[Asterisk-Users] Digium cards and Australian use.
Hi All, Are there any interested parties who would support us getting the Digium cards through the Australian testing. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of HQ Sent: Wednesday, 21 January 2004 11:58 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] how scalable is digium cards? Steven, what about if I want to make a 4x10 system? Should I have to move to E1/T1 anyway? Is that cost effective? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 8:49 AM Subject: Re: [Asterisk-Users] how scalable is digium cards? On Wed, 2004-01-21 at 00:40, lito lampitoc wrote: This might be a newbie question but I'm just wondering how would it be possible to have 30 analog lines using asterisk for PBX by just using TDM40B and X100P (or are there any device), if an ordinary PC support just 4 PCI slots? the maximum scale i guess would just be 2 x 8. Adding a new PC just for this purpose would be costly. I would appreciate your comments. It is unlikely you would want 30 analog lines coming in. It is likely that your telco would change to supporting a T1 or E1 based on your location and maybe they will then change it to analog at your premisis. When working with a PBX, you would want to take that T1 or E1 directly into the PBX without the analog conversion. So your 30 lines in would fit on a E1 if available or 2 T1s into a TE4XXP card. You are left with 2 more ports to move those channel banks inside for analog extensions. With 2 cards as a sane high limit, and the possibility of haveing those two cards be 4 port E1 cards, it is possible to have 240 lines split in some multiple of 30 between ins and outs. This also doesn't account for the VoIP options. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 101
--- dkwok [EMAIL PROTECTED] wrote: Just got GS 101 phone and plugged into the network. Peoplehere complain about these phones but I don't seem to have a problem, well not after getting them set up correctly. I'm running with Software Version: Program--1.0.4.39Bootloader--1.0.0.13 HTML--1.0.0.20 Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. I'm using SIP info also with payload type set to 101 2. there is no sound coming from the other end. For some reason I found I had to place the disallow=...allow=... stuff under [gs] putting it in [General] didn't seem to do the trick. I also put reinvite=no in [gs] I once had sound going only one way due to t stupid error in my firewall config. I was purposfully droping packets and logging each one of them. Are you running firewall software on your * server? ethereal or other ethernet sniffing software is usfull to debug this kind of stuff I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060 dtmf = sip info codec = pcmu codec = pcma Any pointer of a sample of config file would be most appreciate. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
Hello all, Per my last message to the list, and my promise to the Developers that I'd create RPMS if they released 0.7.0, I would like to announce the availability of experimental RPMS for Asterisk release 0.7.1. These are targeted at RedHat 7.3 systems, running the latest Kernel release (2.4.20-28.7). As the RPMS mature and people submit comments, changes, updates and patches, I will begin maintaining RedHat 8,9 and Fedora Core 1 RPMS... but we are a long way away! ;) I have absolutely -NO- idea if these will work, or if the modules will load, so if you feel like giving it a try, please let me know how it works out for you. The RPMS are available at: ftp://ftp.nacs.net/asterisk/RPMS/asterisk-0.7.1-1.i386.rpm ftp://ftp.nacs.net/asterisk/RPMS/kernel-module-zaptel-0.8.0-1_2.4.20_28.7.i386.rpm ftp://ftp.nacs.net/asterisk/RPMS/libpri-0.5.1-1.i386.rpm ftp://ftp.nacs.net/asterisk/RPMS/zaptel-0.8.0-1.i386.rpm The SRPMS are available at: ftp://ftp.nacs.net/asterisk/SRPMS/asterisk-0.7.1-1.src.rpm ftp://ftp.nacs.net/asterisk/SRPMS/libpri-0.5.1-1.src.rpm ftp://ftp.nacs.net/asterisk/SRPMS/zaptel-0.8.0-1.src.rpm The README: Asterisk 0.7.1 RPMS for RedHat 7.3 -- These are experimental RPMS for RedHat 7.3 based systems to install the Asterisk 0.7.1 Open Source PBX. I would caution people against using these on Production systems, as they have not yet been extensively tested, nor have they been optimized. Please help the cause by sending any changes, patches or updates to [EMAIL PROTECTED]. Install in the following order: rpm -Uvh libpri-0.5.1-1.i386.rpm rpm -Uvh zaptel-0.8.0-1.i386.rpm rpm -Uvh kernel-module-zaptel-0.8.0-1_2.4.20_28.7.i386.rpm rpm -Uvh asterisk-0.7.1-1.i386.rpm For configuration and getting started, point your favorit PDF reader at: http://www.digium.com/handbook-draft.pdf These RPMS are made possible from a combination of work that I have done and the excellent work of Tom Moertel (http://community.moertel.com) for the Zaptel RPM. Release: 1/22/2004 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards and Australian use.
G'day Dave! Yes, count me in. May just be moral support, but I'd love to help. Regards, Vic On Thu, 22 Jan 2004, David Hindmarsh wrote: Hi All, Are there any interested parties who would support us getting the Digium cards through the Australian testing. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
Greg Boehnlein wrote: Hello all, Per my last message to the list, and my promise to the Developers that I'd create RPMS if they released 0.7.0, I would like to announce the availability of experimental RPMS for Asterisk release 0.7.1. These are targeted at RedHat 7.3 systems, running the latest Kernel release (2.4.20-28.7). As the RPMS mature and people submit comments, changes, updates and patches, I will begin maintaining RedHat 8,9 and Fedora Core 1 RPMS... but we are a long way away! ;) I have absolutely -NO- idea if these will work, or if the modules will load, so if you feel like giving it a try, please let me know how it works out for you. The RPMS are available at: ftp://ftp.nacs.net/asterisk/RPMS/asterisk-0.7.1-1.i386.rpm ftp://ftp.nacs.net/asterisk/RPMS/kernel-module-zaptel-0.8.0-1_2.4.20_28.7.i386.rpm ftp://ftp.nacs.net/asterisk/RPMS/libpri-0.5.1-1.i386.rpm ftp://ftp.nacs.net/asterisk/RPMS/zaptel-0.8.0-1.i386.rpm The SRPMS are available at: ftp://ftp.nacs.net/asterisk/SRPMS/asterisk-0.7.1-1.src.rpm ftp://ftp.nacs.net/asterisk/SRPMS/libpri-0.5.1-1.src.rpm ftp://ftp.nacs.net/asterisk/SRPMS/zaptel-0.8.0-1.src.rpm The README: Asterisk 0.7.1 RPMS for RedHat 7.3 -- These are experimental RPMS for RedHat 7.3 based systems to install the Asterisk 0.7.1 Open Source PBX. I would caution people against using these on Production systems, as they have not yet been extensively tested, nor have they been optimized. Please help the cause by sending any changes, patches or updates to [EMAIL PROTECTED]. Install in the following order: rpm -Uvh libpri-0.5.1-1.i386.rpm rpm -Uvh zaptel-0.8.0-1.i386.rpm rpm -Uvh kernel-module-zaptel-0.8.0-1_2.4.20_28.7.i386.rpm rpm -Uvh asterisk-0.7.1-1.i386.rpm For configuration and getting started, point your favorit PDF reader at: http://www.digium.com/handbook-draft.pdf These RPMS are made possible from a combination of work that I have done and the excellent work of Tom Moertel (http://community.moertel.com) for the Zaptel RPM. Release: 1/22/2004 This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? For those who use RH or Fedora Core, RH9 is EOL in April and FC2 is scheduled for release in April as well.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Windows IAX Client
Peter, [Full quote deleted] Suggestion for name SwIAX based on Sokol W (windows) IAX I would not use that name, as there is a VoIP company called SWYX. You don't want to risk any problems there, do you. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Iaxclient-devel] New Windows IAX Client
Hi Steven, - Original Message - From: Steven Sokol [EMAIL PROTECTED] Announcing a new Windows-based IAX/IAX2 client. Please download it and give it a try. Let me know about any bugs, and any missing features. I have tried a little bit your soft phone. First comments: - Sometimes I cannot hangup the call made (clicking on Drop does nothing). Clicking several times on Drop/LINE1/LINE2 crashes the application. - many of the checkboxes captions are wrapped on my display (1024x768), Windows XP Pro, Windows Standard Theme - there is no Speaker available in the list as a ring device. How can you select it? Some features are very cool: - MWI - IAX native Transfer Keep up the good work. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Diax IAX2
Hi Mike, - Original Message - From: Michael Welter [EMAIL PROTECTED] I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to * are perfect. However, when calling from * to Diax, I get the following: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[mike]/3 of format GSM since our native format has changed to ULAW In iax.conf I have: allow=all disallow=g723.1 disallow=lpc10 allow gsm In this way you accept ULAW (G.711) as a valid codec for IAX, which is not the case. I am not aware of any IAX phone with another codec support than GSM. So the correct way to do it is: disallow=all allow=gsm Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 Licenses from Digium
zoa wrote: sure, Its not impossible to have g729 and scsi only systems, although several people with scsi systems have had issues with the g729 installation, i did not. That doesnt mean that g729 is rock stable, every now and then the license disappears or stops working for some hours/restarts. If you have a choice, i'd go for ilbc, sound quality is better, packetloss features are great At 22:28 21/01/2004 +, you wrote: zoa wrote: This is absolutely not true. I have 3 (raid) scsi asterisk machines in production. Joachim. At 11:32 21/01/2004 -0500, you wrote: In my view at least one IDE drive must be installed in order for * g729 license to work. To simplyfy, here is the matrix (This is how I think it is please confirm) IDE Disk Install - g729 coder work. IDE/SCCI interfaces. Only a SCSI disk installed - g729 will not work. IDE/SCSI Interfaces. At lease one IDE disk installed - g729 will work. SATA Serial ATA Disk I have no clue how it works. Is SATA considered a IDE disk or a SCSI disk ? This is an issue that VoiceAge need to address soon. - SamW -Original Message- From: Amaury Jacquot [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 4:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 Licenses from Digium Terence Parker wrote: OK - but what counts as a SCSI system? These days there are lots of pseudo-SCSI systems around - such as our server which runs a serial-ATA RAID but the driver is loaded as a SCSI device. Is that still IDE? Or SCSI? technically, it uses the SCSI command set over a serial link, so, it's SCSI Terence I know one thing for sure... G729 WILL NOT WORK after installation *(it never realy installs but does the segmentation faults), * will not start, and you will need to prevent g729 module from Starting in order for * to start. So do not buy if your box is SCSI in any part. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Can you please clarify which part are you referring as not being true? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ok, fair enough. My servers do not have IDE at all (not even CDROM), hence why My g729 installations failed. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
On Thu, 22 Jan 2004, WipeOut wrote: This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? Couple of reasons.. 1. It is a stable, known quantity that uses solid components and closely mirrors the environment that a lot of people develop Asterisk on. It isn't going to drastically change, so those wishing to deploy it in production may look to RedHat 7.3 as a stable platform for that purpose. 2. 8.0 and 9.0 are really not server oriented distributions of RedHat. RedHat started using a lot of edge technology in the later versions of RedHat (newer Glibc, newer GCC) and as a result, I know very few people (and I know a lot in my Industry ;) that are deploying commercial, production servers on top of RH 8 and 9. It's great for the DeskTop, but not in the Data center. As with all things, this is based on my personal opinions, so your mileage may vary! ;) 3. I want to refine the RPMS a bit and do some updates and changes to the .specfiles. If I have to maintain 7.3, 8.0, 9.0 and FC1 releases, that is 3 times the build work. Work will proceed a lot more rapidly if I just have to do a weekly update for one platform. 4. I run Asterisk on top of RH 7.3 currently and it suited my needs. ;) 5. I haven't yet built my 8.0, 9.0 and Fedora Core 1 development environments for Vmware, although the SRPMS that I released -SHOULD- build on them without modifications. For those who use RH or Fedora Core, RH9 is EOL in April and FC2 is scheduled for release in April as well.. See #5! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T400P / T100P with Hong Kong IDA-P Lines
Thanks Steve for the info. that was certainly very helpful! I shall see if PCCW will b*tch about it. Hopefully not! By the way, have you tried DID on IDA-P on any carriers in HK? NTT, HGC, NWT? David - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 3:53 AM Subject: Re: [Asterisk-Users] T400P / T100P with Hong Kong IDA-P Lines Hi David, David Liu wrote: Hi there, Anyone had any success deploying Asterisk with a T100P or T400P card in Hong Kong? To my understanding, Hong Kong carriers only provide IDA-P or IDA-M lines. I am looking to use IDA-P. Is this possible with the card? I know Cisco 2651XMV with a VIC card can do it. But that's just way too expensive! IDA-M is robbed bit signalling. It comes in several forms in HK which are functionally equivalent to the US, but use different names. * should work OK with these. IDA-P is the 5ESS ISDN protocol. Again * should work OK. I haven' t tried * with real PCCW T1 lines, but its unlikely there will be any problems. The Digium cards are not approved in HK, and PCCW make ask what cards you intend to attach (sometimes they do, and sometimes they don't). Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing List Lag
I get exactly the same thing, from here in Belfast, Northern Ireland. I've got a reliable box on the MCI/Worldcom/UUNet network with more than plenty of bandwidth. I'd be willing to host the list, I guess it just depends on how many emails/day the Asterisk list goes through... Steve On Wed, Jan 21, 2004 at 10:42:15PM -0500, mattf wrote: Hello, Ditto here, it seems to be the worst 9am to 5pm in the USA, any other time than that messages get posted right away. Ping times from both of my network connections to digium.com domains are horrible at 300-700ms but the last hop before entering the digium.com land is always really good 30-40ms I am just assuming here, but I'll bet that the fact that the cvs server and FTP server for Asterisk are running on the same connection may mean that emails get sent out at a much slower rate. I also bet that there are hundreds if not thousands of members of the mailing lists. I don't suppose someone on the list has a reliable machine on a large backbone network in some colo that has a lot of bandwidth to burn and would want to host a large listserv? Now for some useless info: Their list server (rattler.digium.com) seems to be running Redhat while their webserver(hoochie.digium.com) is running Debian MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 7:45 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mailing List Lag Has anyone from digium looked at why there is a 30 min to 3 hour lag on messages on this list? I.e looking at the last 50 messages I've received, the lag is about 90 minutes between the time sent and the time received. Sometimes this drops to as little as 4 minutes. Is this problem worse for me because my email address starts with w and my copies of the emails get sent after a-x? Cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and more analog lines?
Hi! Are the GIPS codecs now implemented with the Asterisk? If I need more analog lines, say around 30, what's the easiest way doing it? I checked the Mediatrix box with 24 connections, maybe that would be a good (and rel. cheap) way to go? Any other suggestions? The ports has to support fax machines. rgds, /staffan -- -- Staffan Kerker / KIT Communications, AerotechTelub mail: [EMAIL PROTECTED] Don't get involved in politics man, just play the gig... /Sgt. Floyd, Electric Mayhem Band ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: G.729 Licenses from Digium
Tilghman Lesher [EMAIL PROTECTED] said: Given the closed source nature of the code in question, it is impossible for anyone who does not have the source to be definitive in their answer. Hmm... If someone got $10 to burn and runs the installation procedure through 'strace -f' or similar? I bet they are using a system call, and I bet that with some smart system call redirection you can fool the software into thinking anything you like. (and in case of emergency, simply hack the appropriate kernel drivers ;-)). Copy protection always was braindead, but it has been completely, utterly and absolutely braindead since the 80386. (fond memories of flipping through 'Programming the 80386' and hacking DOS software that never knew it was running inside our homebrew virtualization setup ;-)) -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi: suppress calling number on outbound dialing?
Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the @ in front of the outbound msn in the dialstring. (Dial([EMAIL PROTECTED] vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI: Early-B3 working with AVM-B1?
Hi,
here is an update to my own post to this list.
Following an information from Philipp, I testet this with an passive AVM
card, but the same things happen. What am I doing wrong?
Is there something wrong with my extension.conf?
without Early B3:
exten = _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30)
with Early B3:
exten = _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30)
Thanks,
Karsten
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Re: [Asterisk-Users] New Windows IAX Client
Steven, - Integrated with the Eutectics IPP200 USB handset integration with handsets is a great. Do you support onhook/offhook for the IPP200? Do you plan on supporting other Eutectics phones as well, like the IPP5xx (with dial support) or the IPP210? -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
Greg Boehnlein wrote: On Thu, 22 Jan 2004, WipeOut wrote: This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? Couple of reasons.. 1. It is a stable, known quantity that uses solid components and closely mirrors the environment that a lot of people develop Asterisk on. It isn't going to drastically change, so those wishing to deploy it in production may look to RedHat 7.3 as a stable platform for that purpose. 2. 8.0 and 9.0 are really not server oriented distributions of RedHat. RedHat started using a lot of edge technology in the later versions of RedHat (newer Glibc, newer GCC) and as a result, I know very few people (and I know a lot in my Industry ;) that are deploying commercial, production servers on top of RH 8 and 9. It's great for the DeskTop, but not in the Data center. As with all things, this is based on my personal opinions, so your mileage may vary! ;) 3. I want to refine the RPMS a bit and do some updates and changes to the .specfiles. If I have to maintain 7.3, 8.0, 9.0 and FC1 releases, that is 3 times the build work. Work will proceed a lot more rapidly if I just have to do a weekly update for one platform. 4. I run Asterisk on top of RH 7.3 currently and it suited my needs. ;) 5. I haven't yet built my 8.0, 9.0 and Fedora Core 1 development environments for Vmware, although the SRPMS that I released -SHOULD- build on them without modifications. For those who use RH or Fedora Core, RH9 is EOL in April and FC2 is scheduled for release in April as well.. See #5! ;) I understand or agree with all of your points.. My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
WipeOut wrote: Greg Boehnlein wrote: On Thu, 22 Jan 2004, WipeOut wrote: This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? Couple of reasons.. 1. It is a stable, known quantity that uses solid components and closely mirrors the environment that a lot of people develop Asterisk on. It isn't going to drastically change, so those wishing to deploy it in production may look to RedHat 7.3 as a stable platform for that purpose. 2. 8.0 and 9.0 are really not server oriented distributions of RedHat. RedHat started using a lot of edge technology in the later versions of RedHat (newer Glibc, newer GCC) and as a result, I know very few people (and I know a lot in my Industry ;) that are deploying commercial, production servers on top of RH 8 and 9. It's great for the DeskTop, but not in the Data center. As with all things, this is based on my personal opinions, so your mileage may vary! ;) 3. I want to refine the RPMS a bit and do some updates and changes to the .specfiles. If I have to maintain 7.3, 8.0, 9.0 and FC1 releases, that is 3 times the build work. Work will proceed a lot more rapidly if I just have to do a weekly update for one platform. 4. I run Asterisk on top of RH 7.3 currently and it suited my needs. ;) 5. I haven't yet built my 8.0, 9.0 and Fedora Core 1 development environments for Vmware, although the SRPMS that I released -SHOULD- build on them without modifications. For those who use RH or Fedora Core, RH9 is EOL in April and FC2 is scheduled for release in April as well.. See #5! ;) I understand or agree with all of your points.. My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried www.gentoo.org . We are using it currently on a couple servers And it works great. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI: Early-B3 working with AVM-B1?
Hi Karsten,
are you sure your MSN is correct? If not T-Com will replace it
with your main MSN and probably will ignore the CLIR setting.
best regards
kapejod
--
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CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/
Am Do, 2004-01-22 um 09.00 schrieb Karsten Wemheuer:
Hi,
here is an update to my own post to this list.
Following an information from Philipp, I testet this with an passive AVM
card, but the same things happen. What am I doing wrong?
Is there something wrong with my extension.conf?
without Early B3:
exten = _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30)
with Early B3:
exten = _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30)
Thanks,
Karsten
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[Asterisk-Users] Standalone FXO device
Can anyone recommend me a fxo device with SIP or IAX functionality. I have tried with , http://www.clipcomm.co.kr/ They were worster than any device. Device itself costed me $270/- including shipping but not working. Kannaiyan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
Senad Jordanovic wrote: WipeOut wrote: I understand or agree with all of your points.. My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried www.gentoo.org . We are using it currently on a couple servers And it works great. Yes, I tried it the other day.. Its a little out of my legue i'm afraid, I would not be confident that I would be able to economically administer a system on Gentoo.. and the 2h30min install to bet to a basic bootable system when installing from the CD's is a bit of a nightmare as well if you have a server go down on you and need to get it back up quickly.. I am not knocking it, its probably a good distro if you are competent to that level with Linux.. I unfortunately am not.. I am considdering installing it on my system at home to try and improve my low level Linux skills.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Data calls (ISDN/64k) through * PRI
i all, is it possible to switch data calls through asterisk with the Dial application? The scenario is as following: PSTN (ISDN 64k) -- Asterisk/PRI(TE410P) --- (same) Asterisk/PRI --- PSTN (ISDN 64k) I tried this with normal Dail, but if you come with ISDN/64k, the outgoing call is an audio call. Any ideas ? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)
Jeff Gustafson ([EMAIL PROTECTED]) wrote:
Maybe it's not the new chan_sccp code that's the problem. When I put
in the SEP000785532D5F.cnf.xml on the tftp server, the phone never gets
to a usable screen. Instead it just tries to tftp files over and over.
Th one file, P00305000300.bin, I don't have. As far as I know I can't
get this file unless I buy it from cisco. Is this file absolutely
required for the chan_sccp code to work?
no, its not necessary required. in this case, check that the contents of
OS79xx.TXT if they match with your current version.
Also: check your SEP*.xml resp. xmlDefault File for the software
setting, so that this matches your current installed software.
You can find the current software load somewhere in the menus.
--jan
...Jeff
On Wed, 2004-01-21 at 14:05, Jan Czmok wrote:
Kewl, I was apparently trying to use older chan_sccp code which didn't
work.
Okay... just tried your new code. The phones keep resetting:
Error Verifying Config Info
then
Registering
you should use the CURRENT code, which is not there as a tarball.
i just posted the recent tarball to the /files directory.
use this one !
Then:
Configure your dhcp to serve it like this:
host voip-phone
{
hardware ethernet 00:30:94:C2:89:0B;
fixed-address 212.20.150.206;
option host-name voip-phone;
option domain-name-servers 212.20.144.98;
option routers 212.20.150.1;
option tftp-server-name 193.138.116.111;
}
by using your IPs.
The tftp server should contain:
- xmlDefault.CNF.XML file.
- a symlink from the xmlDefault.CNF.XM file to the SEPxx.cnf.xml file
- within the sepdefault, you need to define the callmanager.
- some more stuff, but we'll see it later...
this should be sufficient to bring up the asterisk withthe 79xx.
--jan
--
Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc.
Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Any ideas ?
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
exten = _X.,50,Dial(Zap/g3d/${EXTEN})
exten = _X.,100,Dial(Zap/g3/${EXTEN})
--
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
pgp0.pgp
Description: PGP signature
[Asterisk-Users] DIAX CallMe feature
Hi all, I kindly ask people who wants to leave me a voice message using the CallMe feature in DIAX to leave an e-mail address too, not only a phone number. I am located in Romania, Europe (GMT+2) and you must take into consideration the difference in time zone. It can be difficult for me to call you at the requested hours. In this way I can easily help you solve all your technical problems regarding DIAX. Thank you for your understanding, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)
Michael Devenijn ([EMAIL PROTECTED]) wrote: Jan, Where can we get any technical documentation about sccp protocol i've searched with google and at cisco but i don't find anything useful ... The only useful resource is imagination :-) Skinny is a Protocol developed by Selsius. Selsius has been bought by Cisco. All what is known is by reverse-engineer using ethereal, tcpdump and the known protocol info within ethereal sources. Cisco is currently not willing to provide more information about the Skinny Protocol :-( --jan -- Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] chan_capi: suppress calling number on outbound dialing?
Sascha Knific wrote: I never had the time to try out CLIR. Now I did and it doesn´t work for me as well. Make sure you have CLIR enabled by your telekom provider (Fallweise Unterdrückung der Rufnummer). It was not enabled on my MSNs, so @ didn't work. Now my provider has enabled CLIR and everything is working as kapejod has said it would. :) rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7
Martin Pycko wrote: You have to contact www.openss7.org. The site may look dead but they sell ss7 together with asterisk. Yes and no. The sell access to the SS7 CVS. It does not work with Asterisk. There is a project page about OpenSS7 - Asterisk integration, but it is a project that never went anywhere. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] R2 or EM for E1 CAS pbx to pbx link
hi, thanx for the response. I just tried to work on R2 CAS but i found that the libr2 has not been implemented well and tested. I think in addition to EM, R2 can also be used in a pbx to pbx E1 link. what do tou suggest Sam ?? About the R2 implementation for asterisk i have seen in the list that steve has implemented 95% of that...but we dont see any release of that. any current info on R2 development?? and Sam you are right i don't have the CAS table of the other switch. But i think i can get one. help me out in this. I have to make a E1 pbx to pbx connection using CAS. thanks in advance janjua Help STOP spam with the new MSN 8 and get 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queue with no agents - Congestion or voicebox instead of MOH?
Hi all, I have successfully set up a call queue with agents and agentCallbackLogin. Works fine, but if no agent is logged in incoming callers get music-on-hold forever (or until some timeout). Is it possible to play congestion tone without answering the call (and thus causing costs to PSTN callers) or send them to unvailable-mailbox directly to leave a message if no agents are logged into the queue? Thanks and best regards, Jan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Thanks Maik,
i try it
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Maik
Schmitt
Gesendet: Donnerstag, 22. Januar 2004 11:21
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Any ideas ?
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
exten = _X.,50,Dial(Zap/g3d/${EXTEN})
exten = _X.,100,Dial(Zap/g3/${EXTEN})
--
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
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Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
exten = _X.,50,Dial(Zap/g3d/${EXTEN})
Now, that is neat. Thanks for pointing this out.
Any chance one such distinction can be made on incoming calls as well
i.e. branch incoming calls on a single DID depending on whether they are
data or speech?
Thilo
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Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)
Jan Czmok wrote: Michael Devenijn ([EMAIL PROTECTED]) wrote: Jan, Where can we get any technical documentation about sccp protocol i've searched with google and at cisco but i don't find anything useful ... The only useful resource is imagination :-) Skinny is a Protocol developed by Selsius. Selsius has been bought by Cisco. All what is known is by reverse-engineer using ethereal, tcpdump and the known protocol info within ethereal sources. Cisco is currently not willing to provide more information about the Skinny Protocol :-( They may not give out the skinny spec., but it seems they do licence it. Ipblue make skinny softphones, certified by Cisco. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Making a call with sample.call
On Wed, Jan 21, 2004 at 08:07:55PM +0100, Philipp von Klitzing wrote: You could also insert a Wait(1) to make sure that the VoIP connection has been correctly established. If your soundfile is short, then maybe it was indeed played before the RTP stream was properly set up. For testing change 3,Hangup into 3,MusicOnHold or something similar that does give sound feedback for quite a while. It seems to start playing the file before the call is answered by the 'callee'. I'm calling a cell phone, not sure if it would work better with a land line, but it shouldn't matter. Has anyone else experienced this or have any suggestions as to how I could fix it!? Cheers :) -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Any chance one such distinction can be made on incoming calls as well
i.e. branch incoming calls on a single DID depending on whether they are
data or speech?
That's what the first line does.
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
${CALLTYPE} can be SPEECH, DIGITAL, RESTRICTED_DIGITAL, 31KAUDIO,
7KAUDIO or VIDEO.
--
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
pgp0.pgp
Description: PGP signature
RE: [Asterisk-Users] Re: Digium X100P for $43
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Wednesday, January 21, 2004 11:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Digium X100P for $43 for the record, mine has the same fcc id number as the Digiums. Is this typical for copied hardware, or is there something a little fishy going on here? No, nothing fishy. It's a WinModem. Digium didn't make it to begin with. It's commodity hardware. You can get them for $14-19 a piece. But that's just not the right thing to do. Asterisk development is paid for in part by sales of this hardware. Buy it from Digium, and you get support as well. I had a problem compiling the zap drivers when I got mine. When I called, the phone was picked up immediately, by a real person who knew exactly what they were talking about. Digium support actually SSHed into my box and fixed it/showed me what I was doing wrong. The support is well worth the price, especially if you are building a production server. Or if your time is worth anything at all for that matter. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI: Early-B3 working with AVM-B1?
Hi Klaus-Peter, Klaus-Peter Junghanns wrote: Hi Karsten, are you sure your MSN is correct? If not T-Com will replace it with your main MSN and probably will ignore the CLIR setting. best regards kapejod Thanks for the reply, but in this thread my problem is the Early-B3, not CLIR. The MSN I used, is one of my own (not the first one), they are correct and will be signaled to my mobile when testing, but I got no ring signal. The mobile shows the correct number. If I press the red button to discard the call, there are many messages in the debug (see my first post in this thread). I thought it was a problem of the card, cause Philipp told me, it works for him with Fritz!-card. So I tested again with aN ISA Fritz! card with the same result... Hope You have any ideas... Thanks Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] chan_capi: suppress calling number on outbound dialing?
Hi Peer Oliver Peer Oliver schmidt wrote: Sascha Knific wrote: I never had the time to try out CLIR. Now I did and it doesn´t work for me as well. Make sure you have CLIR enabled by your telekom provider (Fallweise Unterdrückung der Rufnummer). It was not enabled on my MSNs, so @ didn't work. Now my provider has enabled CLIR and everything is working as kapejod has said it would. :) thanks for Your reply. I checked it with all of my msn's, it doesn't work. But I will check, if Fallweise Unterdrückung... is enabled. I thought my Komfort-Anschluss has enabled it by default, but who knows ;-). I'll report the result here. HAND Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Switchboard interface
I am looking to produce a switchboard interface - hopefully web based I needs to: Show the logged in user the CLI of the call they are currently dealing with Show the number of calls in the queue Give a number of options for working with the call transfer put on hold etc. for transfer it must provide a list of users to transfer to. Allow a call to be initiated Can anyone give me ideas on how I might interact with * to get this informations and provide these services. Thanks Chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Description of Manager events
Hello, does anybody have a list of asterisk Manager events and what they mean? For examples such events as Rename? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Licenses from Digium
Just thought I'd mention that it's REALLY confusing when there is a combination of top and bottom posting, and nobody bothers to trim the posts including multiple copies of the list info footers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
WipeOut wrote: Senad Jordanovic wrote: WipeOut wrote: I understand or agree with all of your points.. My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried www.gentoo.org . We are using it currently on a couple servers And it works great. Yes, I tried it the other day.. Its a little out of my legue i'm afraid, I would not be confident that I would be able to economically administer a system on Gentoo.. and the 2h30min install to bet to a basic bootable system when installing from the CD's is a bit of a nightmare as well if you have a server go down on you and need to get it back up quickly.. I am not knocking it, its probably a good distro if you are competent to that level with Linux.. I unfortunately am not.. I am considdering installing it on my system at home to try and improve my low level Linux skills.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, it is a bit of pain, while installing. However, once installed It is realy easy to use. I can set you up, a VDS system to try it if you wish? (It may take few days to set it up though) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Standalone FXO device
Kannaiyan Natesan wrote: Can anyone recommend me a fxo device with SIP or IAX functionality. I have tried with , http://www.clipcomm.co.kr/ They were worster than any device. Device itself costed me $270/- including shipping but not working. Kannaiyan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I am talking to Andrew from clipcomm with intention of getting some samples. Would you be able to share your experince of not is working and any other related issues? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mp3player not working
Problem solved found that Asterisk is calling mpg123 to playback mp3s which isnt installed on Slackware 9.1 by default. Downloaded mpg123 source from http://www.mpg123.de/ and compiled with make linux; make install and now working. Also discovered that mpg123 doesnt seem to playback mp3s with ID3 tags in them, so strip them out before copying them to your Asterisk box. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Thursday, 22 January 2004 11:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] mp3player not working Hi, Im running the latest Asterisk (built last Saturday) and cant get mp3s to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I dont hear any sound, and the following displays on the console: -- Executing Answer(SIP/931-0efa, ) in new stack -- Executing Wait(SIP/931-0efa, 1) in new stack -- Executing MP3Player(SIP/931-0efa, /var/lib/asterisk/mohmp3/sample-hold .mp3) in new stack Jan 22 11:12:32 WARNING[442386]: rtp.c:375 ast_rtp_read: RTP Read error: Resourc e temporarily unavailable Jan 22 11:12:35 NOTICE[442386]: app_mp3.c:93 timed_read: Selected timed out/erro red out with 0 -- Executing Wait(SIP/931-0efa, 20) in new stack == Spawn extension (local, 902, 4) exited non-zero on 'SIP/931-0efa' The IP phones Im using are Cisco 7940 running G.729a. I have successfully licenced and registered 2x channels of g729 codec (running the new_codec_binary from ftp.digium.com) today, and have no problems checking my voicemail on Asterisk or dialing out through IAXtel or receiving calls. Even when I was running g711ulaw codec on the phones I had the same problem. Is there another dependency that is required for mp3playback in Linux? Is a soundcard required? My Linux box is running Slackware Linux 9.1. Any help to point me in the right direction to getting mp3playback and my music on hold working would be greatly appreciated. Thanks in advance, Chris Lee
[Asterisk-Users] asterisk 0.7.1 - mysql
Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
On Thu, 2004-01-22 at 12:44, Maik Schmitt wrote: That's what the first line does. You are right. Time to blame my lack of caffeine, I guess. Thanks, Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] chan_capi: suppress calling number on outbound dialing?
Sascha Knific wrote: I never had the time to try out CLIR. Now I did and it doesn´t work for me as well. Make sure you have CLIR enabled by your telekom provider (Fallweise Unterdrückung der Rufnummer). It was not enabled on my MSNs, so @ didn't work. Now my provider has enabled CLIR and everything is working as kapejod has said it would. :) I called the telecom provider (T-Com). They told me that my number is set to be always suppressed as I refused to be listed in the telephone directory. The funny thing is that nevertheless my number got always passed by default to the called party no matter what phone or pbx I used... I asked them to change it. Let´s see what happens... Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany Leo +49-8151-773261WGS84: N57°59,875' E011°20,568' [EMAIL PROTECTED] http://www.k-sysdes.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gsm + snom phones
Hi. I'm not using snom phones for a while, but now I want to test again them and I'm gonna buy a snom 200 105 . Some times ago I had a snom 100 , and gsm wasn't working with *. How's now the situation? the snom gsm works well with * ? Thanks for any info, Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
tried gentoo as well, followed all the pages up to point 16 where the kernel is installed tried the 'emerge' command, but to my dissapointment it could find emerge! (and it probably took me 2 hours to get there) David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Senad Jordanovic Verzonden: donderdag 22 januari 2004 13:35 Aan: [EMAIL PROTECTED] Onderwerp: RE: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released WipeOut wrote: Senad Jordanovic wrote: WipeOut wrote: I understand or agree with all of your points.. My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried www.gentoo.org . We are using it currently on a couple servers And it works great. Yes, I tried it the other day.. Its a little out of my legue i'm afraid, I would not be confident that I would be able to economically administer a system on Gentoo.. and the 2h30min install to bet to a basic bootable system when installing from the CD's is a bit of a nightmare as well if you have a server go down on you and need to get it back up quickly.. I am not knocking it, its probably a good distro if you are competent to that level with Linux.. I unfortunately am not.. I am considdering installing it on my system at home to try and improve my low level Linux skills.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, it is a bit of pain, while installing. However, once installed It is realy easy to use. I can set you up, a VDS system to try it if you wish? (It may take few days to set it up though) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN gateway
Hello Has anyone come across a small residential PSTN gateway? Its not worth running a * just as a PSTN gateway as it requries a seperate system / power / etc... I am looking for a device that could connect to * and a pstn line so that i could register that device to * and make pstn calls via that device. Regards Deepak
AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Hi Maik,
is there any special version from libpri or asterisk necessary since it
works ?
I'am runnig version: CVS-11/11/03-11:49:55 and it don't work :-(
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Maik
Schmitt
Gesendet: Donnerstag, 22. Januar 2004 12:45
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Any chance one such distinction can be made on incoming calls as well
i.e. branch incoming calls on a single DID depending on whether they are
data or speech?
That's what the first line does.
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
${CALLTYPE} can be SPEECH, DIGITAL, RESTRICTED_DIGITAL, 31KAUDIO,
7KAUDIO or VIDEO.
--
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
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RE: [Asterisk-Users] Starting with MGCP and Asterisk
I re installed Linux on my machine, and re installed Asterisk... so now is working properly. I was having some compilation problems... I'm still having my initial question. How do I set two call agents in the configuration files? How is the extensions.conf for MGCP?! Thanks in advance! Best regards Ricardo -Mensaje original- De: Girish Gopinath [SMTP:[EMAIL PROTECTED] Enviado el: Miércoles, 21 de Enero de 2004 02:51 p.m. Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Starting with MGCP and Asterisk Ricardo, I think that maybe the asterisk.conf file is missing?.. where i canf find a sample for this file? Run: make samples Did you read the message displayed by Makefile after installing Asterisk? Girish _ Add glamour to your desktop. Let your screen sizzle. http://server1.msn.co.in/msnchannels/Entertainment/wallpaperhome.asp Download the hottest wallpapers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 0.7.1 - mysql
Dawid Mielnik wrote: Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave _ Did you rememebr to build the Asterisk-Addons??.. The MySQL support has removed from the Asterisk core a while back and is now in asterisk-addons on the CVS server.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
On Thu, 22 Jan 2004, WipeOut wrote: I understand or agree with all of your points.. My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( I would suggest either Fedora Core 1 (Which is essentially RedHat 9.1) if you are familiar with RedHat. Or, just bite the bullet and learn about Debian. It's really a wonderful distribution once you learn the ins and the outs. In fact, I actually use apt to manage my RedHat 7.3 boxes, since upgrades are as simple as apt-get dist-upgrade. And just because RedHat isn't supporting 7.3 doesn't mean that others will not. There are several commercial vendors that have announced support for it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN gateway
- Original Message - From: Deepakumar JV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 8:37 AM Subject: [Asterisk-Users] PSTN gateway Hello Has anyone come across a small residential PSTN gateway? Its not worth running a * just as a PSTN gateway as it requries a seperate system / power / etc... I am looking for a device that could connect to * and a pstn line so that i could register that device to * and make pstn calls via that device. I'm confused. Do you want to get rid of *, or not? It sounds like you're just looking for an IP phone to pstn gateway service. See: vonage, voicepulse, etc... - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 0.7.1 - mysql
- Original Message - From: Dawid Mielnik [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 7:48 AM Subject: [Asterisk-Users] asterisk 0.7.1 - mysql Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? MySQL support was moved out to addons. You don't have to use ODBC to point to MySQL, but I would say it's probably a good idea. There is no guarantee that anyone will continue to update the mysql code now that the license on mysql has changed. You can also look into the postgresql support, as it is still in there. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gsm + snom phones
About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something has changed. -- Pertti Matteo Brancaleoni wrote: Hi. I'm not using snom phones for a while, but now I want to test again them and I'm gonna buy a snom 200 105 . Some times ago I had a snom 100 , and gsm wasn't working with *. How's now the situation? the snom gsm works well with * ? Thanks for any info, Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
my previuos message last words should be could NOT find emerge! -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens David Luyens Verzonden: donderdag 22 januari 2004 14:37 Aan: [EMAIL PROTECTED] Onderwerp: RE: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released tried gentoo as well, followed all the pages up to point 16 where the kernel is installed tried the 'emerge' command, but to my dissapointment it could find emerge! (and it probably took me 2 hours to get there) David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Senad Jordanovic Verzonden: donderdag 22 januari 2004 13:35 Aan: [EMAIL PROTECTED] Onderwerp: RE: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released WipeOut wrote: Senad Jordanovic wrote: WipeOut wrote: I understand or agree with all of your points.. My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried www.gentoo.org . We are using it currently on a couple servers And it works great. Yes, I tried it the other day.. Its a little out of my legue i'm afraid, I would not be confident that I would be able to economically administer a system on Gentoo.. and the 2h30min install to bet to a basic bootable system when installing from the CD's is a bit of a nightmare as well if you have a server go down on you and need to get it back up quickly.. I am not knocking it, its probably a good distro if you are competent to that level with Linux.. I unfortunately am not.. I am considdering installing it on my system at home to try and improve my low level Linux skills.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, it is a bit of pain, while installing. However, once installed It is realy easy to use. I can set you up, a VDS system to try it if you wish? (It may take few days to set it up though) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
is there any special version from libpri or asterisk necessary since it works ? I'am runnig version: CVS-11/11/03-11:49:55 and it don't work :-( We have CVS-11/24/03-12:12:10 -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature
RE: [Asterisk-Users] asterisk 0.7.1 - mysql
WipeOut, nope, did not build asterisk-addons thanks.. regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Thursday, January 22, 2004 2:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk 0.7.1 - mysql Dawid Mielnik wrote: Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave _ Did you rememebr to build the Asterisk-Addons??.. The MySQL support has removed from the Asterisk core a while back and is now in asterisk-addons on the CVS server.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and rh9 boot problem
Hi All! I installed * in RH9 with yesterday cvs and i have a x100p in that system. My problem is that when rh9 loads, it loads the zaptel modules ( wcfxo and the usb driver) automagically, and when it calls my rc.local with: modprobe zaptelmodprobe wcfxosafe_asterisk asterisk dont start. I don´t need the usb module because i only have the x100p in the system... anyone knows why it loads in the boot? and how can i stop it? In the previuos version with RH8 it only loads with the rc.local...i´m confuse. thanks Miklos
Re: [Asterisk-Users] PSTN gateway
Sorry for confusing.. let me explain ideally i want to have two * running, one at my place and the other at a remote location. Now the problem in running * at a remote location is the effort / cost involved in setting up / maintaining the * box. Hence i was looking for a device that could register with * (as a client so that i could dial a number and reach it as a normal extension) and also have a PSTN connectivity at the remote location. The reason i need PSTN connectivity at remote location is to make outbound calls from * via the device so called PSTN gateway. If i am still not clear, then in simple terms, i am looking for a hardware device with one FXO port and SIP support. Any help or suggestion please Thanks in advance Deepak - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 01:58 PM Subject: Re: [Asterisk-Users] PSTN gateway - Original Message - From: Deepakumar JV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 8:37 AM Subject: [Asterisk-Users] PSTN gateway Hello Has anyone come across a small residential PSTN gateway? Its not worth running a * just as a PSTN gateway as it requries a seperate system / power / etc... I am looking for a device that could connect to * and a pstn line so that i could register that device to * and make pstn calls via that device. I'm confused. Do you want to get rid of *, or not? It sounds like you're just looking for an IP phone to pstn gateway service. See: vonage, voicepulse, etc... - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Standalone FXO device
SJ, I'm also dealing with Andrew, they were good at telling you stories but nothing professional with the product. I registered with fwd and started dialling 14551 my fauvorite where i get clear voice. It gave me with completely noisy sound, I tried to reduce and increase the gain, but nothing works. I primarily want to share my DSL connected PSTN line to other members, so other members can use my PSTN minutes for free. But when I connect with the Clipcomm device, my DSL gets down and it gets up only when I switch remove the line from the device. I dunno what kind of problem it is. I left with that, Next I want to try to other networks by connecting ATA 186 FXS port to it. It works sometimes and just holds the line without hanging it up. I need to switch it off to get the line hooked on. Something very strange. The overall performance of the devices just sucks, I use the model CG-101E. If you need the device, I can ship you the one which I have got with me, since I'm no more interested in having it. I don't mind paying higher, but I'm looking for a quality device. If you can suggest anything, please share it to all in the list. Kannaiyan - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 12:38 PM Subject: RE: [Asterisk-Users] Standalone FXO device Kannaiyan Natesan wrote: Can anyone recommend me a fxo device with SIP or IAX functionality. I have tried with , http://www.clipcomm.co.kr/ They were worster than any device. Device itself costed me $270/- including shipping but not working. Kannaiyan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I am talking to Andrew from clipcomm with intention of getting some samples. Would you be able to share your experince of not is working and any other related issues? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gsm + snom phones
Hi. About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something but you also tested gsm ? Greets,Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? Couple of reasons.. 1. It is a stable, known quantity that uses solid components and closely mirrors the environment that a lot of people develop Asterisk on. It isn't going to drastically change, so those wishing to deploy it in production may look to RedHat 7.3 as a stable platform for that purpose. I agree, keep up the good work. I personally don't see any reason to upgrade atleast until the 2.6.x kernel is well underway. Maybe that's just me, hell I'm still running a 4.11 Novell server and a SCO Open server that hasn't been touched since y2k upgrades. Also if you look around for stable/available drivers from manufactures you'll find mostly 7.3 and some 8.0 supported drivers. Just try to call a manufacture and tell'em your having problems running their hardware with the newest greatest version of x.x.x, but if you're using one of their supported drivers you'll get the support you need. So moral of the story, always check with the hardware manufacture and stay with supported distributions. Just my .02 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3328334778' ] Called Number (len=11) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '63494441' ] -- Making new call for cr 5635 -- Processing Q.931 Call Setup -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 108 (Calling Party Number) -- Processing IE 112 (Called Party Number) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 38403/0x9603) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Accepting call from '3328334778' to '63494441' on channel 30, span 2 -- Executing GotoIf(Zap/61-1, 0?50:100) in new stack -- Goto (pri2,63494441,100) -- Executing Dial(Zap/61-1, Zap/g2/033283077733SPEECH) in new stack -- Making new call for cr 39439 Protocol Discriminator: Q.931 (8) len=50 Call Ref: len= 2 (reference 6671/0x1A0F) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '3328334778' ] Called Number (len=21) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '033283077733SPEECH' ] -- Called g2/033283077733SPEECH Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 39439/0x9A0F) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (Channel Identification) beroasterisk*CLI Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (Cause) -- Channel 30, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap sending, peerstate Overlap Receiving Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6671/0x1A0F) (Originator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/32-1' == Spawn extension (pri2, 63494441, 100) exited non-zero on 'Zap/61-1' Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 39439/0x9A0F) (Terminator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
[Asterisk-Users] Cause of transfer problem (GRANDSTREAM!)
It turns out that the cause of the transfer problem is the Grandstream 1.0.4.39 firmware. I was shipped a bunch of HandyTone-286 devices that contained the 1.0.4.30 firmware. This version had a bug where the phone would sometimes not ring at all. I was told by Grandstream to upgrade to the 1.0.4.39 version. This broke the Use # as Dial Key option, and evidently transfer as well. I still do not have any problems with my 1.0.3.81 phones, but I've read that I cannot downgrade from a 1.0.4x version to a 1.0.3x version. I'm pretty pissed that they shipped me what I consider to be defective devices, do not give me a way to back down to a usable version, and do not have a fix for this problem that makes all of the devices completely unusable to me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2588 - 11 msgs
Message: 5 To: [EMAIL PROTECTED] From: Doug Meredith [EMAIL PROTECTED] Date: Wed, 21 Jan 2004 20:05:19 -0400 Organization: Skyridge Systems Inc. Subject: [Asterisk-Users] Re: What technology could my phone company be using? Reply-To: [EMAIL PROTECTED] Mark Hazlewood [EMAIL PROTECTED] wrote: Sounds like Centrex services, we had it from Telus in Alberta a few years ago. I believe this is used for Centrex. I thought Centrex was basically a CO-hosted PBX. Is it also a local-loop technology? Are there PCI cards or SIP gateway boxes available? You can think of Centrex as a virtual PBX residing within a LEC or CLEC switch. It gives you most of the functionality of an on premise PBX delivered over local loops from the LEC or CLEC switch. This allows three or four digit dialing between extensions and no charges incurred for calls unless the user dials 9 for an outside line. Centrex can use POTS or ISDN desksets or a mixture of both. For instance, individual users can have POTS desksets but a receptionist could have an ISDN deskset to provide programmable keys for multiline answer and transfer. ADSI desksets can also be used to provide this functionality. I assume you are questioning the availability of Centrex compatible hardware for a system that doesn't employ Asterisk, as Centrex would be redundant when connected to Asterisk. Any voice capable PCI modem card can be used to terminate a Centrex POTS loop. ADSI capability may exist, as a Winmodem DSP could be programmed to handle voice and ADSI, but I haven't seen any applicable drivers/software. ISDN Centrex loops can be terminated on a PCI ISDN card and software is available to provide voice functionality. If you want a SIP gateway, you might as well use Asterisk and non-Centrex loops. There is no sense in duplicating PBX functionality and paying the monthly Centrex charges. Regards, George Bean Puwaba Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Canada's Primus introduces SIP localserv ice
If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David [EMAIL PROTECTED] 1/21/2004 6:39:34 AM I'm not sure Primus uses SIP. I think it's MGCP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Liu Sent: Tuesday, January 20, 2004 9:16 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: Canada's Primus introduces SIP localservice Hey Colin, Do let me know if Primus' SIP service can work with Asterisk. We tried setting it up like how you would for iconnecthere However, we even failed to register in the first place! (Of course password and username are correct). Anyone else on the list successfully used Primus' SIP with Asterisk? David [EMAIL PROTECTED] 1/20/2004 12:25:50 PM Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign up, they ship you this Dlink puppy for free: ftp://ftp10.dlink.com/pdfs/products/DVG-1120/DVG-1120_ds.pdf It has 2 FXS ports + ethernet + POTS backup port My order's in already, I'll be pleased to tell Telus where to put their value pricing once I get it installed. If anyone in Canada wants to know my experiences with it, email me off-list next month. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gsm + snom phones
Yes, absolutely. sorry, I was unclear .. -- Pertti Matteo Brancaleoni wrote: Hi. About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something but you also tested gsm ? Greets,Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Digium X100P for $43
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 3:48 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Digium X100P for $43 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Wednesday, January 21, 2004 11:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Digium X100P for $43 for the record, mine has the same fcc id number as the Digiums. Is this typical for copied hardware, or is there something a little fishy going on here? - I looked at the site of www.digitnetworks.com today. The thing I noticed was that nowhere on the site they listed the real (registered) company name or mention their address. One line has a (801) phone number listed otherwise only email addresses. Whois revealed the following: Registrant: Domains by Proxy, Inc. 15111 N Hayden Rd., Suite 160 PMB353 Scottsdale, Arizona 85260 United States Again no name, no address, sucessful way of hiding their identity. I stick with Digium, I know who they are, where they are and what I am getting, and I am supporting the developmnet of *. Alfred. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background Noise
Just to add some info from recent experience. May help, May not 1 X100P 2 X 4 port TDM400P Had to hook a dial-in palm PDA base for a custom software implementation to my * system and have the modem dial out and work properly. Phone connects to second port on PDA Base Experienced very bad electrical type noise on the line, hum, buzz, fad in and out. would come and go. Switched ports, wires, rxgain and txgain changes, phone changes, nothing helped the PDA base is also connected via serial port (or USB did not matter) to desktop computer for sync purposes Through trial and error. Found noise coming from connection to desk top computer. On, Off did not matter. Resolution. The power strip surge protector I was using on the desktop computer has two modes for noise filtration built in. 75 Hertz (or something, don;t remember) and 50 Hertz. I moved the plug for the desktop from the 75 side to the 50 side, All noise on line now gone. Not sure if this helps, constant noise on all sides may be power and noise filtration related... --- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi All i have some background noise problem with * and a diva srv 4bri + chan_capi 0.3.0 + X-Ten PRO on my pc. Both in incoming and outgoing call have a background noise. there is some tuning to do? where can i find documentation about capi.conf? which is the best codec for sip (ulaw, alaw, gsm...)? mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Canada's Primus introduces SIPlocalservice
Two different companies with two different platforms. Primus U.S. uses a SIP based service. Primus Canada's new service, Talk Broadband, is pure MGCP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Liu Sent: Wednesday, January 21, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] OT: Canada's Primus introduces SIPlocalservice I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David [EMAIL PROTECTED] 1/21/2004 6:39:34 AM I'm not sure Primus uses SIP. I think it's MGCP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Liu Sent: Tuesday, January 20, 2004 9:16 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: Canada's Primus introduces SIP localservice Hey Colin, Do let me know if Primus' SIP service can work with Asterisk. We tried setting it up like how you would for iconnecthere However, we even failed to register in the first place! (Of course password and username are correct). Anyone else on the list successfully used Primus' SIP with Asterisk? David [EMAIL PROTECTED] 1/20/2004 12:25:50 PM Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign up, they ship you this Dlink puppy for free: ftp://ftp10.dlink.com/pdfs/products/DVG-1120/DVG-1120_ds.pdf It has 2 FXS ports + ethernet + POTS backup port My order's in already, I'll be pleased to tell Telus where to put their value pricing once I get it installed. If anyone in Canada wants to know my experiences with it, email me off-list next month. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
Ken Godee wrote: This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? Couple of reasons.. 1. It is a stable, known quantity that uses solid components and closely mirrors the environment that a lot of people develop Asterisk on. It isn't going to drastically change, so those wishing to deploy it in production may look to RedHat 7.3 as a stable platform for that purpose. I agree, keep up the good work. I personally don't see any reason to upgrade atleast until the 2.6.x kernel is well underway. Maybe that's just me, hell I'm still running a 4.11 Novell server and a SCO Open server that hasn't been touched since y2k upgrades. I am guessing your systems are not connected to the internet then.. :) The problem with running servers based on RH 6.x, 7.x and 8 is that RH is not providing errata (security specifically) updates any more.. If you servers are not connected to the internet then, sure stay with the versions that are working for you, but if you have you server live on the internet for ant reason then this is a big issue.. I realise that many vulnerabilities require local access but I am still not going to take the chance.. I want my servers as safe as possible, and if that means running the latest versions of whatever then thats what I am going to do.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
What hardware is on the other side of the call(initiating)? Is it set up to send as voice to avoid data call costs? I remember at one point that was a neat trick to keep the telco from charging their data premium, the data would be passed over the circuit as voice. I think it had to do with the PSCs trying to keep ISDN in the US from being metered by the minute, but giving ground on data calls. On Thu, 2004-01-22 at 09:28, Thomas Haeger wrote: Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?
On Wed, 21 Jan 2004, Bill Hamel waxed: Hi, Looking around I can't seem to find a way to show the number of agents currently logged into a queue and if possible who they are. Is there a way to do this ? Thanks -b I attached a patch I've been using to show the # of agents (members) and callers on a per queue basis. It adds a new manager command, AgentQueues. It returns on the manager interface the following for each queue: Queue: queuename Agents: # Callers: # There's another manager command, QueueStatus, that might be what your are looking for. There's also Queues but that is a PITA to parse. Fine if you just want to display it in a text widget or something. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queue with no agents - Congestion or voicebox instead of MOH?
Do not define any members in the queues.conf. Instead have them login to the queue using the AddQueueMember application. If there is no one logged into the queue when a call comes in, it will go to the priority in the context. Hope this helps. B. J. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Baumann Sent: Thursday, January 22, 2004 5:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Queue with no agents - Congestion or voicebox instead of MOH? Hi all, I have successfully set up a call queue with agents and agentCallbackLogin. Works fine, but if no agent is logged in incoming callers get music-on-hold forever (or until some timeout). Is it possible to play congestion tone without answering the call (and thus causing costs to PSTN callers) or send them to unvailable-mailbox directly to leave a message if no agents are logged into the queue? Thanks and best regards, Jan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
The incoming call request Unrestricted and 64K, and this looks like ok, but in the SETUP_ACK the called number parameters shows: Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN equipment. In the most of cases, Information transfer rate = to '64 kbit/s', and Info transfer capability = 'real bw required'. Are you sure that the equipment attached to * can be used in 64K? Regards, Gus - Original Message - From: Thomas Haeger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 12:28 PM Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3328334778' ] Called Number (len=11) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '63494441' ] -- Making new call for cr 5635 -- Processing Q.931 Call Setup -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 108 (Calling Party Number) -- Processing IE 112 (Called Party Number) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 38403/0x9603) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Accepting call from '3328334778' to '63494441' on channel 30, span 2 -- Executing GotoIf(Zap/61-1, 0?50:100) in new stack -- Goto (pri2,63494441,100) -- Executing Dial(Zap/61-1, Zap/g2/033283077733SPEECH) in new stack -- Making new call for cr 39439 Protocol Discriminator: Q.931 (8) len=50 Call Ref: len= 2 (reference 6671/0x1A0F) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '3328334778' ] Called Number (len=21) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '033283077733SPEECH' ] -- Called g2/033283077733SPEECH Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 39439/0x9A0F) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (Channel Identification) beroasterisk*CLI Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (Cause) -- Channel 30, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap sending, peerstate Overlap Receiving Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6671/0x1A0F) (Originator)
Re: [Asterisk-Users] need help configuring IAX to make outbound calls through a remote server
On Wed, 21 Jan 2004, Paul Mahler waxed: I am trying to make outbound calls from my Asterisk client through a remote Asterisk server with IAX. In iax.conf on both sides [dar] context=trusted secret=xx type=friend host=192.168.1.1 I'm not going to try and fix all of this, but if you've got the same hostname on both hosts, one of them doesn't know about the other. You need to set the host differently on each of the hosts. Ie, on 192.168.1.1, you need to set host=192.168.1.2 and on 192.168.1.2, you need to set host=192.168.1.1. in extensions.conf at the client making the call Exten=_1NXXNXX,1,Dial(IAX2/dar:[EMAIL PROTECTED]/) What goes in extensions.conf at the remote server? What is needed for the remote server to accept the call from my client, figure out the dialed number and then dial it outbound on some line? You'll need to have a trusted context in each, for starters. But there's a lot more dialplan work you'll need to do, depending on where your outbound lines are, what numbers they can dial without incurring toll charges, etc. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
On Thursday 22 January 2004 03:08, WipeOut wrote: My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Why not subscribe to Progeny? They offer continuing support for RedHat 7.3 installations. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Standalone FXO device
Kannaiyan Natesan wrote: SJ, I'm also dealing with Andrew, they were good at telling you stories but nothing professional with the product. I registered with fwd and started dialling 14551 my fauvorite where i get clear voice. It gave me with completely noisy sound, I tried to reduce and increase the gain, but nothing works. I primarily want to share my DSL connected PSTN line to other members, so other members can use my PSTN minutes for free. But when I connect with the Clipcomm device, my DSL gets down and it gets up only when I switch remove the line from the device. I dunno what kind of problem it is. I left with that, Next I want to try to other networks by connecting ATA 186 FXS port to it. It works sometimes and just holds the line without hanging it up. I need to switch it off to get the line hooked on. Something very strange. The overall performance of the devices just sucks, I use the model CG-101E. If you need the device, I can ship you the one which I have got with me, since I'm no more interested in having it. I don't mind paying higher, but I'm looking for a quality device. If you can suggest anything, please share it to all in the list. Kannaiyan Hmm.. Very odd and strange problems. Have you pointed these problems to clipcomm people? Well, I was/am looking for a device with PSTN FXO backup. www.dlink.com does one like that, but is way too expensive. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gsm + snom phones
Matteo Brancaleoni wrote: Hi. About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something but you also tested gsm ? It works for me with gsm :-) 6 snom 200 and one snom 105 with gsm over german t-dsl (128 Kbit) sound is really good :-) Greets,Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards Detlef Wengorz [EMAIL PROTECTED] Detlef Wengorz [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium X100P for $43
This sounds like a good business. Get a no-name X100P-alike that retails everywhere else for about $15 and then put it on e-bay for $43 to fools that don't know any better. I love America. John - Original Message - From: Alfred R. Nurnberger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 9:59 AM Subject: RE: [Asterisk-Users] Re: Digium X100P for $43 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 3:48 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Digium X100P for $43 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Wednesday, January 21, 2004 11:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Digium X100P for $43 for the record, mine has the same fcc id number as the Digiums. Is this typical for copied hardware, or is there something a little fishy going on here? -- -- - I looked at the site of www.digitnetworks.com today. The thing I noticed was that nowhere on the site they listed the real (registered) company name or mention their address. One line has a (801) phone number listed otherwise only email addresses. Whois revealed the following: Registrant: Domains by Proxy, Inc. 15111 N Hayden Rd., Suite 160 PMB353 Scottsdale, Arizona 85260 United States Again no name, no address, sucessful way of hiding their identity. I stick with Digium, I know who they are, where they are and what I am getting, and I am supporting the developmnet of *. Alfred. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integrating * with a legacy Nec NEAX 1400
Title: Message I have searched the lists and the wiki for some info regarding Integrating * with a legacy Nec NEAX 1400. I am trying to build a test PBX and eventually integrate it with the current PBX here atmy office. Do further testing and if all goes well replace the neax with *. I have a T1 line coming in from the Phone company, but as far as I can tell the most expensive part of this is going to be replacing the telephones. If * could talk to my existing phones (Dterm Series II by NEC) I would be able to break the cost down into modules eventually replacing the phones for VOIP models. I have about 70 multiline terminals,20 single line, and 3 conference rooms. What is the best way to start off for this project? Are there any companies that have done something similar to this that I could talk to about their experiences with * ? Is there anyone else in the Wisconsin area that has an * system running that I could talk to about their experiences with * ? I am very interested in hearing any comments and suggestions you may have about this. Thank you.
AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Hi, we tried following scenario: DTAG (S0) at our office Datacall with AVMFritz (PSTN) --- Colo TelesSwitch -- CoLo Asterisk (--- PSTN) I think, no i know that the Teles Switch can route 64k data calls here is the Teles Trace: #08SETUP--| 15:29:40,378 02 01 78 AE | 08 02 03 90 05| Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 9B| Calling PN 6C 0C 21 83 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 09 C1 36 33 34 39 34 | 34 34 31 | |--RR #08 | 15:29:40,388 02 01 01 7A |--SETUP ACKNOWLEDGE #08 | 15:29:40,398 00 01 AE 7A |08 02 83 90 0D | Channel Id 18 03 A9 83 9B |--SETUP #12 | 15:29:40,408 00 01 2A D4 |08 02 16 60 05 | Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 88 | Calling PN 6C 0C 21 80 33 33 32 38 |33 33 34 37 37 38 | Called PN70 09 81 36 33 34 39 34 |34 34 31 #08 RR--| 15:29:40,408 00 01 01 B0 | #12 RR--| 15:29:40,418 00 01 01 2C | #12SETUP ACKNOWLEDGE--| 15:29:40,418 02 01 D4 2C | 08 02 96 60 0D| Channel Id 18 03 A9 83 88| Progress Ind 1E 02 81 82 | |--RR #12 | 15:29:40,418 02 01 01 D6 #12SETUP--| 15:29:40,428 02 01 D6 2C | 08 02 1A 21 05| Bearer Caps 04 03 88 90 A3| Channel Id 18 03 A1 83 81| Calling PN 6C 0C 41 81 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 0D C1 30 33 33 32 38 | 33 30 37 37 37 33 33 | |--RELEASE COMPLETE #12 | 15:29:40,428 00 01 2C D8 |08 02 9A 21 5A |08 02 80 D8 |[Incompatible destinat |ion] #12 RR--| -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN - Gus Gesendet: Donnerstag, 22. Januar 2004 17:24 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI The incoming call request Unrestricted and 64K, and this looks like ok, but in the SETUP_ACK the called number parameters shows: Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN equipment. In the most of cases, Information transfer rate = to '64 kbit/s', and Info transfer capability = 'real bw required'. Are you sure that the equipment attached to * can be used in 64K? Regards, Gus - Original Message - From: Thomas Haeger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 12:28 PM Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3328334778' ] Called Number (len=11) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '63494441' ] -- Making
[Asterisk-Users] Using varables in MeetMe?
Hi,
Im trying to enable users to enter a conference number and then do a
calculation on this and then send them to the conference. Lokk at this
example:
exten = s,1,Read(room)
exten = s,2,SetVar,${room}=[${room} + 2000];
exten = s,3,Meetme($room|pqsd)
What happens is that the conference gets setup and everything, but the
conference number is $room. Not really what i expected...
Is this a bug of a feature?
/Chris
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Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
Tilghman Lesher wrote: On Thursday 22 January 2004 03:08, WipeOut wrote: My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Why not subscribe to Progeny? They offer continuing support for RedHat 7.3 installations. -Tilghman Didn't know it existed.. Looks very interesting.. Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
I am researching the use of White Box Enterprise Linux. Someone else in a similar position with a bunch of 7.x boxes created it. He took all the SRPM files for REL v3 and removed all the Red Hat logos and trademarks. It is the same software as Enterprise but you can freely copy it. They also modded the update scripts to work with more generic update sources. The cool thing is the system is completely compatible with the REL source errata which Red Hat has promised to continue updating for at least five years. They have also setup a small system of mirrors to host the update files. It looks very promising. I am trying this and Debian to see which will be easier to keep updates for. Info and ISO file available at http://www.beau.org/~jmorris/linux/whitebox/index.html Someone else has a similar project going but it didn't seem to be as far along. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Tilghman Lesher [EMAIL PROTECTED]: On Thursday 22 January 2004 03:08, WipeOut wrote: My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Why not subscribe to Progeny? They offer continuing support for RedHat 7.3 installations. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 0.7.1 - mysql
On Thursday 22 January 2004 08:01, Andrew Thompson wrote: - Original Message - From: Dawid Mielnik [EMAIL PROTECTED] To: [EMAIL PROTECTED] Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? MySQL support was moved out to addons. You don't have to use ODBC to point to MySQL, but I would say it's probably a good idea. There is no guarantee that anyone will continue to update the mysql code now that the license on mysql has changed. There's no guarantee that Asterisk will be maintained either. There are lots of people who are interested enough, though, to make sure that it continues to be maintained. As far as cdr_addon_mysql.c support, I'm committed to maintaining it for the forseeable future. In fact, I've just uploaded a patch to the bugtracker to add a CLI command to this module: http://bugs.digium.com/bug_view_page.php?bug_id=902 -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing List Lag
- Original Message - From: Steve Foy [EMAIL PROTECTED] To: Asterisk-Users [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 3:18 AM Subject: Re: [Asterisk-Users] Mailing List Lag On Thu, Jan 22, 2004 at 08:39:09AM +, Steve Foy wrote: I'd be willing to host the list, I guess it just depends on how many emails/day the Asterisk list goes through... Seems to be around 1200 emails per week, for the one week that I counted anyway... Last week I was complaining to John, at Digium, about the list lag. Mark walked by his office so John voiced my complaint. Mark then said that the mailing list does 9 million messages a day. Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Zaptel Modprobe driver failure
On Wednesday 21 January 2004 15:53, Mark Rizzo wrote: Hello, my first post to the list. I have started to install and play with Asterisk. I was following some basic instructions to 'jump-start' my system. I have a TDM400P with one port and a X100P. I am running the latest CVS versions (from today). Following the steps I found for basic jump-start I did the following: Modprobe zaptel Modprobe wcfxo Modprobe wcfxs Edited /etc/zaptel.conf and added the following lines: fxsks=1 fxoks=2 loadzone=us defaultzone=us I then ran ztcfg -vv and receive a good response. Asterisk started and running just fine! YES! I then rebooted my computer for other reasons. Now the following happens: Modprobe zaptel (works) Modprobe wcfxo ZT_CHANCONFIG failed on channel 2: No such device or address (6) /lib/modules/2.4.20-gentoo-r9/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-gentoo-r9/misc/wcfxo.o: insmod wcfxo failed Modprobe wcfxs (works) lsmod shows that both modules are loaded. Asterisk runs, though I have yet to try and use both boards. If I remove my changes to the zaptel.conf file, then run modprobe, then re-add my changes to zaptel.conf file I am fine. The problem is not that the modules aren't loading, but that when the first module is loaded, the entry in modules.conf attempts to configure the second channel, for which the driver is not yet loaded. This is the error that you see about the second channel. I have found three possible solutions: 1) Remove the post-install wcfxo line from /etc/modules.conf. This is the easiest solution, but it is not permanent, as everytime you recompile and reinstall the drivers, the line will be readded. 2) Create /etc/zaptel.conf.0 and /etc/zaptel.conf.1, the first config file with only the configuration for your wcfxo, and the second for all channels. Before each modprobe, symlink (or copy) the appropriate file into place, e.g. modprobe zaptel ln -sf /etc/zaptel.conf.0 /etc/zaptel.conf modprobe wcfxo ln -sf /etc/zaptel.conf.1 /etc/zaptel.conf modprobe wcfxs The benefit here is that you don't ever have to bother with this again until you add more hardware. The downside is that you need a script to do this. 3) Ignore the error. Upside: no action is needed. Downside: if you get in the habit of ignoring errors, sometime, you're going to miss something important. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and gnugk
This is quite possibly a daft question, but it is possible to run * and gnugk on the same system with gnugk acting as a proxy for netmeeting endpoints and feeding everything for PSTN and SIP out through *? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
On Thu, 22 Jan 2004, WipeOut wrote: Ken Godee wrote: This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? Couple of reasons.. 1. It is a stable, known quantity that uses solid components and closely mirrors the environment that a lot of people develop Asterisk on. It isn't going to drastically change, so those wishing to deploy it in production may look to RedHat 7.3 as a stable platform for that purpose. I agree, keep up the good work. I personally don't see any reason to upgrade atleast until the 2.6.x kernel is well underway. Maybe that's just me, hell I'm still running a 4.11 Novell server and a SCO Open server that hasn't been touched since y2k upgrades. I am guessing your systems are not connected to the internet then.. :) I am, but I am also intelligent enough to firewall systems and properly secure them, no matter what distribution I run. The problem with running servers based on RH 6.x, 7.x and 8 is that RH is not providing errata (security specifically) updates any more.. If you servers are not connected to the internet then, sure stay with the versions that are working for you, but if you have you server live on the internet for ant reason then this is a big issue.. No it isn't. If you follow best practices for your system, remove all unneccessary packages, and properly firewall it, you are at no greater or lesser risk than any other version of RedHat. Take a look at the following: http://www.nacs.net/~damin/linux-best-practices.pdf I realise that many vulnerabilities require local access but I am still not going to take the chance.. I want my servers as safe as possible, and if that means running the latest versions of whatever then thats what I am going to do.. :) Take a look at the number of exploits that are available for RH 8 and 9, and how quickly they are mounting up, and then rethink that statement. There are more exploits being targeted at these platforms, in a shorter period of time, than 7.3 and the earlier versions. Personal opinion here, but if you are relying on RedHat to be your security provider, you have no business administering a system connected to the Internet. Sure, they make it easier, but common sense and a solid understanding of the applications and code that your system is based on are a hell of a lot more comforting. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
On Thu, 22 Jan 2004, Tilghman Lesher wrote: On Thursday 22 January 2004 03:08, WipeOut wrote: My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Why not subscribe to Progeny? They offer continuing support for RedHat 7.3 installations. That was my point a bit earlier in the thread! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with flashing FXO callwaiting from FXS
Hey all...I can't seem to figure out *exactly* what needs to be done when I can't flash over to an incoming callwaiting call on FXO from an FXS card. Right now if I get a callwaiting call from the FXO and hit flash nothing happens. I've been over the archives and google and it appears that this is possible and I guess it can be done with *0 but I haven't found the config on how to glue all this together. If anybody can help out I'd really appriecate it! Thanks, Chris -- Since light travels faster than sound, isn't that why some people appear bright until you hear them speak? -Steven Wright http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Standalone FXO device
I connect with the Clipcomm device, my DSL gets down and it gets up only when I switch remove the line from the device. I dunno what kind of problem it is. With DSL one line shares voice and data. _Every_ device plugged into your phone line except your DSL modem needs an in-line filter. The filter prevents the non-modem from either changing the line impedance or putting high frequency noise on the line, either of which will kill the DSL signal. So if you plug an FXO device into the same line that also carries DSL and don't use a filter the FXO device could very well kill your DSL connection. But it all depends, some times you can skip the filters. I did but then I plugged in one more analog phone and broke DSL. I installed a few in-line filters and now the DSL works better. If you did place a filter between the clipcom device and the rj11 wall jack then something else is going on = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
I have been using Mandrake 9.2 and it has been totally stable and haven't had any problems with installations of asterisk. I stopped using RH9 because of the upcoming end of their support. J.C. - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 11:29 AM Subject: Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released On Thursday 22 January 2004 03:08, WipeOut wrote: My biggest problem is that RH has basically dropped me in the poo by killing off their free version and stopping support for all the free versions as well.. I have been looking at alternatives but so far nothing is going to fit the bill.. The other distro's are either way off the mark or too difficult to get running in the first place or to difficult to manage in a production enviroment.. also I can't affort $400 for RH Enterprise Linux for each of my test/demo/dev servers.. I guess there are many with the same problem.. :( Why not subscribe to Progeny? They offer continuing support for RedHat 7.3 installations. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Has somebody got it work at all ? I mean data calls (ISDN 64k) through asterisk. Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Donnerstag, 22. Januar 2004 19:07 An: [EMAIL PROTECTED] Betreff: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi, we tried following scenario: DTAG (S0) at our office Datacall with AVMFritz (PSTN) --- Colo TelesSwitch -- CoLo Asterisk (--- PSTN) I think, no i know that the Teles Switch can route 64k data calls here is the Teles Trace: #08SETUP--| 15:29:40,378 02 01 78 AE | 08 02 03 90 05| Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 9B| Calling PN 6C 0C 21 83 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 09 C1 36 33 34 39 34 | 34 34 31 | |--RR #08 | 15:29:40,388 02 01 01 7A |--SETUP ACKNOWLEDGE #08 | 15:29:40,398 00 01 AE 7A |08 02 83 90 0D | Channel Id 18 03 A9 83 9B |--SETUP #12 | 15:29:40,408 00 01 2A D4 |08 02 16 60 05 | Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 88 | Calling PN 6C 0C 21 80 33 33 32 38 |33 33 34 37 37 38 | Called PN70 09 81 36 33 34 39 34 |34 34 31 #08 RR--| 15:29:40,408 00 01 01 B0 | #12 RR--| 15:29:40,418 00 01 01 2C | #12SETUP ACKNOWLEDGE--| 15:29:40,418 02 01 D4 2C | 08 02 96 60 0D| Channel Id 18 03 A9 83 88| Progress Ind 1E 02 81 82 | |--RR #12 | 15:29:40,418 02 01 01 D6 #12SETUP--| 15:29:40,428 02 01 D6 2C | 08 02 1A 21 05| Bearer Caps 04 03 88 90 A3| Channel Id 18 03 A1 83 81| Calling PN 6C 0C 41 81 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 0D C1 30 33 33 32 38 | 33 30 37 37 37 33 33 | |--RELEASE COMPLETE #12 | 15:29:40,428 00 01 2C D8 |08 02 9A 21 5A |08 02 80 D8 |[Incompatible destinat |ion] #12 RR--| -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN - Gus Gesendet: Donnerstag, 22. Januar 2004 17:24 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI The incoming call request Unrestricted and 64K, and this looks like ok, but in the SETUP_ACK the called number parameters shows: Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN equipment. In the most of cases, Information transfer rate = to '64 kbit/s', and Info transfer capability = 'real bw required'. Are you sure that the equipment attached to * can be used in 64K? Regards, Gus - Original Message - From: Thomas Haeger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 12:28 PM Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Calling Number (len=14)
Re: [Asterisk-Users] Using varables in MeetMe?
On Thu, 2004-01-22 at 12:06, Christopher Arnold wrote:
Hi,
Im trying to enable users to enter a conference number and then do a
calculation on this and then send them to the conference. Lokk at this
example:
exten = s,1,Read(room)
exten = s,2,SetVar,${room}=[${room} + 2000];
exten = s,3,Meetme($room|pqsd)
What happens is that the conference gets setup and everything, but the
conference number is $room. Not really what i expected...
Is this a bug of a feature?
Reread the documentation on variables again.
exten = s,1,read(room)
exten = s,2,SetVar(room=[${room} + 2000]
exten = s,3,MeetMe(${room}|pqsd)
--
Steven Critchfield [EMAIL PROTECTED]
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