Re: [asterisk-users] Nightly tarballs, would you use them?
Per Jessen wrote: Russell Bryant wrote: Greetings, During the past week, there have been some requests for nightly tarballs to help making testing new Asterisk code easier. There was some debate as to whether they would be useful. The reason that they may not be useful is because you can get equivalent access to new code just by accessing the subversion repository directly. However, for one reason or another, some people would prefer to have a tarball. If this was available, would you be interested in it? On occasion, yes. I think nightly tarballs could be quite useful. Whilst it's easy to check out from subversion directly, a nightly tarball provides a specific point of reference which can be helpful when trying to identify a problem. If we had a specific problem we were trying to fix, I would very likely grab the latest tarball and try it out. /Per Jessen, Zürich In subversion can you specify what revision you want to check out so it is equally easy to know what version you want to test. I can agree that a nightly tarball is a bit more spoon feeding for none developer people. And to create a nightly tarball is a script and a cron jobb so the resources to maintain it should be low. And for the poll, I would unlikely use the tarball. /Mats ___ -- 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 ports and CentOS firewall
Check this out: http://www.voip-info.org/wiki-Asterisk+firewall+rules dave cantera wrote: ed, this may be somewhat liberal but should do the trick... daveC -A RH-Firewall-1-INPUT -p tcp -m tcp --dport 69 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 69 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5061 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5062 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 4569 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m tcp --dport 5038 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5036 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5004 -j ACCEPT # -A RH-Firewall-1-INPUT -p icmp -m icmp --icmp-type any -j ACCEPT -A RH-Firewall-1-INPUT -p ipv6-crypt -j REJECT -A RH-Firewall-1-INPUT -p ipv6-auth -j REJECT -A RH-Firewall-1-INPUT -d 224.0.0.251 -p udp -m udp --dport 5353 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m state --state NEW -m tcp --dport 22 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m state --state NEW -m tcp --dport 80 -j ACCEPT -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited Ed Nunez wrote: If I enable the firewall on my Server, which ports should I open for Asterisk to work properly. Is it enough to just open the SIP ports? ___ -- 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 CLI and no such command stop
Vieri wrote: Hi, I'm probably missing something trivial but I don't understand what. Asterisk is loading fine but when I connect to the console (asterisk -vr) and type stop I get a no such command reply: *CLI help (...) skinny show lines Show defined Skinny lines per device soft hangup Request a hangup on a given channel unload Unload a dynamic module by name *CLI stop No such command 'stop' (type 'help' for help) # tail -n 1000 /var/log/messages | grep -i error Does anyone know why the stop command doesn't appear on the help list and is unavailable? Try *CLI help stop and that will show you the syntax for the commands that fails! I can agree that No such command 'stop' (type 'help' for help) is a bit misleading and should have been Wrong syntax or Incomplete command or something like that And always state witch * version you use! ___ --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 Hp servers
Anselm Martin Hoffmeister wrote: Am Samstag, den 05.01.2008, 19:50 + schrieb Gres +: please can anyone help me knowing if i can install Linux and Asterisk on HP servers Gres, you will have to find out if _YOU_ can do that. Generally speaking it is very well possible. For a quick start, you might want to try an asterisk-centric distribution that makes starting with Linux and Asterisk quite a bit easier than e.g. LFS, Debian, or Gentoo might. Anselm HP has certified two distros on their servers, http://h71028.www7.hp.com/enterprise/cache/321097-0-0-0-121.html Other server vendors has done the same. The distros is: » Novell SUSE LINUX Enterprise Server » Red Hat Enterprise Linux (RHEL) You can use CentOS.org instead of RHEL. And as an advice, ask your reseller or google for it before asking, this info is easily found! /Mats ___ --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] Detailed Instructions
Shane D wrote: Hello List, I am getting Asterisk set up. I am going to be installing Debian Linux on a laptop later. I would appreciate some detailed instructions on: (A) What to type into the shell to download and install Asterisk. (B) How to open the configuration files (*.conf) (C) If there is a way that I can change the configuration files remotely (SSH?). Thanks in advance. (A) http://www.asteriskdocs.org/ or # apt-get asterisk (B) # apt-get mc # mc (C) Yes SSH ___ --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 Hp servers
Tzafrir Cohen wrote: On Sat, Jan 05, 2008 at 11:57:04PM +0100, MatsK wrote: Anselm Martin Hoffmeister wrote: Am Samstag, den 05.01.2008, 19:50 + schrieb Gres +: please can anyone help me knowing if i can install Linux and Asterisk on HP servers Gres, you will have to find out if _YOU_ can do that. Generally speaking it is very well possible. For a quick start, you might want to try an asterisk-centric distribution that makes starting with Linux and Asterisk quite a bit easier than e.g. LFS, Debian, or Gentoo might. Anselm HP has certified two distros on their servers, http://h71028.www7.hp.com/enterprise/cache/321097-0-0-0-121.html Other server vendors has done the same. The distros is: » Novell SUSE LINUX Enterprise Server » Red Hat Enterprise Linux (RHEL) You can use CentOS.org instead of RHEL. But then you are not using a certified distribution. CentOS is an Enterprise-class Linux Distribution derived from sources freely provided to the public by a prominent North American Enterprise Linux vendor. CentOS conforms fully with the upstream vendors redistribution policy and aims to be 100% binary compatible. (CentOS mainly changes packages to remove upstream vendor branding and artwork.) CentOS is free. ___ --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 thougt
A more recent one is, http://www.iana.org/assignments/uri-schemes.html http://www.ietf.org/rfc/rfc4395.txt But I have seen more web pages with a callto: tag than with the tel: tag. But from what I have seen have the callto: tag been disregarded since there is a tel: tag. /Mats Tim Panton wrote: The official URI for this is tel: see http://www.ietf.org/rfc/rfc3966.txt It isn't implemented everywhere, but most cellphone browsers seem to. Tim On 4 Jan 2008, at 12:48, Dean Collins wrote: Snapanumber does this but with only certain browsers. (it doesn't work with ie which is what I use). Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of dave cantera Sent: Friday, 4 January 2008 2:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] A thougt dean, fredrik, when I installed skype, ugh, it asked me if I wanted to link phone numbers on the web page to be click2dial... I did it and every phone number on a web page was a link... I ended up turning it off... it was too annoying... so there are some plug-ins out there that can do that sort of thing... daveC Dean Collins wrote: I think Snapanumber might be what you are looking for. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Fredrik Söderlund Sent: Thursday, 3 January 2008 2:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] A thougt Is there any possibilletys to klick on a telephone nr an it will dail like the case in a mail program if you klick a url://a.b.se it opens a browser and in this case it would open a dailplane ?? Is there sucha thing ? Asking just out of curisoty /Fredrik Söderlund -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] OT - GEOPRIV and location based SIP services
Olivier wrote: Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is connected from office or home. Another option is to directly ask user or let them change default option but having this automatically detected is a bonus. Has anyone tried to build such location based SIP services ? I've read few lines about GEOPRIV which seems to be a building block for location based services but I could make sure if such DHCP extensions are implemented somewhere. Do you think GEOPRIV would help ? Regards Hi Oliver, Linux Journal had an article about timezone handling in asterisk with perlscript for checking the GeoIP database with the IP adr. from the location db. Maybe that could give you a clue how to solve your question. http://www.linuxjournal.com/article/9190 The challange with GEOPRIV is that its rarely used so I would recomend GeoIP, http://www.maxmind.com. /Mats ___ --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] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
Vincent wrote: On Tue, 01 Jan 2008 16:10:47 +0100, MatsK [EMAIL PROTECTED] wrote: And you know that you can convert the files to every codec format that is in use then will the cpu load be minimalized ! Yup, but the CPU is just a Pentium 233MHz. I just converted a 20MB WAV file from a CD-quality (44KHz sample rate, stereo) into the format Asterisk likes (8HKz, mono), and it took about 10mn. So conversion is out of the question, as Asterisk is likely to have a problem answering other incoming calls while it's busy converting the last voicemail message. Thats true, so I normaly convert files on my development box, don't wan't to disturb the production. And the conversion of prompts is a one time task! To re encode is a cpu load that happens every time the prompt is used! ___ --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 Power fail transfer possible with asterisk?
John covici wrote: OK, to clarify a bit, he wants to fix things so that all we are depending on are the pots lines -- I know if they go out you are gone. So what can we do in that case? There is ATA boxes with a port that you connect to a analog line and when SIP fails to register will it use the analog port. Some ATA boxes routes emergency calls direct to the analog port. I think that Sipura SPA-3000, http://www.sipura.com/products/spa3000.htm will cover your needs. on Wednesday 01/02/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote On Wednesday 02 January 2008 17:10:05 John covici wrote: Hi. I have a client who wants some way that his analog phones can call out even after the power is out and the UPS has died -- some way that a phone can connect directly to an fxo or some such when power is gone. Any hardware around which can do this? I have heard of some ATA's which do this, do any of the channel banks have this capability? 1) If his phones are this critical, he needs a triple redundant generator. 2) Ask him what he would like to do after 36 hours of power outage, when even the telco stops being able to provide battery on their POTS lines. If your provider is out, there's very little you can do. Perhaps a ham radio attached to a car battery? Speed costs; how fast would you like to go? -- Tilghman No need to be rude Mr Tilghman, try to be constructive. ___ --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] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
Vincent wrote: On Tue, 1 Jan 2008 17:23:29 +0530, Godson Gera [EMAIL PROTECTED] wrote: s,2,Playback(/usr/local/lib/asterisk/test_wav_out) And asterisk will automatically pickup the file that it can play with any asterisk supported format from the specified path. OK. Is there a way to tell Asterisk which codec to use so it doesn't try figuring out the file format used? Thanks. The codec is specified (for a sip device) in sip.conf, like this: [general] disallow=all allow=ulaw allow=alaw allow=gsm And you know that you can convert the files to every codec format that is in use then will the cpu load be minimalized ! To convert between different codec formats can you use the asterisk CLI command: file convert file_in.format file_out.format To convert from a shell script can you do like this: #!/bin/bash # Converts a audio file from alaw to a ulaw rasterisk -x file convert /tmp/file_in.alaw /tmp/file_out.ulaw More examples: The old way: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk I will try to update this page with convert. As a final touch, I have heard that sln should be the prefered format where you dont have the same format as the codec used in a channel. Asterisk native format is sln /Mats ___ --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] One Way Delay in Audio Over Analog
Brian Alexander wrote: I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian Brian, What about some facts ? Hardware ? Software versions ? /Mats ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Anthony Francis wrote: Axel Thimm wrote: On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote: Olle E Johansson [EMAIL PROTECTED] writes: But on the other hand, if people rely on third-party distributions we might want to set up some kind of peer pressure on the maintainers - and possibly identify them so we can support them and speed up their process. Third-party distributions are very important, and Asterisk has for various reasons done relatively badly there. Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk isn't even available in the most popular extra repositories, but only in ATrpms, my least favourite of the larger repositories. It happens to be my favourite thrid party repo though, ;) and indeed there is quite some asterisk support happening 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 is fairly easy to build, I don't see why it needs to be in a repo. IMO There are several benefits to have it in a repo. One is that it is a security issue, you don't want to have dev tools on a exposed server. Another is, if you have hundreds of similar machines, why compile Asterisk 100 times when you need to compile it once and then just copy the binaries to the other 99 machines. So as you see it is an advantage with repo's. Merry Christmas Mats ___ --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] best way for night ringer??
Doug Lytle wrote: BerkHolz, Steven wrote: Option 3 (I believe this is best, but am not sure where to start) When asterisk is in night mode, I'm doing option 3, menu item on the IVR to ring the night bell. Plays an awfully loud horn noise on the PA while ringing a phone out an the plant floor every 15 seconds. The breakroom and plant manager's phones are apart of the same pickup group. Any of them can do a *7 to grab an incoming call. exten = 4173,1,GotoIfTime(07:45-17:00|mon-fri|*|*?press-officehours,s,1) exten = 4173,n,System(/bin/cp /usr/local/bin/bullhorn.call /var/spool/asterisk/outgoing/bullhorn`date +%s`.call) exten = 4173,n,Dial(SIP/4173,15,tT) exten = 4173,n,Goto(analog-extensions,4173,1) Doug Dough, I see that you use cp to copy the call file to spool directory, that is not recommended, use mv instead since it is a atomic command whitch cp isnt. So an if you change it to : exten = 4173,1,GotoIfTime(07:45-17:00|mon-fri|*|*?press-officehours,s,1) exten = 4173,n,System(/bin/cp /usr/local/bin/bullhorn.call /temp/bullhorn.call) exten = 4173,n,System(/bin/mv /temp/bullhorn.call /var/spool/asterisk/outgoing/bullhorn`date +%s`.call) exten = 4173,n,Dial(SIP/4173,15,tT) exten = 4173,n,Goto(analog-extensions,4173,1) should solve it. /Mats ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Could this CLI syntax move over to the dev list, since it's mobing further away from the original question! /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] Upgrade to Asterisk 1.4 - it's one year's old!
Steve Edwards wrote: On Wed, 19 Dec 2007, Patrick wrote: On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Sorry but I'm not a native English speaker and I don't get it. Why is dogs show black the correct form as opposed to the imho more correct (in spoken language) show black dogs? It's not. I think it was a humorous reply to a humorous reply. The core bit should die, die, die. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 Please move this discussion away from this thread. Read Olles reply, that that has been discussed in the dev list so take it over 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
