[asterisk-users] Calling Party Category Field
For the past month I've been having trouble dialing certain numbers. We have Asterisk 1.4.14, Zaptel 1.4.7, Libpri 1.4.2 on a CentOS 5 server. We are using PRI on a TE110P card /etcwith a provider called Alestra in Monterrey, Mexico. There are some numbers that whenever we dial them we always get a busy tone. These numbers do not all belong to the same provider, but they all do not belong to Alestra. In the CLI Asterisk says that the number is not available, like the number does not exist. After some testing the provider has told us that the problem is on our side and gave this explanation: Whenever we dial one of those numbers Asterisk is sending the following: 52 Calling Party Category Field 0x00 When it should be sending: 52 Calling Party Category Field 0x0a As they explain it our server is sending the wrong signal and that is causing the other side to drop the call. Where can I check on this? Is it possible to change this behavior? Here is the relevant part of the config: /etc/zaptel.conf # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,0,0,ccs,hdb3 #,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone = mx defaultzone=mx /etc/asterisk/zapata.conf language=es usecallerid=yes callwaiting=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no echotraining=yes rxgain=0.0 txgain=0.0 immediate=no context=e1-incoming accountcode=Alestra group=1 switchtype=euroisdn callerid=asreceived signalling=pri_cpe pridialplan=unknown faxdetect=both channel=1-15,17-31 Any ideas on how to solve this problem? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
At 03:54 PM 12/15/2007, you wrote: I'm curious to hear how you would have approached the problem of retrieving multiple columns out of a database and setting each column to its own variable. That is precisely what ARRAY() is designed to accomplish, and it CANNOT be done by letting Set have multiple key/value pairs. Whatever, but don't call it array(). I was so excited when I saw there was a new function called array because some part of my dial plan would be so much cleaner with arrays. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Am Samstag, den 15.12.2007, 16:55 -0800 schrieb Philip Prindeville: I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling number. This topic has been covered in length. In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stanaphone issues. can someone verify my config?
Sorry, being really busy recently and only now have the time to dedicate to this (finished uni for the summer break) The asterisk is running on the machine that does the nat for the network here at home, it is set as the dmz on the adsl router so all ports should be coming into it. I have done a sip debug and copied it (and sanitized it) and put it here - well up till all the retrys start to appear. ; richards stanaphone incoming ;register = 0892: (MY PASSWORD)@sip.stanaphone.com/0892 register = 0892: (MY PASSWORD)@sip.stanaphone.com/101 (tried it both ways, having the stanaphone number as extension makes no difference) 101 just goto's a thing that answers, plays a voice and thenputs it on hold which work on all other sip providers. [stanaphone-richard] type=friend username=0892 secret=(MY PASSWORD) host=sip.stanaphone.com allow=all ;allow=g729 ;allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Monday, September 24, 2007 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stanaphone issues. can someone verify my config? any firewall in between? On 9/18/07, Richard [EMAIL PROTECTED] wrote: Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0 http://192.168.0.0/255.255.0.0 ; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [ [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [ [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error) The lack of RTP everywhere makes it look to be a nat issue, but I have done everything I can think of to have that work, and the config is the same other then host, username and password on italk which is working fine. I have googled for the Maximum retries exceeded on transmission - I could only see some stuff related to
Re: [asterisk-users] stanaphone issues. can someone verify my config?
And I forgot the pastebin link - DOH - http://pastebin.com/m782bcee4 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Sent: Monday, December 17, 2007 12:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] stanaphone issues. can someone verify my config? Sorry, being really busy recently and only now have the time to dedicate to this (finished uni for the summer break) The asterisk is running on the machine that does the nat for the network here at home, it is set as the dmz on the adsl router so all ports should be coming into it. I have done a sip debug and copied it (and sanitized it) and put it here - well up till all the retrys start to appear. ; richards stanaphone incoming ;register = 0892: (MY PASSWORD)@sip.stanaphone.com/0892 register = 0892: (MY PASSWORD)@sip.stanaphone.com/101 (tried it both ways, having the stanaphone number as extension makes no difference) 101 just goto's a thing that answers, plays a voice and thenputs it on hold which work on all other sip providers. [stanaphone-richard] type=friend username=0892 secret=(MY PASSWORD) host=sip.stanaphone.com allow=all ;allow=g729 ;allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Monday, September 24, 2007 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stanaphone issues. can someone verify my config? any firewall in between? On 9/18/07, Richard [EMAIL PROTECTED] wrote: Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0 http://192.168.0.0/255.255.0.0 ; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [ [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [ [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error)
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
We build and maintain 7 Asterisk boxes for our customers, I have recently moved 3 to 1.4. I also use iaxmodem and on the last one 1.4.14 I was getting iax thread errors - which was reported as a bug in much earlier versions but apparently fixed. When 1.4.15 came out (two days later) it solved this problem, for me at least. I didn't dig any further but it did moderate my confidence somewhat. We run everything on ubuntu server 6.06 LTS and also use freepbx as the interface with some minor customisations. It works very well and we are now shifting some others to 1.4 but the issue is if anything goes wrong its too costly to fix, as part of maintenance we keep them uptodate. The main blocker for 1.4 was freepbx but now it supports 1.4 and seems to manage the transition really well. However being a small self employed group of two the main reason to stick with what works is the risk of cost. We don't generally do major upgrades without charging but there isn't any seriously missing functionality yet, and the effort involved to be sure it will be hassle free is significant. The clients have to see value in the upgrade. We also work with people still on version 1.0, because the risk of change to a working system is too high This seems to be the same issue already mentioned but my take on it is most people can't cope with any risk on production machines unless there is some significant gain. Its been a year now, generally I would think that means its probably starting to become stable but a year isn't very long really. Give it another year and the new installs will mostly be 1.4 and the migration process will be a lot more trusted. I don't think a year is really long enough to expect much more than where you are at. The debian stable, unstable, and testing model would be useful here, debian stable is so reliable it just rocks, if there was a version like that it would be fantastic (of course you trade access to the latest features for it) . We find ubuntu server a great balance between debian stability and getting the latest options. Is there a performance analysis of 1.2 vs 1.4 around or a clear business analysis of the distinctions in value for each? Cheers Duncan Lyle Giese wrote: Olle E Johansson wrote: All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. We might want to be more informative about those changes. We need to make a clear list of things you need to start changing as a user of 1.4 to prepare for lost functionality in 1.6. This information already exist, but should maybe be a bit more public. In some cases we do have to change in a dramatic way and can't preserve the old functionality to solve a bug in the software. This requires thorough discussion in the developer group and is something we really want to avoid at all costs. If this happens, it's clearly documented in the software. Thank you for your feedback, it's important to us. /O Along that this same line, I ran 1.0.something for a long time and it was working just fine for my SOHO. I had a channel bank to interface pots lines from the local Telco and feed the analog phones in the house. Over time, I replaced most of those analog phones with SIP phones. An unfortunate incident caused us to lose that server and several sip phones. When I recovered enough to rebuild *, I tried 1.4 and it would not compile completely and zaptel did not load properly. I download 1.2 and it worked with the same configs as 1.0, but the quality was poor. That was due to hardware issues. I purchased a new motherboard and rebuilt using a newer Asterisk 1.4 with the then current libpri and zaptel and the call quality came back. But I had a hard time with syntax changes. Basically I was jumping from 1.0.x to 1.4.x in one leap. My biggest gripe is that everything loaded and seemed to work. A day later we found this did not work and discovered a syntax change. A day later something else did not work, an other syntax change. Why isn't there some pre-processor to check the syntax of the config files? Would have saved me a whole bunch of time I didn't have to spare and still don't. Lyle As it is syntax problems or changes are not noticed or logged until Asterisk tries to execute them. If there is a chunk of code that is only hit once a week??? It almost
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Ira [EMAIL PROTECTED] writes: Well, I'd be happy if they came up with an elegant language with functions, parameters and proper variable scoping while getting rid of line numbers and all the rest of the baggage that shouldn't have been there in the first place. AEL is an attempt to solve some of that, but as it's just a precompiler to the underlying language it has limitations that shouldn't be there. I could not agree more strongly. The big question is what such a language should look like. The SIP Express Router language is not the solution either, it is way too low level and tied to SIP. Then there is Freeswitch, which seems a bit better -- unfortunately the XML makes it hard to read. Perhaps the best idea would be to use an existing programming language. I just have a hard time imagining how it could be made easy to read. /Benny ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Olle E Johansson [EMAIL PROTECTED] writes: asterisk -c starts Asterisk in the foreground and outputs all messages to the console, things that you may not catch otherwise when you start Asterisk in the background. You can do that, but not while Asterisk is running. So it isn't really an option for production environments, where you need it the most. /Benny ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote: If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. What we had attempted to do in deprecation modes was to print the warning ONCE for each deprecated operation, per Asterisk startup. I think that this was much too conservative. It is very easy to miss that deprecation warning, since it occurs so few times. Of course, the opposite side is that we don't want deprecation warnings to fill up your logs, so there's a balancing act here. But we could probably do with making the deprecation warnings a bit more prominent and print them multiple times (for example, every 10th usage). That should make it more clear that there's something to change. How about an asterisk-lint kind of program, That analyses all of the config files, and gves an error with file-name line number of the offendig config (perhaps with a suggestion of what it MIGHT be... Would be worthwhile for everybody writing configfiles manually, Not only for migrating purposes... Hans ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
We run everything on ubuntu server 6.06 LTS and also use freepbx as the interface with some minor customisations. It works very well and we are now shifting some others to 1.4 but the issue is if anything goes wrong its too costly to fix, as part of maintenance we keep them uptodate. The main blocker for 1.4 was freepbx but now it supports 1.4 and seems to manage the transition really well. I missed that fact. Yes, FreePBX support is an important piece of the puzzle. However being a small self employed group of two the main reason to stick with what works is the risk of cost. We don't generally do major upgrades without charging but there isn't any seriously missing functionality yet, and the effort involved to be sure it will be hassle free is significant. The clients have to see value in the upgrade. Absolutely. And we don't want to force upgrades, as an Open Source project there's no value in that. But at some point we want to quit supporting these old installations from the project side and move on. We also work with people still on version 1.0, because the risk of change to a working system is too high ...and if it works, why change? This seems to be the same issue already mentioned but my take on it is most people can't cope with any risk on production machines unless there is some significant gain. Its been a year now, generally I would think that means its probably starting to become stable but a year isn't very long really. Give it another year and the new installs will mostly be 1.4 and the migration process will be a lot more trusted. I don't think a year is really long enough to expect much more than where you are at. Guess we're learning that for a PBX, we have to look into a longer upgrade path, but a quicker uptake on new installations. That's really what we look for. In the case that something else changes and the customer needs to upgrade, we want it to be as smooth as possible. Is there a performance analysis of 1.2 vs 1.4 around or a clear business analysis of the distinctions in value for each? I haven't seen that. Anyone else? Again, thanks for valuable feedback! I am learning a lot from this discussion. /O ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Anselm Martin Hoffmeister wrote: In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. Do you know of any GSM providers/contracts where faking for a valid reason is possible? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
The only reason I am not upgrading to 1.4 is because out-of-the-tar it just won't build on my Fedora Core 4 machine. http://bugs.digium.com/view.php?id=9643 Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E Sent: Saturday, December 15, 2007 05:57 To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Saturday 15 December 2007 12:14:29 David Boyd wrote: On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote: Of course, all of these deprecations should be covered in UPGRADE.txt, so please read that file every time you upgrade to a new version. It will contain everything that has changed in a possibly incompatible way. And if you find something that broke that wasn't in this file, please let us know, so we can revise that file. We may not have gotten everything, but we do try. So if I read you correctly, all of the pain of the upgrade is due to lack of effort on the participants part! I wouldn't say all of it, but it would be a lot easier if people paid attention to the deprecation notices and resolved them. The whole point of deprecating methods is to allow people a transitional period in which they stop using said method and move to its replacement. This seems a whole lot like the attitude of proprietary vendors when they don't want to support a feature that is outside the scope of what they want to maintain. I thought this was an open source project that would allow participants to have a voice in what is or isn't included in a new release. Even an non developing end user provides valuable benefit to the project in QA and bug information to improve the project as a whole. Most (With exceptions) projects have a bit more interest in what the user community wants or needs in a package. The attitude of this project seems to be If you want it code it yourself, however if it something that doesn't map to the ideas of what Digium wants then it will never make it into the official release. Digium is a company; it does not want anything. The developers of the project, of which Digium has sponsored a great many, most of whom were developers prior to being employed by Digium, get to make those types of calls. Do you see the distinction? One of the nice things about working for Digium is that I maintain my individual perspective as a developer; we do not engage in groupthink. I don't understand why so much community support is placed into the project considering that the typical end user is treated like a second class citizen. I can't think of a single software project where the typical end user is anything but. Every open source project is not a democracy; they are meritocracies. That is, the degree to which your opinion matters is the degree to which you are able to contribute. And this isn't just code writers, either. People who put forth the effort to document the code also get a kudos and karma, as do people who report bugs, suggest fixes, and give feedback on candidate patches. To a lesser extent, knowledgable users who help on the various forums and business leaders who sponsor developers to work on Asterisk also have a greater voice than the typical end user. And that's true for closed source, as well. When was the last time that an end user asked for and received a new feature from Microsoft? So Digium, (I address the company since Tilghman now works for you) do you have any plans to query the user community and determine what a typical end user of the product needs? With the knowledge and skill that exists in your organization it would seem trivial to put something in place to allow user feedback not only developer feedback for release direction. It is extremely insulting for you to try to address my employer, when we're discussing code practice. For one thing, the company (though legally a person) does not generally respond on these lists. And secondly, as I mentioned before, all developers maintain their individual perspective, so when I make points on here, I do so as an individual contributor. If you have an issue with the way that I have approached something, then please talk to me. Trying to go over my head is rude and unlikely to produce better results. As far as user feedback, there are multiple forums that exist that will influence individual developers, to a certain extent, which are the -dev list (please discuss code or policy, NOT user-level assistance; that's what this list is for), the #asterisk-dev channel on Freenode (same condition applies; use #asterisk for user-level questions), and the bugtracker (which is for reporting bugs, inconsistencies, and other things that relate to execution, not policy, which should be discussed on the mailing lists). Of course, if you want your voice heard more loudly, then contribute some of your efforts towards furthering the project. Complaints are always heard more critically when they come from somebody who has made the effort to give back in some way. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Mike wrote: The only reason I am not upgrading to 1.4 is because out-of-the-tar it just won't build on my Fedora Core 4 machine. http://bugs.digium.com/view.php?id=9643 Umm... forgive me for jumping in here, but that bug is for a (now unsupported) H.323 channel driver in asterisk-addons, with a very simple Makefile fix (for those users where the channel driver does work), and isn't actually part of Asterisk 1.4 at all. In fact, the in-tree H.323 channel driver in Asterisk 1.4 is vastly improved over the one in Asterisk 1.2 and most users are happy with it and aren't using chan_ooh323c any longer. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Sun, 2007-12-16 at 12:12 -0500, Mike wrote: The only reason I am not upgrading to 1.4 is because out-of-the-tar it just won't build on my Fedora Core 4 machine. http://bugs.digium.com/view.php?id=9643 Seen that one on and off. Don't know why this error keeps popping up. Would be nice if the responsible developer would check if chan_h323 installs after making changes Iirc the fix For FC7, F8 and CentOS 5 is: change libchan_h323.so.1.0.1 to libchan_h323.1.0.1 in the Makefile(s) in asterisk-ooh323c (remove the .so part) Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reputable company for SIP/IAX2 trunking
Steve, if you want quality and reliability, then you need to get as close as you can to the actual big guys operating the equipment, such as Level3, GlobalCrossing, XO, CommPartners. But they won't be interested in doing business with you for just 1 DID and couple thousand minutes a month. So find yourself a good first-hand reseller of those big guys who is interested in doing business with you. There are many out there. We have been getting 90% of our west-coast DIDs from CommPartners directly, and over the last 3 years, I don't recall a single indecent when they let us down service. The actual VoIP service is excellent; billing and paperwork can be messy at times. Luki On Dec 15, 2007 4:25 PM, Steve Finkelstein [EMAIL PROTECTED] wrote: Hi all, There's a myriad of options these days and I haven't been keeping up to date with what's respectable any longer. I essentially need a provider that will provide me with one DID to start and let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on my end and have full control over the dial plan. This way I can branch out my DID into extensions and have it dial individual peers according to an extension. Looking for some feedback on what provider is quality these days. I don't mind paying an extra dollar or two. Thanks, - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reputable company for SIP/IAX2 trunking
Thanks for the reply. :-) Yeah. I need a service that's going to allow multiple channels eventually. Perhaps 1000 simultaneous calls through one number. I'm just doing my research now and it looks like I'm going to have to start getting multiple ds3s for this type of call center setup. On 12/16/07, Luki [EMAIL PROTECTED] wrote: Steve, if you want quality and reliability, then you need to get as close as you can to the actual big guys operating the equipment, such as Level3, GlobalCrossing, XO, CommPartners. But they won't be interested in doing business with you for just 1 DID and couple thousand minutes a month. So find yourself a good first-hand reseller of those big guys who is interested in doing business with you. There are many out there. We have been getting 90% of our west-coast DIDs from CommPartners directly, and over the last 3 years, I don't recall a single indecent when they let us down service. The actual VoIP service is excellent; billing and paperwork can be messy at times. Luki On Dec 15, 2007 4:25 PM, Steve Finkelstein [EMAIL PROTECTED] wrote: Hi all, There's a myriad of options these days and I haven't been keeping up to date with what's respectable any longer. I essentially need a provider that will provide me with one DID to start and let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on my end and have full control over the dial plan. This way I can branch out my DID into extensions and have it dial individual peers according to an extension. Looking for some feedback on what provider is quality these days. I don't mind paying an extra dollar or two. Thanks, - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
I have not made the switch from 1.2 to 1.4 yet due to operating a ITSP Asterisk Cluster. I cannot upgrade any one machine without upgrading all. Basically I need to build a duplicate cluster with 1.4, debug it then roll traffic to it. This is a pretty gargantuan effort that I'm currently planning and will hopefully accomplish within Jan-Feb 08. I'm looking forward to the upgrade and having some of the new features 1.4 has like multi threading IAX2. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Sunday 16 December 2007 02:19:16 Ira wrote: At 03:54 PM 12/15/2007, you wrote: I'm curious to hear how you would have approached the problem of retrieving multiple columns out of a database and setting each column to its own variable. That is precisely what ARRAY() is designed to accomplish, and it CANNOT be done by letting Set have multiple key/value pairs. Whatever, but don't call it array(). I was so excited when I saw there was a new function called array because some part of my dial plan would be so much cleaner with arrays. I'm open to suggestions for another name. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Tilghman Lesher wrote: On Sunday 16 December 2007 02:19:16 Ira wrote: At 03:54 PM 12/15/2007, you wrote: I'm curious to hear how you would have approached the problem of retrieving multiple columns out of a database and setting each column to its own variable. That is precisely what ARRAY() is designed to accomplish, and it CANNOT be done by letting Set have multiple key/value pairs. Whatever, but don't call it array(). I was so excited when I saw there was a new function called array because some part of my dial plan would be so much cleaner with arrays. I'm open to suggestions for another name. VARS() ROW() ? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Thanks for your thoughtful response. Dave On Sun, 2007-12-16 at 10:43 -0600, Tilghman Lesher wrote: On Saturday 15 December 2007 12:14:29 David Boyd wrote: On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote: Of course, all of these deprecations should be covered in UPGRADE.txt, so please read that file every time you upgrade to a new version. It will contain everything that has changed in a possibly incompatible way. And if you find something that broke that wasn't in this file, please let us know, so we can revise that file. We may not have gotten everything, but we do try. So if I read you correctly, all of the pain of the upgrade is due to lack of effort on the participants part! I wouldn't say all of it, but it would be a lot easier if people paid attention to the deprecation notices and resolved them. The whole point of deprecating methods is to allow people a transitional period in which they stop using said method and move to its replacement. This seems a whole lot like the attitude of proprietary vendors when they don't want to support a feature that is outside the scope of what they want to maintain. I thought this was an open source project that would allow participants to have a voice in what is or isn't included in a new release. Even an non developing end user provides valuable benefit to the project in QA and bug information to improve the project as a whole. Most (With exceptions) projects have a bit more interest in what the user community wants or needs in a package. The attitude of this project seems to be If you want it code it yourself, however if it something that doesn't map to the ideas of what Digium wants then it will never make it into the official release. Digium is a company; it does not want anything. The developers of the project, of which Digium has sponsored a great many, most of whom were developers prior to being employed by Digium, get to make those types of calls. Do you see the distinction? One of the nice things about working for Digium is that I maintain my individual perspective as a developer; we do not engage in groupthink. I don't understand why so much community support is placed into the project considering that the typical end user is treated like a second class citizen. I can't think of a single software project where the typical end user is anything but. Every open source project is not a democracy; they are meritocracies. That is, the degree to which your opinion matters is the degree to which you are able to contribute. And this isn't just code writers, either. People who put forth the effort to document the code also get a kudos and karma, as do people who report bugs, suggest fixes, and give feedback on candidate patches. To a lesser extent, knowledgable users who help on the various forums and business leaders who sponsor developers to work on Asterisk also have a greater voice than the typical end user. And that's true for closed source, as well. When was the last time that an end user asked for and received a new feature from Microsoft? So Digium, (I address the company since Tilghman now works for you) do you have any plans to query the user community and determine what a typical end user of the product needs? With the knowledge and skill that exists in your organization it would seem trivial to put something in place to allow user feedback not only developer feedback for release direction. It is extremely insulting for you to try to address my employer, when we're discussing code practice. For one thing, the company (though legally a person) does not generally respond on these lists. And secondly, as I mentioned before, all developers maintain their individual perspective, so when I make points on here, I do so as an individual contributor. If you have an issue with the way that I have approached something, then please talk to me. Trying to go over my head is rude and unlikely to produce better results. As far as user feedback, there are multiple forums that exist that will influence individual developers, to a certain extent, which are the -dev list (please discuss code or policy, NOT user-level assistance; that's what this list is for), the #asterisk-dev channel on Freenode (same condition applies; use #asterisk for user-level questions), and the bugtracker (which is for reporting bugs, inconsistencies, and other things that relate to execution, not policy, which should be discussed on the mailing lists). Of course, if you want your voice heard more loudly, then contribute some of your efforts towards furthering the project. Complaints are always heard more critically when they come from somebody who has made the effort to give back in some way. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
At 11:12 AM 12/16/2007, you wrote: That is precisely what ARRAY() is designed to accomplish, and it CANNOT be done by letting Set have multiple key/value pairs. Whatever, but don't call it array(). I was so excited when I saw there was a new function called array because some part of my dial plan would be so much cleaner with arrays. I'm open to suggestions for another name. Weird, I can't find it or any description of the syntax except what's on voip-info. Core show function array claims array doesn't exist on my version of 1.4 and make menuselect doesn't offer any obvious way to get it installed. Without that it's a bit hard to suggest another name. If it's intended for use with database requests only, than a prefix of DB and maybe a name of dbgetrow or dbputrow. And if it's actually a front for some sort of a SQL request than maybe it should be part of a set of SQLx commands. Maybe I'm wrong but in my life, array has always referred to an indexable structure of some sort. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Ira wrote: At 11:12 AM 12/16/2007, you wrote: That is precisely what ARRAY() is designed to accomplish, and it CANNOT be done by letting Set have multiple key/value pairs. Whatever, but don't call it array(). I was so excited when I saw there was a new function called array because some part of my dial plan would be so much cleaner with arrays. I'm open to suggestions for another name. Weird, I can't find it or any description of the syntax except what's on voip-info. Core show function array claims array doesn't exist on my version of 1.4 core show function ARRAY Names of functions are all uppercase in Asterisk. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Why no port to Windos?
You nailed it Randy! When an Asterisk appliance and associated phones can compete with a Panasonic KXTG-4000 (or similar) on terms including price, ease of use reliabilitythat's when Asterisk for every grandma, aunt, uncle counsins (who never finished high school) will be viable for the broader home/residential market. Hmmm - I have to think that the two markets will stay apart for a bit longer. Asterisk does some things well, some things not so well. And the same could be said for the Panasonic. The fact that the phones and the system are still very separate entities makes the break - the older phone systems are tightly integrated with their phones. It's back to me dealing with a purchasing manager at one of my old workplaces, who wanted a small laptop with a big screen. PaulH ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Philipp Kempgen wrote: Anselm Martin Hoffmeister wrote: In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. Do you know of any GSM providers/contracts where faking for a valid reason is possible? Regards, Philipp Kempgen I can think of some... in rural Idaho, cell coverage is sparse. I might check my voice mail of my cell phone via a land line, and want to call back with a response originating from my cell number... Also the case if I have my cell set to forward-on-busy to my land line, or if I'm hosting an answering service, etc. -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Philip Prindeville wrote: Philipp Kempgen wrote: Do you know of any GSM providers/contracts where faking for a valid reason is possible? I can think of some... in rural Idaho, cell coverage is sparse. I might check my voice mail of my cell phone via a land line, and want to call back with a response originating from my cell number... Also the case if I have my cell set to forward-on-busy to my land line, or if I'm hosting an answering service, etc. What I'm looking for is this scenario: I call someone's cell phone number via my GSM gateway (to save money). But I'd like to set my landline number as the callerid (instead of one of the numbers of the GSM gateway or no callerid). Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox Phones Home
I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/43 ) that Trixbox has been phoning home with statistics about their installations, as a Trixbox user exposed in Trixbox Phones Home at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home . -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial, answered and then hangup
Hi all,
I will a TDM card with FXO modules on it. Below is the dial plan.
When someone can 9123456, CLI will show dialing to 123456 and
answered. After answered, the call hangup. I would like to know what
will cause the case to happen. Anyone can give me some advice to
solve it?
exten = _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT})
exten = _9X.,n,Hangup
zapata.conf
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
;context=cs
channel = 1-8
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Re: [asterisk-users] dial, answered and then hangup
rilawich,
can you post the CLI output so we can see what is going on?
from the exten, it is doing exactly what you tell it to do... dial then
hangup
daveC
Rilawich Ango wrote:
Hi all,
I will a TDM card with FXO modules on it. Below is the dial plan.
When someone can 9123456, CLI will show dialing to 123456 and
answered. After answered, the call hangup. I would like to know what
will cause the case to happen. Anyone can give me some advice to
solve it?
exten = _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT})
exten = _9X.,n,Hangup
zapata.conf
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
;context=cs
channel = 1-8
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Re: [asterisk-users] dial, answered and then hangup
Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Saturday 15 December 2007 15:48:01 Philipp Kempgen wrote: Tilghman Lesher wrote: If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. After thinking about it for a while this is not true. Well, it's true for the dialplan. Changing CALLERIDNUM to CALLERID(num) is easy. But i guess people use a lot of custom applications built around Asterisk 1.2. If any of the interfaces (AGI, AMI, CDRs, queue log, ...) change that might break the app. Fixing these apps might not be trivial and probably requires a lot of fine-tuning. While that's true, the incompatible changes to AMI, AGI, and other non-C interfaces should all be documented in UPGRADE.txt, which users have been asked to read. Internal C APIs are quite a different matter, of course. Sometimes the interface simply must change, even in minor upgrades, in order to fix bugs. That is unavoidable. The best way about this is that if you've written an app in C, you should genericize it to be useful to others and contribute it back upstream. Once it's in Asterisk, you no longer need to worry about keeping the interfaces in sync -- the developers will do that for you, as an integral part of the changeset. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP service providers/PSTN termination points
Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere http://www.iconnecthere.com Vonage http://www.vonage.com Teliax http://www.teliax.com I found something known as Inphonex http://www.inphonex.com. These had the cheapest rates and quite a good coverage too. Anyone with experience on this one? I am looking at a combination of decent prices and good quality. Any other suggestions or ideas welcome too. TiA - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox Phones Home
As I pointed out here last night, there is also a very serious security vulnerability associated with this. Example: An attacker could compromise the script that is used on the remote host, and set it to force clients that connect to run a command such as rm -rf /. There are about half a dozen ways I could see this being abused - in either a one off or an every installation scenario. Fonality has yet to acknowledge this aspect of the issue - and I fear that they never will. See: http://voipsa.org/pipermail/voipsec_voipsa.org/2007-December/002522.html P.S.: On behalf of Rob (of FreePBX fame), I'd like to also point out this this is something that was added to trixbox, and not FreePBX. Quoting Rob: when someone mistakenly says 'trixbox does...' they usually mean 'freepbx does...' as FreePBX is the GUI Trixbox uses to configure Asterisk. In this instance, that is not the case - it is only a trixbox issue. From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Date: Sun, 16 Dec 2007 20:53:53 -0500 Subject: [asterisk-users] Trixbox Phones Home I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/43 ) that Trixbox has been phoning home with statistics about their installations, as a Trixbox user exposed in Trixbox Phones Home at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home . -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The best games are on Xbox 360. Click here for a special offer on an Xbox 360 Console. http://www.xbox.com/en-US/hardware/wheretobuy/___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Why no port to Windos?
On Fri, 2007-12-14 at 11:33 +0100, Gergo Csibra wrote: Friday, December 14, 2007, 5:47:38 AM, Paul wrote: Umm - you could just buy a SPA-3000/3102/3666/etc. What is SPA-3666? The special red model. PaulH ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
