing situation there may not be a way to run more wires, but I evaluated them a while back and decided to stay away. You may want to take another look at them. Just like Asterisk has made great strides since the release of 0.7, UPB has brought the quality level way up. Of course this higher quality also has a higher price. John Anyone on the list interested on working on a project where we can create devices that work over Ethernet ? ___ --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] A linksys SPA921 behind NAT and firewall
I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 1-2. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side, coz' how would it know that UDP-traffic to SPA publicIP:5060 needs to be delivered to 192.168.x.x:5060 ? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] phone as control interface(was99bottlesofbeer)
Hello Marco, could you explain how you did the interfacing to the Asterisk PBX? does your prototype speak SIP to receive commands? Thanks l. On Thu, 18 Oct 2007 12:27:33 +0200, marcotasto [EMAIL PROTECTED] wrote: Hi All, sorry if I post again this e-mail but I think the first one was lost. I don't know if this is OT but I'm working in my spare time at a small hardware project that match to what's requested below. It's a board with Input/Output capabilities and 10Mbps ethernet interface. It has Microchip software TCP/IP stack on it. Being at a very beginning stage, you can see a little preview (and hopefully play with an online prototype) at this address: http://www.auto-matica.com/index.php?id=16 I've used it in the past to open a garage door when calling a private extension with Asterisk PBX and a little perl-AGI glue. I hope could interest to someone and I'm open to any suggestion/collaborations. Thank you and bye, Marco. - Original Message - From: John Faubion [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 16, 2007 9:46 AM Subject: Re: [asterisk-users] phone as control interface(was99bottlesof beer) has anyone actually been satisfied with the performance of these powerline signalling devices ? Universal Powerline Bus is a vast improvement over the original X10. I believe the X10 devices used a 4V signal during the zero crossing point of the AC voltage to transmit 1 bit. Needless to say this made X10 slow, susceptible to line noise and not very reliable. IIRC X10 is only 75% reliable. By contrast, UPB is 20-40 times faster, it uses a higher signaling voltage so line noise isn't a big factor any more and has the advantage of controlling 250 times more devices than X10. This will help to prevent stray signals from the neighbors controller from accidentally controlling your devices. UPB is supposed to be 99.9% reliable with a latency of less than 100 milliseconds. Granted that still leaves a tenth of a percent of uncertainly. However that is without resorting to filters, couplers and the like. Granted in an existing situation there may not be a way to run more wires, but I evaluated them a while back and decided to stay away. You may want to take another look at them. Just like Asterisk has made great strides since the release of 0.7, UPB has brought the quality level way up. Of course this higher quality also has a higher price. John Anyone on the list interested on working on a project where we can create devices that work over Ethernet ? ___ --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 -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] Blocking collect calls in Brazil
Hi list. I was googling about this subject and found a message to this list asking the same thing I want, but see no response. So, I'm reposting it in case someone has a solution for this. I'd like to know if there's some way to detect (not block) a collect call in a ISDN E1? I need this cause I have my own billing system and I need to charge my customers for collect calls they receive. Currently I have to get the PSTN billing and locate all collect calls and then enter it manually. I want to detect these calls in asterisk so I can automatically charge my customers. For Brazilians in the list, I have a Embratel E1 - ISDN. Thanks Carlos Barros ___ --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 on USB Flash?
On Thu, 18 Oct 2007 00:32:44 -0400, Brian Capouch wrote: shadowym wrote: Size/Speed/write cycles have gone way up, price has gone way down. More common than CompactFlash and no need for an adapter. So is it feasible to run an Asterisk server on something like this? With a MTBF of 1million write cycles coupled with dynamic wear management on a 4Gig USB drive, lifetime is a non-issue. Just wondering how well it works, if it works. My main server for my home and teensy business runs on a Netgear WGT634U running SVN-trunk under openWRT. The Asterisk binary sits on the machine's flash, but all the modules, prompt files, voicemail, etc., goes to the flash. It works just fine. This system doesn't get a huge amount of voicemail, so I don't know about how long it would be before the wear issue would surface (pun?). I know it works just fine. I've been doing this now for almost two years, although with various versions of embedded Linux and various versions of Asterisk. Yep, me too. I've been running Asterisk on Soekris Net4801 and H-P T5700 booting from either CF or USB key for just over two years. No problems at all for a small office. In fact, its kinda a waste to have to buy 1 GB+ USB keys when it really needs 64 MB. In fact, my Net4801 box is using a 64 MB CF card that I receyled from my first digital camera. That card must be 7 years old. Still works. Not much VM so not much wear. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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] sorta OT: Bounty for Click to Call plugin for IE
I'm in process of transitioning a number of offices to a hosted virtual pbx from Junction Networks. It's a combination of OpenSER and Asterisk. They have a nice click-to-call extension for Firefox, but I need the equivalent for IE so that it can work with our CRM system. Junction told me that they have a bounty on offer for this if someone's interested in doing the work. Would the availability of the Firefox code make it easier to do an ActiveX implementation? Any takers? Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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] BBC on Atserix
Just for fun. http://news.bbc.co.uk/1/hi/magazine/7049642.stm Dave * This email is intended solely for the use of the individual to whom it is addressed and may contain confidential and/or privileged material. Any views or opinions presented are solely those of the author and do not necessarily represent those of AGCO. If you are not the intended recipient, be advised that you have received this email in error and that any use, dissemination, forwarding, printing or copying of this email is strictly prohibited. Neither AGCO nor the sender accepts any responsibility for viruses and it is your responsibility to scan and virus check the e-mail and its attachment(s) (if any). * AGCO Limited, a limited company, registered in England (registered no.509133) with its registered office at Abbey Park Stoneleigh, Kenilworth CV8 2TQ, England. ___ --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] Refrigerator Alarms
Kevin, What kind of device are you using on the fridge ? Dovid - Original Message - From: Kevin Withnall To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, October 18, 2007 12:04 AM Subject: Re: [asterisk-users] Refrigerator Alarms We use similar things here for issues like our generator battery voltage monitoring. We just have a relay going into our alarm system and as asterisk monitors our alarms it initiates emails or calls out. The alarm system is also linked into a seperate SMS unit for emergency backup so we also get SMS when any alarm goes off. My basic alarmreceiver scripts are available at http://kevin.withnall.com/2007/07/09/asterisk-alarm-receiver-using-triggers-mysql5/ if anyone wants them. -- Kevin Withnall http://kevin.withnall.com/ ILB Computing http://www.ilb.com.au PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 Please consider the environment before printing this e-mail From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, 18 October 2007 7:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Refrigerator Alarms The refrigerators will have external outputs to trip relays (even if your customer doesn't know that), ask for the number of their refrigeration mechanic he will tell you how to get electrical/relay outputs for the alarms. These are then connected to asterisk via an interface board so when something trips it results in an event in asterisk (check out the asterisk at home X10 configurations for some ideas), this will then result in a call with a number (or even better a recording being played), eg meat fridge number 1 at 6 degrees celcius That's pretty much your answer J Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balu Raman Sent: Wednesday, 17 October 2007 4:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Refrigerator Alarms Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? Thanks, balu raman -- ___ --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] Refrigerator Alarms
Hi Balu, http://www.digitemp.com/ has all the info you need. Cost of hardware around US$70. Will give you pretty fridge temperature graphs too! You can easily hack some of Brian's scripts to do different levels of temp alarm and trigger calls in Asterisk. regards, Drew Balu Raman wrote: Omar, I am hoping that there may be some temp sensor interface that can be routed to a pc and if the temp falls out of a range, I can have this event call someone. I know what to do in asterisk to make a call. I have to do some research. may be, someone has already done a similar thing. Has to be event driven. Thanks, balu raman On 10/17/07, Omar A. Sabek [EMAIL PROTECTED] wrote: Balu, Do you want events passed to Asterisk from the refrigerator? Or does a reminder type phone call need to be placed on an interval? Please be more specific, since this sounds like a special purpose refrigerator, does it have any way of passing events to an external device? Omar A. Sabek On 10/17/07, Balu Raman [EMAIL PROTECTED] wrote: Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? Thanks, balu raman ___ --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 ___ --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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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] phone as control interface(was99bottlesofbeer)
Hi Lenz. What I did to interface asterisk with the door opener was to implement, in the board, a custom embedded server that receives and parses a set of UDP packets containing a known data in it. In the dialplan I then call a perl AGI script that sends UDP packets in the correct sequence and with proper contents inside. I've implemented in this way to be able to open the door not only by calling an internal extension but even through a dedicated external pushbutton connected to a second board that sends the same UDP packet sequence when the pushbutton is pressed. I did some other experiments trying to embed a very lite SIP layer (written from scratch to be the more embeddable as possible) and I was able to register to my Asterisk PBX and to answer to OPTION packets (sent by the PBX) to be qualified as a valid SIP channels... but it's today only an experiment because I never had time to terminate it. :-( SIP is complex and to write a SIP compliant layer is a very time consuming stuff. Another think that could be done is to use the already working HTTP server layer. Think about sending an HTTP GET or POST, again, with an external perl AGI script. Thank you and bye. Marco Signorini. Original Message Subject: Re: [asterisk-users] phone as control interface(was99bottlesofbeer) From:Lenz [EMAIL PROTECTED] Date:Thu, October 18, 2007 1:15 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com -- Hello Marco, could you explain how you did the interfacing to the Asterisk PBX? does your prototype speak SIP to receive commands? Thanks l. On Thu, 18 Oct 2007 12:27:33 +0200, marcotasto [EMAIL PROTECTED] wrote: Hi All, sorry if I post again this e-mail but I think the first one was lost. I don't know if this is OT but I'm working in my spare time at a small hardware project that match to what's requested below. It's a board with Input/Output capabilities and 10Mbps ethernet interface. It has Microchip software TCP/IP stack on it. Being at a very beginning stage, you can see a little preview (and hopefully play with an online prototype) at this address: http://www.auto-matica.com/index.php?id=16 I've used it in the past to open a garage door when calling a private extension with Asterisk PBX and a little perl-AGI glue. I hope could interest to someone and I'm open to any suggestion/collaborations. Thank you and bye, Marco. - Original Message - From: John Faubion [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 16, 2007 9:46 AM Subject: Re: [asterisk-users] phone as control interface(was99bottlesof beer) has anyone actually been satisfied with the performance of these powerline signalling devices ? Universal Powerline Bus is a vast improvement over the original X10. I believe the X10 devices used a 4V signal during the zero crossing point of the AC voltage to transmit 1 bit. Needless to say this made X10 slow, susceptible to line noise and not very reliable. IIRC X10 is only 75% reliable. By contrast, UPB is 20-40 times faster, it uses a higher signaling voltage so line noise isn't a big factor any more and has the advantage of controlling 250 times more devices than X10. This will help to prevent stray signals from the neighbors controller from accidentally controlling your devices. UPB is supposed to be 99.9% reliable with a latency of less than 100 milliseconds. Granted that still leaves a tenth of a percent of uncertainly. However that is without resorting to filters, couplers and the like. Granted in an existing situation there may not be a way to run more wires, but I evaluated them a while back and decided to stay away. You may want to take another look at them. Just like Asterisk has made great strides since the release of 0.7, UPB has brought the quality level way up. Of course this higher quality also has a higher price. John Anyone on the list interested on working on a project where we can create devices that work over Ethernet ? ___ --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 -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] CLI not showing DTMF
On Thu, 2007-10-18 at 09:25 +0200, Volker Sauer wrote: after I upgraded from 1.2 to 1.4.13 the CLI does not show DTMF anymore, even at high debug level. Do I need to activate that? You need to enable DTMF debugging in logger.conf, then type logger reload at the Asterisk CLI for those changes to take effect. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Automating blacklists
Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? Thanks for any ideas or 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] Mandriva 2008
Just a note for those of you trying to install Asterisk 1.4.x under Mandriva 2008. I wasn't able to get make menuselect to work. It kept telling me that I didn't have ncurses installed (I have ncurses-5.6-1) even though the configure script said it found it. I eventually went into the menuselect directory and did a ./configure and make. After I did that, I went back into the Asterisk root and did a make menuselect and everything worked. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Automating blacklists
On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote: I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? You could do it with an AGI, or with dialplan logic and the AstDB database. If you use the AstDB database to store blacklisted numbers, you can also use the BLACKLIST dialplan function to check to see if a given number is blacklisted or not. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Refrigerator Alarms
Balu Raman wrote: Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? OWFS for sure! Here is a screenshot of a program I created a couple years ago to monitor refrigerators/warmers. It would be trivial to have the alarm script call a phone. I have been out of the OWFS loop for a while, but I'm pretty sure there is native alarming and set points now. http://sourceforge.net/project/screenshots.php?group_id=85502ssid=33253 Here the owfs website: http://www.owfs.org/ Msg me off list if you have any questions. -jc ___ --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] Limit number of times a call can be forwarded
We have had a few different times when a user has forwarded their phone to himself. This has overloaded the communications to our operator panel (FOP). One user should not be able to effect the whole phone system! Is there a way that the number of times that a call can be forwarded could be limited like to 10 or even 100? Then even if a user does something stupid like forwarding their calls to himself, it wouldn't cause problems for others. Don Pobanz ___ --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] IAX2: Calls answered before extension is tested?
[Sorry if this arrives more than once. I have sent this twice and it never arrived, despite other messages getting to the list O.K.] --- Hello, I would like an incoming caller to be able to choose from the menu options in my extension.conf below. Once They have dialled the appropriate digit, * should call two extensions simultaneously: one SIP phone on this * server, and one over a working IAX2 link. If either party answers, great. If neither party answers or both are busy/unavailable go to relevant voicemail box. This *almost* works... ;-) The issue is the call that goes down the IAX2 channel always seems to get answered, then cleared down almost straight away. It looks like the remote * server is accepting the incoming call sending a connected message back, thereby completing the Macro, and only *then* checking if the extension is actually available. Here's the last bit of the log (I've edited the IP address) - we are both deliberately NOT answering our phones... Executing [EMAIL PROTECTED]:1] Macro(SIP/101-081d1050, belllord|SIP/101IAX2/alanb/201|tolc) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-081d1050, SIP/101IAX2/alanb/201|10|tr) in new stack -- Called 101 -- Called alanb/201 [Oct 17 16:09:47] WARNING[2836]: channel.c:2634 ast_indicate_data: Unable to handle indication 3 for 'SIP/101-081d1050' -- SIP/101-081d4fc0 is ringing -- Call accepted by 80.XXX.XX.XX (format alaw) -- Format for call is alaw -- IAX2/alanb-3 answered SIP/101-081d1050 [Oct 17 16:09:47] NOTICE[2836]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/101-081d4fc0' not posted [Oct 17 16:09:47] DEBUG[1419]: chan_iax2.c:7435 socket_process: Immediately destroying 3, having received hangup [Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3176 iax2_hangup: We're hanging up IAX2/alanb-3 now... [Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3191 iax2_hangup: Really destroying IAX2/alanb-3 now... -- Hungup 'IAX2/alanb-3' == Spawn extension (macro-belllord, s, 1) exited non-zero on 'SIP/101-081d1050' in macro 'belllord' == Spawn extension (macro-belllord, s, 1) exited non-zero on 'SIP/101-081d1050' And here's the relevant bits of my extension.conf [globals] ALANL=SIP/101 ; My Soft Phone ALANB=IAX2/alanb/201 ; Alan's Extension [main_menu] ; Test Dialplan for IVR exten = s,1,Answer() exten = s,n,Set(TIMEOUT(digit)=5) ; Max time between digits exten = s,n,Set(TIMEOUT(response)=15) ; Max time to wait exten = s,n,Wait(1) exten = s,n,Background(welcome-to-bell-lord) exten = s,n(resume),Background(press-3-for-tolc) ; Short dialogues, exten = s,n,Background(press-4-for-fondoo) ; rather than one long one exten = s,n,Background(press-5-for-arrowtees) ; might need to change exten = s,n,Background(press-6-for-gen-enq) ; frequently. exten = s,n,WaitExten() exten = 3,1,Goto(tolc,s,1) ; Dial 3 For The Open Learning Centre exten = 4,1,Goto(fondoo,s,1) ; Dial 4 for Fondoo Internet exten = 5,1,Goto(arrowtees,s,1) ; Dial 5 for ArrowTees exten = 6,1,Goto(gen_enq,s,1) ; For all other enquiries press 6 exten = i,1,Playback(pbx-invalid) exten = i,n,Goto(resume) exten = t,1,Playback(vm-goodbye) exten = t,n,Hangup() ; Might change this section to go to [gen_enq] voicemail rather than just hangup. [tolc] exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT}) ; Calls the belllord Macro with the channel(s) to dial and the current context (for business voicemail) [fondoo] exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT}) [arrowtees] exten = s,1,Macro(belllord,${ALANL},${CONTEXT}) [gen_enq] exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT}) ; Call with Macro(belllord,channel,vmbox) [macro-belllord] ; Uses macro and DIALSTATUS for local devices exten = s,1,Dial(${ARG1},10,tr) exten = s,n,Goto(s-${DIALSTATUS},1) exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED],u) ; business is the voicemail context, ${ARG2} is the context from which this call came exten = s-BUSY,1,Voicemail([EMAIL PROTECTED],b) exten = _s-.,1,Goto(s-NOANSWER,1) == Can anyone see where the problem is? Or suggest a better way? Many thanks. Alan -- The way out is open! http://www.theopensourcerer.com -- The way out is open! http://www.theopensourcerer.com ___ --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] Automating blacklists
Is there a function to write the timestamp of the first call? I started thinking about AGI and PHP/MySQL since that is what I'm familiar with. I couldn't find methods to write timestamp info to AstDB or if I could ... how to read it back and compare it to time now to decide to increment my counter and have the dial plan decide to allow the call or not. On 10/18/07, Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote: I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? You could do it with an AGI, or with dialplan logic and the AstDB database. If you use the AstDB database to store blacklisted numbers, you can also use the BLACKLIST dialplan function to check to see if a given number is blacklisted or not. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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
[asterisk-users] receiving fax over sip extension.
Sir, I am having runing asterisk 1.4 server which is runing without any problem now i want to receive fax over sip extension. how it is possible and what the change i have make in extensions.conf. Thanks Rajeev. ___ --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] Automating blacklists
On Thu, 2007-10-18 at 16:51 +0300, Brian Hutchinson wrote: Is there a function to write the timestamp of the first call? You can use the built-in channel variable ${EPOCH}, which will give you the current time in Unix timestamp format (number of seconds since Jan 1, 1970). Then you can do something like: exten = 123,n,Set(DB(myblacklist/${CALLERID(num)}/lastcall)=${EPOCH})) This obviously doesn't solve the entire problem for you, but should at least help you get moving in the right direction. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] BBC on Atserix
On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote: Just for fun. http://news.bbc.co.uk/1/hi/magazine/7049642.stm It's Asterix != Asterisk. Though named after *. -- Tzafrir Cohen icq#16849755 jabber:[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
Re: [asterisk-users] Mandriva 2008
On Thu, 2007-10-18 at 09:08 -0400, Doug Lytle wrote: I wasn't able to get make menuselect to work. It kept telling me that I didn't have ncurses installed (I have ncurses-5.6-1) even though the configure script said it found it. I eventually went into the menuselect directory and did a ./configure and make. After I did that, I went back into the Asterisk root and did a make menuselect and everything worked. Yes, that's not an uncommon problem, and it's no no way limited to just Mandriva. What happens is that the menuselect autoconf doesn't get re-run, so it never learns that you've installed the ncurses libraries. I always suggest people do a make distclean after installing any additional libraries, and then go on to the ./configure step. This will ensure that the menuselect autoconf stuff gets cleaned out. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Automating blacklists
It's not technically complex to do - you can probably use the astdb for that, or store all incoming numbers with timestamp in MySQL and run something like: SELECT count(*) 5 AS blacklisted FROM incoming_calls WHERE callerid = 12345 AND timestamp DATE_SUB( NOW(), INTERVAL 15 MINUTE ) you should be very well aware of the risks that can stem from such a program - in case of bugs, or anomalous situations, you might end up blacklisting somebody who actually needs to call in. I hope this helps l. On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson [EMAIL PROTECTED] wrote: Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? Thanks for any ideas or pointers! -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] Issues with making calls
Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) My sources are: libpri-1.4.1.tar.gz zaptel-1.4.5.1.tar.gz asterisk-1.4.11.tar.gz asterisk-addons-1.4.2.tar.gz asterisk-perl-0.10.tar.gz I have 1/2 E1 from my provider telephony, my configuration is [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us #cat /etc/asterisk/zapata.conf [channels] context=default switchtype = euroisdn pridialplan = unknown signalling = pri_cpe 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 immediate=no amaflags=documentation musiconhold=default ;Configure Channels group=0 callgroup=0 pickupgroup=0 channel = 1-4 group=1 callgroup=1 pickupgroup=1 channel = 5-8 group=2 callgroup=2 pickupgroup=2 channel = 9-12 group=3 callgroup=3 pickupgroup=3 channel = 13-14 group=4 callgroup=4 pickupgroup=4 channel = 15 I have could make calls but, after of some minutes my server is hung, suggestions are welcome. Thanks for any help 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
Re: [asterisk-users] Is anyone successfully using IMAP storage
Not using IMAP storage here, although that was one of the primary drivers for upgrading to Asterisk 1.4. Why? The short answer is that it's too confining. There are too many caveats that don't fit into our existing IMAP structure and make the entire project rather iffy. Security is a concern too - we don't want to have one set of credentials that have absolute access to everyone's email from the CEO down. I suspect we /could/ make it work, but the thought of experimenting on our live IMAP server sends chills down my spine. (we don't have the budget or time to duplicate our mail server to allow fiddling on a non-production system) The idea is brilliant. The weird dependency on having a compiled-but-not-installed UW-IMAP isn't - I would have thought that using a library such as libvmime would have made more sense. Olivier wrote: Hi, From personal experience, Asterisk 1.4 IMAP storage seems broken and unusable. Is anyone using it successfully ? This kind of poll would be very useful to estimate is a code rewrite has a chance to disturb a running system. I we get no successful report, it would help developers to consider a code rewrite as patch instead of a new feature. So please, do not hesitate to report successful or unsuccessful or never tried use of IMAP storage, it will increase our chances to get this feature in 1.4 branch and not wait for 1.6. ___ --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] Issues with making calls
Make sure chan_zap.so is loaded. /b On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote: Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) My sources are: libpri-1.4.1.tar.gz zaptel-1.4.5.1.tar.gz asterisk-1.4.11.tar.gz asterisk-addons-1.4.2.tar.gz asterisk-perl-0.10.tar.gz I have 1/2 E1 from my provider telephony, my configuration is [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us #cat /etc/asterisk/zapata.conf [channels] context=default switchtype = euroisdn pridialplan = unknown signalling = pri_cpe 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 immediate=no amaflags=documentation musiconhold=default ;Configure Channels group=0 callgroup=0 pickupgroup=0 channel = 1-4 group=1 callgroup=1 pickupgroup=1 channel = 5-8 group=2 callgroup=2 pickupgroup=2 channel = 9-12 group=3 callgroup=3 pickupgroup=3 channel = 13-14 group=4 callgroup=4 pickupgroup=4 channel = 15 I have could make calls but, after of some minutes my server is hung, suggestions are welcome. Thanks for any help 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] Automating blacklists
I feel more comfortable with MySQL ... just need to learn how to get the dialplan to use it. Also figure out the pro's/con's to MySQL vs AstDB. If I used MySQL then I could put myphpadmin and get a pseudo GUI to manipulate the blacklist database for almost no effort so that is another reason for leaning towards MySQL. Thanks for the nudge toward MySQL. On 10/18/07, Lenz [EMAIL PROTECTED] wrote: It's not technically complex to do - you can probably use the astdb for that, or store all incoming numbers with timestamp in MySQL and run something like: SELECT count(*) 5 AS blacklisted FROM incoming_calls WHERE callerid = 12345 AND timestamp DATE_SUB( NOW(), INTERVAL 15 MINUTE ) you should be very well aware of the risks that can stem from such a program - in case of bugs, or anomalous situations, you might end up blacklisting somebody who actually needs to call in. I hope this helps l. On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson [EMAIL PROTECTED] wrote: Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? Thanks for any ideas or pointers! -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] Issues with making calls
On Thu, 2007-10-18 at 09:34 -0500, Pablo Almido wrote: [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) That would seem to indicate that the chan_zap.so module isn't being loaded. What happens if you type module unload chan_zap.so and then module load chan_zap.so from the Asterisk CLI? I'll bet you'll find that either there's a problem in your zapata.conf file, or that chan_zap hasn't been compiled for some reason. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Automating blacklists
How do you know that the call is a prank call, an not just someone that likes calling your company alot... ? If you just want a database of callerid's to block, here is what I have used, I hope it helps some My SQL table looks has 4 columns id (autoincrement), callerid, blockenabled (to enable or disable the block), and notes. [general] realdb_host=localhost realdb_user=asterisk realdb_pass=password realdb_db=asterisk_realtime [pri-in] ; Conference Room Number exten = 193,1,Answer() exten = 193,2,Macro(checkblacklist,${CALLERID(num)}) exten = 193,3,GoTo(us-conference,s,1) [macro-checkblacklist] ; This Macro will check the blacklist table to see if the callerid of the ; caller exist and blockenabled =1 (TRUE). If the callerid is listed, then ; tell the caller they have been blacklisted and politely HangUp() ; ; ${ARG1} = CallerID of incoming call ; exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user} $ {realdb_pass} ${realdb_db}) exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\ blacklist\ where\ callerid=${ARG1} and blockenabled = 1) exten = s,3,MYSQL(Fetch fetchid ${resultid} blacklistid) exten = s,4,MYSQL(Clear ${resultid}) exten = s,5,MYSQL(Disconnect ${connid}) exten = s,6,GoToIf($[${blacklistid} = ]?7:fail,1) exten = s,7,NoOp(Not blocked in Blacklist) ; If the callerid is listed in the database, then send to blacklistednumber ; context ; exten = fail,1,NoOp(${blacklistid}) exten = fail,2,GoTo(blacklistednumber,s,1) [blacklistednumber] ; This is where a call will land if the macro-checkblacklist decides that ; the number should not be allowed to dial DA exten = s,1,Wait(2) exten = s,2,Playback(privacy-you-are-blacklisted) exten = s,3,HangUp() Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Oct 18, 2007, at 10:25 AM, Lenz wrote: It's not technically complex to do - you can probably use the astdb for that, or store all incoming numbers with timestamp in MySQL and run something like: SELECT count(*) 5 AS blacklisted FROM incoming_calls WHERE callerid = 12345 AND timestamp DATE_SUB( NOW(), INTERVAL 15 MINUTE ) you should be very well aware of the risks that can stem from such a program - in case of bugs, or anomalous situations, you might end up blacklisting somebody who actually needs to call in. I hope this helps l. On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson [EMAIL PROTECTED] wrote: Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? Thanks for any ideas or pointers! -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] Issues with Zaptel 1.4.5.1
On Thu, Oct 18, 2007 at 08:35:01AM +0100, Alan Lord wrote: Tzafrir Cohen wrote: snip /... is required to properly start zaptel. It will also run ztcfg. Otherwise users run into issues where misconfigured zaptel.conf fails loading of a module. That is a buggy behaviour. If your card is an analog one, take a look at http://bugs.digium.com/7613 and tell me what you think. Something similar for digital spans would require more information in sysfs. So do I understand this correctly? I can use this patch to alleviate the need for a userspace init.d start-up script? I just patch the zaptel module source and rebuild? Does it read /etc/zaptel.conf and thus load the zone data? I only have an x100p so this could be a useful patch for me. As I mentioned in recent notes: that patch is not up-to-date. It also does not set the tone zones. -- Tzafrir Cohen icq#16849755 jabber:[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
Re: [asterisk-users] BBC on Atserix
Tzafrir Cohen wrote: On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote: Just for fun. http://news.bbc.co.uk/1/hi/magazine/7049642.stm It's Asterix != Asterisk. Though named after *. In Britain, it's called humour :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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] Automating blacklists
Yeah I would use MySQL as well for more or less the same reasons. Using MySQL right from the dialplan is not very elegant but it's pretty simple - see http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MYSQL Thanks l. On Thu, 18 Oct 2007 16:49:48 +0200, Brian Hutchinson [EMAIL PROTECTED] wrote: I feel more comfortable with MySQL ... just need to learn how to get the dialplan to use it. Also figure out the pro's/con's to MySQL vs AstDB. If I used MySQL then I could put myphpadmin and get a pseudo GUI to manipulate the blacklist database for almost no effort so that is another reason for leaning towards MySQL. Thanks for the nudge toward MySQL. On 10/18/07, Lenz [EMAIL PROTECTED] wrote: It's not technically complex to do - you can probably use the astdb for that, or store all incoming numbers with timestamp in MySQL and run something like: SELECT count(*) 5 AS blacklisted FROM incoming_calls WHERE callerid = 12345 AND timestamp DATE_SUB( NOW(), INTERVAL 15 MINUTE ) you should be very well aware of the risks that can stem from such a program - in case of bugs, or anomalous situations, you might end up blacklisting somebody who actually needs to call in. I hope this helps l. On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson [EMAIL PROTECTED] wrote: Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? Thanks for any ideas or pointers! -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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 -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] Issues with making calls
Why would a config error stop the module from loading? That seems like a suboptimal behavior. /b On Oct 18, 2007, at 9:50 AM, Jared Smith wrote: That would seem to indicate that the chan_zap.so module isn't being loaded. What happens if you type module unload chan_zap.so and then module load chan_zap.so from the Asterisk CLI? I'll bet you'll find that either there's a problem in your zapata.conf file, or that chan_zap hasn't been compiled for some reason. ___ --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] Is anyone successfully using IMAP storage
On 10/18/07, Olivier [EMAIL PROTECTED] wrote: Hi, From personal experience, Asterisk 1.4 IMAP storage seems broken and unusable. Is anyone using it successfully ? I've read blog entries that indicate that people have used it successfully but I have not been able to get it to connect to the IMAP server in all my trials. ___ --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] Issues with making calls
Yes, the module is load # asterisk -r ippbx*CLI module show like chan_zap.so Module Description Use Count chan_zap.soZapata Telephony 0 1 modules loaded ippbx*CLI ippbx*CLI 2007/10/18, Brian West [EMAIL PROTECTED]: Make sure chan_zap.so is loaded. /b On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote: Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) My sources are: libpri-1.4.1.tar.gz zaptel-1.4.5.1.tar.gz asterisk-1.4.11.tar.gz asterisk-addons-1.4.2.tar.gz asterisk-perl-0.10.tar.gz I have 1/2 E1 from my provider telephony, my configuration is [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us #cat /etc/asterisk/zapata.conf [channels] context=default switchtype = euroisdn pridialplan = unknown signalling = pri_cpe 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 immediate=no amaflags=documentation musiconhold=default ;Configure Channels group=0 callgroup=0 pickupgroup=0 channel = 1-4 group=1 callgroup=1 pickupgroup=1 channel = 5-8 group=2 callgroup=2 pickupgroup=2 channel = 9-12 group=3 callgroup=3 pickupgroup=3 channel = 13-14 group=4 callgroup=4 pickupgroup=4 channel = 15 I have could make calls but, after of some minutes my server is hung, suggestions are welcome. Thanks for any help 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 ___ --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] centos 5 vs OpenSuse 10.3
Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian ___ --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] Is anyone successfully using IMAP storage
On 10/18/07, Olivier [EMAIL PROTECTED] wrote: Hi, From personal experience, Asterisk 1.4 IMAP storage seems broken and unusable. Is anyone using it successfully ? This kind of poll would be very useful to estimate is a code rewrite has a chance to disturb a running system. I we get no successful report, it would help developers to consider a code rewrite as patch instead of a new feature. So please, do not hesitate to report successful or unsuccessful or never tried use of IMAP storage, it will increase our chances to get this feature in 1.4 branch and not wait for 1.6. Cheers I wanted to use it but there's *NO* documentation. We don't use dovecot but our IMAP server has a master user function... can't figure out if its the same as the dovecot function, because: 1) The dovecot function isn't documented 2) There's no documentation as to how asterisk writes to the IMAP. So I'll put it out to the world to see if someone knows how it works Our email server can do this: You can log into any mailbox via IMAP if your user has mboxadmin capabilities. So if you want to use UserA to log into UserB's mailbox, you can do the following: Username: mboxadmin:UserA:UserB Password: Password_User_A Is that what Asterisk is expecting? ___ --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] Issues with making calls
I have unload and load the module, it is output ippbx*CLI module unload chan_zap.so == Unregistered application 'ZapSendKeypadFacility' ippbx*CLI module load chan_zap.so == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Oct 18 10:46:38] WARNING[2790]: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:7160 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:10466 build_channels: Unable to register channel '1-4' ippbx*CLI 2007/10/18, Brian West [EMAIL PROTECTED]: Why would a config error stop the module from loading? That seems like a suboptimal behavior. /b On Oct 18, 2007, at 9:50 AM, Jared Smith wrote: That would seem to indicate that the chan_zap.so module isn't being loaded. What happens if you type module unload chan_zap.so and then module load chan_zap.so from the Asterisk CLI? I'll bet you'll find that either there's a problem in your zapata.conf file, or that chan_zap hasn't been compiled for some reason. ___ --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] centos 5 vs OpenSuse 10.3
On Thu, Oct 18, 2007 at 12:22:24PM -0400, [EMAIL PROTECTED] wrote: Just 5 months ago CENTOS started to use Linux 2.6 Centos 4 (based on RHEL4) used kernel 2.6 as well. It was released over two years ago (it has 2.6.9). -- Tzafrir Cohen icq#16849755 jabber:[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
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
I'm sorry I call bullshit on this one. CentOS has been 2.6 for some time. /b On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote: Just 5 months ago CENTOS started to use Linux 2.6 one of the reasons I'd abandoned for SuSE a while back. ___ --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] centos 5 vs OpenSuse 10.3
Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --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] Anyone having any luck with Bluetooth?
On Wed, 2007-10-17 at 15:09 -0700, shadowym wrote: I have read all the wiki's and blogs and how to links about Bluetooth but so far no luck. I can confirm that CentOS5 sees my Bluetooth adapter and my cell phone. No Joy on Asterisk 1.4. The information out there is kind of confusing as there is a lot of outdated info sometimes referring to software no longer actively developed. What I think I have managed to conclude is that there is a Bluetooth module for Asterisk 1.4 that supposedly works but it's not part of any released branches so I will have to use a development branch. That is ok but I still can't get it to work. I want to use my Cell phone as a secondary trunk for a business so I need it to work reasonably well. Does it (if and when I get it working)? Is there a good (up to date relevant) how to somewhere? Haven't tried chan_mobile against 1.4; just against trunk; and have it working OK with 2 of 3 bluetooth dongles with my samsung cellphone. Look in asterisk-addons/trunk for chan_mobile.c; set up your config file; mine looks like this: ; ; configuration file for chan_mobile ; [general] interval=30 ; Number of seconds between trying to connect to devices. ; The following is a list of adapters we use. ; id must be unique and address is the bdaddr of the adapter from hciconfig. ; Each adapter may only have one device (headset or phone) connected at a time. ; Add an [adapter] entry for each adapter you have. [adapter] id=encore address=00:11:C6:09:DF:99 [adapter] id=iogear_gbu221 address=00:02:49:07:FC:44 [adapter] id=asus address=00:02:72:27:92:8D ; The following is a list of the devices we deal with. ; Every device listed below will be available for calls in and out of Asterisk. ; Each device needs an adapter= entry which determines which bluetooth adapter is used. ; Use the CLI command 'mobile search' to discover devices. ; Use the CLI command 'mobile show devices' to see device status. ; ; To place a call out through a mobile phone use Dial(Mobile/[device]/NNN.) or Dial(Mobile/gn/NNN..) in your dialplan. ; To call a headset use Dial(Mobile/[device]). [steve] context=incoming address=00:1B:62:52:F4:F2 port=4 adapter=encore group=1 ;; The following is a list of the devices we deal with. ;; Every device listed below will be available for calls in and out of Asterisk. ;; Discovered devices not in this list are not available. ;; Use the CLI command 'mobile search' to discover devices. ;; Use the CLI command 'mobile show devices' to see device status. ;; ;; To place out through a cell phone use Dial(Mobile/[device]/NNN.) in your dialplan. ;; To call a headset use Dial(Mobile/[device]). ;; -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic 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
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
On 10/18/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian SuSE is known for using the latest packages with each release, and RHEL/CENTOS are known for using old citing stability. SuSE Is just as stable, they don't use beta version only tested stable releases. Just 5 months ago CENTOS started to use Linux 2.6 one of the reasons I'd abandoned for SuSE a while back. ___ --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] Automating blacklists
On Thursday 18 October 2007 09:49:48 Brian Hutchinson wrote: I feel more comfortable with MySQL ... just need to learn how to get the dialplan to use it. See func_odbc.conf. -- Tilghman ___ --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] Softphone that emulates Skype API ?
On Thu, Oct 18, 2007 at 01:05:16PM +0300, Cosmin Prund wrote: There's a large number of gadgets one can buy that work with Skype through the API. One of the things I'm interested right now is the ability to properly use a mobile phone headset with a SIP/IAX softphone. Is there an softphone that emulates the Skype API? Are there legal implications in writing an softphone that emulates the Skype API? Should I just give up and buy a Siemens DECT phone that supports a bluetooth headset? Many of the skype phones can at least be used as an extra sound device. Using them as a handset is probably more complicated. -- Tzafrir Cohen icq#16849755 jabber:[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
Re: [asterisk-users] Issues with making calls
On Thu, Oct 18, 2007 at 10:53:15AM -0500, Pablo Almido wrote: I have unload and load the module, it is output ippbx*CLI module unload chan_zap.so == Unregistered application 'ZapSendKeypadFacility' ippbx*CLI module load chan_zap.so == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Oct 18 10:46:38] WARNING[2790]: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:7160 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:10466 build_channels: Unable to register channel '1-4' cat /proc/zaptel/1 Did it get properly configured? http://rapid.tzafrir.org.il/docs/README.html#toc20 (the section about the procfs interface) -- Tzafrir Cohen icq#16849755 jabber:[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
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
On 10/18/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Oct 18, 2007 at 12:22:24PM -0400, [EMAIL PROTECTED] wrote: Just 5 months ago CENTOS started to use Linux 2.6 Centos 4 (based on RHEL4) used kernel 2.6 as well. It was released over two years ago (it has 2.6.9). Never trust anyone... I've been anti-Cent/RHEL for a while because of these guys: http://distrowatch.com/table.php?distribution=centos ___ --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] Blocking collect calls in Brazil
Hi Carlos, The method commonly used is flash E1 to block this calls. A Better way could be detect the category of call. I don´t know if there is a way to get the call category in extensions as we can get the CALLERID. Collect calls have B-8 category in Brazil. Thanks. Luis A P Barbosa 2007/10/18, Carlos Barros [EMAIL PROTECTED]: Hi list. I was googling about this subject and found a message to this list asking the same thing I want, but see no response. So, I'm reposting it in case someone has a solution for this. I'd like to know if there's some way to detect (not block) a collect call in a ISDN E1? I need this cause I have my own billing system and I need to charge my customers for collect calls they receive. Currently I have to get the PSTN billing and locate all collect calls and then enter it manually. I want to detect these calls in asterisk so I can automatically charge my customers. For Brazilians in the list, I have a Embratel E1 - ISDN. Thanks Carlos Barros ___ --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
[asterisk-users] Incoming calls
Hello I have a question about incoming calls on TDM400P cards. I want to know why an incoming call appear in a sorpresive way on a phone that I pickup to call out. I am using ChanIsAvailable to check those lines ( Zap channels )that are free. I have four lines connected to my TDM400P card and when I get a free Zap channel to call I hear the voice of a people on the other side from an incomming call, I think that asterisk bridge my free channel with incomming calls but how do this?Thanks for any idea. Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006 01:45 p.m. ___ --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] BBC on Atserix
Drew Gibson wrote: Tzafrir Cohen wrote: On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote: Just for fun. http://news.bbc.co.uk/1/hi/magazine/7049642.stm It's Asterix != Asterisk. Though named after *. In Britain, it's called humour :-) regards, Drew Only if it's actually humourous. ;) N. ___ --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] Automating blacklists
On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote: Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be looking at AGI to do something like this? Thanks for any ideas or pointers! I hate telemarketers. And at ***least*** one calls every day. But I rarely have to talk to them. Using AstDB, and dialplan programming in AEL, I have implemented some measures that cut the number of telemarketers who make it through to only maybe 1 per year. All my stuff is in AEL. Along this line, my home phone system records all CID's coming in, and counts each incoming call for each CID. Telemarketer autodialers tend to call in without any CID. So, I play anonymous callers the telezapper tri-tones. It's kinda irritating to callers to get this, so I only play it to eligible callers. Autodialers usually immediately hang up, and because they usually interpret tri-tones as wrong number, they remove me from their db, and I don't hear from them again... But there is a growing number of telemarketers who actually supply some sort of CID info for the call. I count how many times they call, but get filtered out. After I get the same party calling in 4 times or more in row, that don't actually end up talking to anyone, I start playing them the tri-tones. That usually is the end of that sequence. All this is done with simple dialplan statements and the use of AstDB (see the DB() function). I also whitelist, and route certain callers directly to particular extensions, bypassing all filters. I also have DB entries that will override the default MOH for some callers, based either on CID, or who they are calling. For instance, one of my sons preferred his callers got rock music instead of the normal elevator stuff. Also available to those who prefer not to be bothered continuously by charity seekers, political pollsters, long-distance dialing hawkers, septic-tank bacteria salesmen, etc, are the privacy related options to the Dial() app. You can do call screening, and route unwanted incoming calls to voicemail. Might even help with debt collectors, etc. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic 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
[asterisk-users] Multi-site / Multi-server coordination
Okay, so we are planning for the future here where I work so we are trying to do testing ahead of time as we might be setting up a satellite campus that would need its own Asterisk phone system but still tied into our main campus phone system. This much we have accomplished. We have a central database that routes calls to the correct server before dialing the final destination SIP phone (or Zap, IAX, whatever). Our last big question is this: How, if at all, does asterisk deal with multi-server voicemail? i.e. User A is at site A and has a voicemail in his mailbox. He knows it really belongs to user B (who is at site B) so after listening to the voicemail he forwards it to user B. Is there a way for Asterisk to realize this box belongs on a different machine and communicate the voicemail over the network to that machine, or is this something I need to write? :) Daniel Hazelbaker Information Technology Director High Desert Church ___ --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] Crash related to asterisk -rx ?
Atis Lezdins a écrit : Yup, it's also a problem for me, but it haven't ever crashed server. It just makes specific remote process unresponsive. There's a patch for 1.4, but i guess it wouldn't be hard to backport it for 1.2 http://bugs.digium.com/view.php?id=10847 you might also want the one mentioned in comments: http://bugs.digium.com/view.php?id=10888 Regards, Atis Atis, Thanks for the reply and pointers. Best regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --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] BBC on Atserix
SIP wrote: Drew Gibson wrote: Tzafrir Cohen wrote: On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote: Just for fun. http://news.bbc.co.uk/1/hi/magazine/7049642.stm It's Asterix != Asterisk. Though named after *. In Britain, it's called humour :-) regards, Drew Only if it's actually humourous. ;) N. ___ --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 It's looking good ... http://news.bbc.co.uk/2/hi/entertainment/4343264.stm regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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] centos 5 vs OpenSuse 10.3
On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote: Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. Well, no. /Per Jessen, Zürich We use only openSUSE. If A distro is a distro, then why? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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] Issues with making calls
I run this command [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE12xP Card 0 IRQ misses: 40 1 WCT1/0/1 2 WCT1/0/2 3 WCT1/0/3 4 WCT1/0/4 5 WCT1/0/5 6 WCT1/0/6 7 WCT1/0/7 8 WCT1/0/8 9 WCT1/0/9 10 WCT1/0/10 11 WCT1/0/11 12 WCT1/0/12 13 WCT1/0/13 14 WCT1/0/14 15 WCT1/0/15 16 WCT1/0/16 17 WCT1/0/17 18 WCT1/0/18 19 WCT1/0/19 20 WCT1/0/20 21 WCT1/0/21 22 WCT1/0/22 23 WCT1/0/23 24 WCT1/0/24 25 WCT1/0/25 26 WCT1/0/26 27 WCT1/0/27 28 WCT1/0/28 29 WCT1/0/29 30 WCT1/0/30 31 WCT1/0/31 Then I run ztcfg [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Version: 1.4.5.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to Clear channel Changing signalling on channel 4 from Unused to Clear channel Changing signalling on channel 5 from Unused to Clear channel Changing signalling on channel 6 from Unused to Clear channel Changing signalling on channel 7 from Unused to Clear channel Changing signalling on channel 8 from Unused to Clear channel Changing signalling on channel 9 from Unused to Clear channel Changing signalling on channel 10 from Unused to Clear channel Changing signalling on channel 11 from Unused to Clear channel Changing signalling on channel 12 from Unused to Clear channel Changing signalling on channel 13 from Unused to Clear channel Changing signalling on channel 14 from Unused to Clear channel Changing signalling on channel 15 from Unused to Clear channel Changing signalling on channel 16 from Unused to HDLC with FCS check Changing signalling on channel 17 from Unused to Clear channel Changing signalling on channel 18 from Unused to Clear channel Changing signalling on channel 19 from Unused to Clear channel Changing signalling on channel 20 from Unused to Clear channel Changing signalling on channel 21 from Unused to Clear channel Changing signalling on channel 22 from Unused to Clear channel Changing signalling on channel 23 from Unused to Clear channel Changing signalling on channel 24 from Unused to Clear channel Changing signalling on channel 25 from Unused to Clear channel Changing signalling on channel 26 from Unused to Clear channel Changing signalling on channel 27 from Unused to Clear channel Changing signalling on channel 28 from Unused to Clear channel Changing signalling on channel 29 from Unused to Clear channel Changing signalling on channel 30 from Unused to Clear channel Changing signalling on channel 31 from Unused to Clear channel Then I back to run this command , but I can not understand why it change in channel 16, However I can not make calls [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE12xP Card 0 HDB3/CCS/CRC4 IRQ misses: 40 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
Jay R. Ashworth wrote: On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote: Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. Well, no. /Per Jessen, Zürich We use only openSUSE. If A distro is a distro, then why? Because we like it, and because we're used to it. That we have an operational preference doesn't change a distro is a distro. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] centos 5 vs OpenSuse 10.3
I doubt it. hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/ Original Message Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3 From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 18/10/2007 11:25 a.m. Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --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] centos 5 vs OpenSuse 10.3
On Thu, Oct 18, 2007 at 02:54:36PM -0500, Perssy Llamosas wrote: I doubt it. hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/ The above page (even with the proper protocol name) is very low on actual facts and reasonings. It mentions nothing specific to openSUSE 10.3 . So let's get back to the topic: anybody tried the package of Asterisk included in 10.3? -- Tzafrir Cohen icq#16849755 jabber:[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
Re: [asterisk-users] Incoming calls
Glare that's what it's called, if the number you advertise as your business number is zap/1 then use zap/G1 to dial out, otherwise use zap/g1 to dial out. This will reduce but not eliminate the problem. http://www.telos-systems.com/techtalk/gldefs.htm#Glare On 10/18/07, Gustavo Gonzalez [EMAIL PROTECTED] wrote: Hello I have a question about incoming calls on TDM400P cards. I want to know why an incoming call appear in a sorpresive way on a phone that I pickup to call out. I am using ChanIsAvailable to check those lines ( Zap channels )that are free. I have four lines connected to my TDM400P card and when I get a free Zap channel to call I hear the voice of a people on the other side from an incomming call, I think that asterisk bridge my free channel with incomming calls but how do this?Thanks for any idea. Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006 01:45 p.m. ___ --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
[asterisk-users] Ring Groups
Here's what I'm looking to do exten = 10,1,Dial(SIP/1000SIP/1001,15,wW) exten = 10,2,Voicemail(u1000) This should ring both phones and they should keep ringing for the alloted time before moving on. However, it appears that if one of the phones is Busy, it returns with a busy and moves on without really ringing the second phone. Short of checking if the channels are available or using a queue, is there a way to ignore the return value and just make it ring for 10 seconds and then move on to the second step? Any Suggestions? ___ --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] First Time T1 Questions
Hi all, i have been using asterisk for a few years but i am about to do my first t1 setup. After terrible quality issues between two business locations, we have decided to purchase a point to point t1 from the local phone co. The internet is too crappy, too much lag, queing and jitter. Most calls were dropped. I was about to order two cisco routers with csu cards and remembered our wonderful asterisk supports direct t1. I remembered digium and sangoma both make these cards. After some problems with a digium fxo card, i just ordered a sangoma a200 with echo cancellation. I was also leaning towards getting the single t1 sangoma card that is $499 from voip supply. But i know digium also makes one. I was wondering if the digium card works better or much easier with asterisk? The digium description says you can split the t1 for voice and data which sounds nice since i will only be using probably 4 channels max of the t1. Does the sangoma card also do this? I noticed the sangoma card has a 5 year warranty which is nice since i have had multiple digium fxo cards die. Is there any other reason to get or the other? Thank you all for your help. I am hoping this opens up a whole new world in asterisk for me. -Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] 64 bit asterisk
I hope you have better success than I did, my problem was not so much with asterisk in particular but 64-bit in general. Examples of problems using CentOS 4.5 on x86_64 - many problems loading php5 mysql from package repositories. - a few asterisk functions don't work, eg STRFTIME() Perhaps the distro you are using is more caught up on 64 bit. Everything upgraded/updated without a hitch on 32 bit. 64 bit is a no go unless you are running packages that have matured for atleast a couple of years old...imho. -- On 10/18/07, joakimsen wrote: Nope it should just work. Just finished setting up 1.4 for the first time in a while and just works. Been running 1.2 for the longest time and same thing. On 10/18/07, Wai Wu wrote: Hi list, I just installed 64 bit Linux, and ready to install Asterisk through source on it. Are there any settings have to change to build 64 bit Asterisk? Thnx a million. ___ ___ --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] Blocking collect calls in Brazil
Hi Luis. On 10/18/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Hi Carlos, The method commonly used is flash E1 to block this calls. Yes.. But I need to detect it so I can set a billing on it. A Better way could be detect the category of call. I don´t know if there is a way to get the call category in extensions as we can get the CALLERID. Collect calls have B-8 category in Brazil. Hmm.. I'll investigate it. The callerID does not specify (I guess Embratel should send a 9 or something as a CAllerID prefix, but unfortunately not). Thanks for you information. Carlos Thanks. Luis A P Barbosa 2007/10/18, Carlos Barros [EMAIL PROTECTED]: Hi list. I was googling about this subject and found a message to this list asking the same thing I want, but see no response. So, I'm reposting it in case someone has a solution for this. I'd like to know if there's some way to detect (not block) a collect call in a ISDN E1? I need this cause I have my own billing system and I need to charge my customers for collect calls they receive. Currently I have to get the PSTN billing and locate all collect calls and then enter it manually. I want to detect these calls in asterisk so I can automatically charge my customers. For Brazilians in the list, I have a Embratel E1 - ISDN. Thanks Carlos Barros ___ --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 ___ --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] Free help
Hello all, i would like to have references so i'm giving free help for any project (commercial or public). regards, -- Your next Partner ! ___ --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] Issues with making calls
ZAP/G1 will dial starting from channel 1. ZAP/R1 will dial starting from the last channel of the group. Pablo, please tell us what version of Linux and which distribution are you using. Maybe for the time being try the stock asterisk of your distro or the one they provide in the buildservice? On 10/18/07, Brett Crapser [EMAIL PROTECTED] wrote: Pablo - You said you have 1/2 E1 - which half??? That might be your problem. Unless 1/2 E1 means something else... Asterisk normally dials out on the low end unless you specify G instead of g ??? or something like that. Brett ___ --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] Softphone that emulates Skype API ?
On 10/18/07, Cosmin Prund [EMAIL PROTECTED] wrote: One of the things I'm interested right now is the ability to properly use a mobile phone headset with a SIP/IAX softphone. Thats a function of the bluetooth stack. Go ahead and pair your bluetooth headset to your PC it will work like any audio device. Any softphone that supports selecting the input source should work. With the Widcomm bluetooth stack anyways Is there an softphone that emulates the Skype API? Not that I am aware. Skype is proprietary and closed platform. Are there legal implications in writing an softphone that emulates the Skype API? Not that I am aware. Even the draconian DMCA laws permit circumvention of these digital copyright protection mechanisms for software interoperability. Should I just give up and buy a Siemens DECT phone that supports a bluetooth headset? Giving up is your prerogative ___ --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] 64 bit asterisk
Hi list, I just installed 64 bit Linux, and ready to install Asterisk through source on it. Are there any settings have to change to build 64 bit Asterisk? Thnx a million. ___ --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] centos 5 vs OpenSuse 10.3
I doubt it. boycottnovell.com/2007/10/02/opensuse-103-release/ Original Message Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3 From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 18/10/2007 11:25 a.m. Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --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] centos 5 vs OpenSuse 10.3
I doubt it. boycottnovell.com/2007/10/02/opensuse-103-release Original Message Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3 From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 18/10/2007 11:25 a.m. Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --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] Issues with making calls
Pablo - You said you have 1/2 E1 - which half??? That might be your problem. Unless 1/2 E1 means something else... Asterisk normally dials out on the low end unless you specify G instead of g ??? or something like that. Brett ___ --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] centos 5 vs OpenSuse 10.3
While I am a fan of CentOS some pople just take it tooo far. - Original Message - From: Perssy Llamosas [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 18, 2007 9:54 PM Subject: Re: [asterisk-users] centos 5 vs OpenSuse 10.3 I doubt it. hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/ Original Message Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3 From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 18/10/2007 11:25 a.m. Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --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 ___ --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] centos 5 vs OpenSuse 10.3
We used to use CentOS 4 here but about 6-8 months ago we found that they were too slow with updates their repos for some of the 3rd party software that we were developing. We switched to SuSe 10.2 and haven't looked back. However Asterisk works equally well on both. Just pick your favorite flavor. Cheers, Joel. On Thu, 2007-10-18 at 11:34 -0500, Brian West wrote: I'm sorry I call bullshit on this one. CentOS has been 2.6 for some time. /b On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote: Just 5 months ago CENTOS started to use Linux 2.6 one of the reasons I'd abandoned for SuSE a while back. ___ --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] ABE 1.4 release date
[EMAIL PROTECTED] wrote: Do you think there might be a working t.38 implementation in ABE 1.4? Even passthru Between endpoints and gateways with Asterisk in the middle? ABE does not contain functionality that isn't in an open source version of Asterisk, except for license- or patent-encumbered bits. The T.38 support in ABE version C.x is identical to that in Asterisk 1.4. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] A linksys SPA921 behind NAT and firewall
Did you set NAT Keep Alive Enable: = Yes for the line in question in the SPA's configuration? On 10/18/07, Per Jessen [EMAIL PROTECTED] wrote: I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 1-2. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side, coz' how would it know that UDP-traffic to SPA publicIP:5060 needs to be delivered to 192.168.x.x:5060 ? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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
[asterisk-users] polycom ip330/ip501 second ethernet port
Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught ___ --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] Looking for free DID with IAX
I know I can get free DID's with SIP, is anyone giving out free DID's with IAX? Thanks in advance, In the words of the great jbot in the #asterisk channel on irc.freenode.net Dovid ~ygwypf jbot well, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. ___ --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] Zaptel 1.2.21 and 1.4.6 released
The Asterisk.org development team has announced the release of Zaptel versions 1.2.21 and 1.4.6. These releases contain many bug fixes as well as performance enhancements (Too many to list here). A couple of major changes: there is an update to the Octasic API version as well as a considerable rewrite of the wct4xxp driver. The xpp drivers have been updated quite a bit as well. There was also a fix for the wctdm24xxp driver which sometimes reported false power alarms. For further details as well as the additional changes, see the respective Changelogs. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Thank you for your support! ___ --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] Limit number of times a call can be forwarded
- Original Message - From: Don Pobanz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 18, 2007 3:38 PM Subject: [asterisk-users] Limit number of times a call can be forwarded We have had a few different times when a user has forwarded their phone to himself. This has overloaded the communications to our operator panel (FOP). One user should not be able to effect the whole phone system! Is there a way that the number of times that a call can be forwarded could be limited like to 10 or even 100? Then even if a user does something stupid like forwarding their calls to himself, it wouldn't cause problems for others. Don Pobanz In sip.conf (assuming you are using SIP) you can set call-limit=10. This would limit the amount of calls the SIP account can place to 10. Another thing you can is have a macro compare the CID of the phone to where the call is going to. If they are the same dump the call and maybe have an agi that emails the sysadmin and or user of the issue. ___ --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] 64 bit asterisk
Nope it should just work. Just finished setting up 1.4 for the first time in a while and just works. Been running 1.2 for the longest time and same thing. On 10/18/07, Wai Wu [EMAIL PROTECTED] wrote: Hi list, I just installed 64 bit Linux, and ready to install Asterisk through source on it. Are there any settings have to change to build 64 bit Asterisk? Thnx a million. ___ --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] IAX2: Calls answered before extension is tested?
So your problem is: -- IAX2/alanb-3 answered SIP/101-081d1050 Except the remote end didn't actually answer the call? The problem is your remote end... its answering the call. All the IAX hardphones I've seen don't seem to be the highest of quality honestly. ___ --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