Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf
Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra configuration included in sip.conf becomes a problem, I'll move it to another text-based config file - not my preferred option (since I'd like to keep everything close-to-hand) but not a major problem since I'm likely to need a separate config file for global configuration options anyway. Paul Hales wrote: It should ignore the keywords, but you will get lots of errors in the CLI. My guess is that if you put it all in a DB (and use realtime) you can probably do whatever you want. PaulH Rob Hillis wrote: Hi guys, I'm about to embark on a small (undoubtedly to get much larger) project to write a set of scripts to handle provisioning of phones - Snom to begin with, possibly with others (most likely Polycom and Linksys) to follow later. Since I want this script to handle *all* aspects of phone provisioning (such as BLF buttons and so on) I need a place to store data. My preference is to keep all phone related configuration in the one place - such as sip.conf or users.conf. How would having additional keywords (most likely with a prefix of some type to reduce the likelihood of conflicts with real keywords) in Asterisk's .conf files affect Asterisk? I would expect that Asterisk should ignore unknown keywords, but I'd rather check on this with those in the know first. Any insights? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 interdigit timers bug?
2008/11/1 Rodolfo Alcazar Portillo [EMAIL PROTECTED]: Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? I found this only last week... The problem is not the short timer, it is the dialtone audio definition (top of the same page IIRC). If you look at the tone definition for Dialtone it is only a few seconds long. When it runs out, the call is disconnected. I have lengthened our dialtone pattern to 50 seconds, and the long timer to 60 seconds, which works here. Regards, Steve ___ -- 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] Hang up detection with TDM400P and Telewest/Virgin Media line
robb wrote: I have a TDM400 working quite well, Digium dialled in and recompiled chan_zap with some changes , to get BT Callerid working and I have set hangup on polarity in the zaptel.conf which seems to work well this is a BT home line, not business, if you have a business line you should get the DCT set to 800ms and the disconnect clear should work Would you be kind enough to share the changes you made to get callerID working please? Any chance of posting the relevant bits of your zap config also? My situation is that I have callerid working most of the time on a home BT line. Hangup is fairly reliably detected. TDM400P However, at a customers site on a bunch of business BT lines and the same model of TDM400P we see unreliable hangups (not frequent, but occasional times that lines are getting stuck off hook). Also callerId is working about 50-60% of the time and when it doesn't work (or genuinely that the callerId is witheld) there is a long pause for about 2-3 rings before Asterisk answers the zap line. It would be desirable to limit this pause because it makes it look like they are being slow to answer all the calls! Just wondering what changes you made? Also, anyone understand why DCT is different between home and business lines? Can the Zap code be changed to avoid needing something tweaking on the exchange? Thanks Ed W ___ -- 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] loading misdn.conf strange error regarding out of range
Hello all! I just noticed, that since installing the latest SVN branch (152803), I receive the following error, when loading/reloading the misdn.conf file misdn reload [...] [Nov 3 16:20:37] WARNING[5267]: misdn_config.c:938 _build_general_config: misdn.conf: misdn_init=/etc/misdn-init.conf (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. The line in /etc/asterisk/misdn.conf is: misdn_init=/etc/misdn-init.conf that's EXACTLY copied from the sample and that's EXACTLY where my misdn-init.conf is placed. I've checked on access rights, all OK. What can that be? Another thing, probably related: If I call my mailbox it's much too loud and I can't change this with rxgain or txgain. This wasn't before. Before I used asterisk 1.6.0-beta9. Any ideas on how to solve this? It would be very appreciated! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] say load new
Hello! On Mon, Nov 3, 2008 at 6:30 PM, [EMAIL PROTECTED] [EMAIL PROTECTED]wrote: I would like to use say.conf settings but every time i restart asterisk i have to load manualy say load new is there a way to do it automaticaly i use asterisk 1.4.19 There is option in say.conf to do it in 1.6.0. Also you can add in you dialplan #exec statement to execute special script on sturtup or configuration reload. Into script add command asterisk -rx say load new. -- Best regards, Goncharovsky Igor ___ -- 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] Call problems
Emmanuel Pascal Bruno wrote: I have tried that too with no results On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Emmanuel Pascal Bruno wrote: I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-( Disabling firewalls is almost certainly going to ensure the problem persists. You need to ensure that all SIP and RTP ports are port-forwarded from your firewall to your Asterisk box. ___ Maybe you use wireshark to diagnose what is going on in your network? Kind regards Eberhard ___ -- 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] help with debugging phone call
Jerry Geis wrote: I am running 1.4.22. I am doing a simple call into the dialplan and am getting a strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user 404 sip:[EMAIL PROTECTED];tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/404-18afe560, SIP/bt610tmm/1044) in new stack -- Called bt610tmm/1044 -- SIP/bt610tmm-b4046c70 answered SIP/404-18afe560 -- Packet2Packet bridging SIP/404-18afe560 and SIP/bt610tmm-b4046c70 == Spawn extension (smvoice-sip, 33, 1) exited non-zero on 'SIP/404-18afe560' Note that both call attempts are from the same phone 404. How can I find out why the first situation above is not showing me dialplan messages like case number 2 above and debug this situation? THanks, Jerry -- extensions.conf [smvoice-sip] ; case 2 above exten = 33,1,Dial(SIP/bt610tmm/1044) ; case 1 above exten = 1044,1,Dial(SIP/bt610tmm/1044) I enabled sip debug, say it was trying to take 10 instread of 1044. I dont have a 10 in my dialplan, I dont have a _XX in my dialplan... If I put a 10 in my dialplan it runs that as expected like playback(demo-congrats). How do I tell what/what its matching on the 10? Jerry ___ -- 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] Call quality issue across VPN- POTS vs SIP
Hi All, Got a strange (at least IMHO) issue that doesn't make much sense to me. Basic configuration is two sites with a site-to-site (aka router-to-router) VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks, and the only VoIP is internal - all of our outward telecom is on POTS or Centrex-enabled POTS lines. Site 1 has a Dell PowerEdge 1950 with Asterisk built from source and an AEX804E to connect to the outside world. Site 2 has an Asterisk Appliance with the 4 FXO / 4 FXS configuration, with the FXS ports currently unused. The PBXes at each site are configured to be essentially independent but with a unified dial plan so that calls can be placed or transferred across the VPN with a SIP trunk connecting the two PBXes, and canreinvite=no is set everywhere. The only other heavy consumer of bandwidth across the VPN is a real-time file replication suite that we use for file synchronization. While this is the ultimate issue, I don't understand the phenomena I'm seeing: If a user dials in to one of Site 2 FXO lines then dials across the VPN to a user at Site 1 while the file replication job is running audio quality (to the caller on the POTS line only) is abysmal-- audibly it sounds like about 50% of the packets are dropped (He__ Th__ __u __r, T__s is ___ln) On the other hand if a user at Site 2 picks up one of the Cisco phones [with the replication job still running] and dials across the VPN to a user at Site 1 audio quality is fantastic, ditto if a user at Site 1 calls to a user at Site 2. Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Replacing the border gear with equipment that's QOS aware and can handle prioritization is already on the list (and may be in the process of being ordered at this point) Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer ___ -- 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] help with debugging phone call
I am running 1.4.22. I am doing a simple call into the dialplan and am getting a strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user 404 sip:[EMAIL PROTECTED];tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/404-18afe560, SIP/bt610tmm/1044) in new stack -- Called bt610tmm/1044 -- SIP/bt610tmm-b4046c70 answered SIP/404-18afe560 -- Packet2Packet bridging SIP/404-18afe560 and SIP/bt610tmm-b4046c70 == Spawn extension (smvoice-sip, 33, 1) exited non-zero on 'SIP/404-18afe560' Note that both call attempts are from the same phone 404. How can I find out why the first situation above is not showing me dialplan messages like case number 2 above and debug this situation? THanks, Jerry -- extensions.conf [smvoice-sip] ; case 2 above exten = 33,1,Dial(SIP/bt610tmm/1044) ; case 1 above exten = 1044,1,Dial(SIP/bt610tmm/1044) ___ -- 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] say load new
Hello all, I would like to use say.conf settings but every time i restart asterisk i have to load manualy say load new is there a way to do it automaticaly i use asterisk 1.4.19 Thanks ___ -- 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] Looking for a web video phone?
Is there anything like that? Any experiences? Sincerely, Robert Augustyn www.linqone.com http://www.linqone.com/ LinqOneLogoSM1.jpg___ -- 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] Call terminates after 20 minutes
Go to sip.conf. Find the SIP Session-Timers section. Ensure that you have this option set: session-timers=refuse This might help. If not, try other variations of the session-timers value. The default session-timer is 10 minutes - exactly half of what you claim is your duration maximums, so it seems suspiciously like that might have something to do with it. Maybe not. In any case, fire up wireshark/tethereal and watch the SIP packets for a particular call to see what's happening - distrust everything other than what you see on the wire and then work backwards. An understanding of SIP packet flows will be helpful here, or the ladder view of SIP transactions that is built into wireshark's graphical interface will certainly help as well. JT On Nov 2, 2008, at 11:07 AM, Jim Boykin wrote: Any help. Thanks On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin [EMAIL PROTECTED] wrote: Marcin, can you elaborate. No timer has been set and call is not idle either. Thanks Jim On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk [EMAIL PROTECTED] wrote: Jim Boykin pisze: We are running Asterisk SVN. We are facing a strange and repetable problem. All outgoing call gets terminated in approx 20 minutes. Asterisk initiates BYE message to the remote end and call terminates. Sesion-timer set but not supported by sip-peers? --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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] Call quality issue across VPN- POTS vs SIP
On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- 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] giving a user asterisk CLI access: how bad could it get
On Nov 1, 2008, at 5:15 PM, Tilghman Lesher wrote: On Saturday 01 November 2008 18:52:41 Alexander Lopez wrote: No need to compile ! out of asterisk source Just put SHELL=/bin/false in your login script The ! command will not work... That's not completely true. The only thing that will prevent is the ability to get a shell prompt from the command line. The user could still type '!' commands and get whatever he wanted. However, there are more indirect ways to get anything a user desires: the CLI has the ability to create extensions, extensions which could execute the System application, pick up his phone, dial the extension, execute the command, and even cover his tracks by putting NoCDR in the extension path and removing the incriminating extension afterwards (again with the CLI). In 1.4, it's even easier: he can originate a call from the command line, perhaps even to a phone of a person he wanted to take the fall for his exploit. So you can see, removing the '!' command can be done, but it will lead to a very false sense of security. It will stop only the extremely casual user, one who was unlikely to have been very much a threat in the first place. -- Tilghman Alex - There is also an enhancement to Asterisk that is seeing some work which will allow CLI permissions applied to each command - Eliel Sardanons is the most active (only?) developer on this code. This will be undoubtedly some time before completion and inclusion into TRUNK, but perhaps you might be interested in helping with the debugging/development of that branch: http://svn.digium.com/view/asterisk/team/eliel/cli-permissions/ Example config file: http://svn.digium.com/view/asterisk/team/eliel/cli-permissions/configs/cli_permissions.conf.sample?revision=151904view=markup JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a web video phone?
On Nov 3, 2008, at 12:11 PM, Robert Augustyn wrote: Is there anything like that? Any experiences? X-Lite is a free download and has video capabilities. Fred Posner [EMAIL PROTECTED] Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a web video phone?
Thank you, How do I embed it into the web site though? robert _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fred Posner Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for a web video phone? On Nov 3, 2008, at 12:11 PM, Robert Augustyn wrote: Is there anything like that? Any experiences? X-Lite is a free download and has video capabilities. Fred Posner [EMAIL PROTECTED] Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI dial and echo recorder
Say any ideas how to do the following from the cli In order to test I would like to dial my phone from the Asterisk cli and then record my voice on asterisk and have it played back to me? Also how can a I specify a specific callerid? Thanks Shaun ___ -- 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] Call quality issue across VPN- POTS vs SIP
Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is done for you but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- 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] Blank Voicemail.Conf after Password Change
Rob, Thanks for your time and assistance. The directory is owned by asterisk and permissions seem fine there too: drwxrwxr-x 9 asterisk asterisk 3072 2008-11-03 13:12 . I never see a voicemail.conf.new created. I have done a locate on the whole server and don't see a blank one there either which might be causing any confusion. ps xaguwww confirms asterisk is running as UID asterisk: asterisk 24560 1.0 3.1 50648 32692 ?Ssl Oct02 506:35 /usr/sbin/asterisk -U asterisk I see that it's been up for a while, and I'm wondering if that coincides with when we started noticing this behavior. I'd have to restart asterisk to use strace, but restarting for all I know might help... Any other input before I do so? TIA, LN [EMAIL PROTECTED] wrote: Message: 15 Date: Sat, 1 Nov 2008 00:22:30 + From: Robert Lister [EMAIL PROTECTED] Subject: Re: [asterisk-users] Blank Voicemail.Conf after Password Change To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Oct 29, 2008 at 01:13:56PM -0400, Leah Newmark wrote: From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707 2008-10-29 12:14 /etc/asterisk/voicemail.conf I believe what it does it create a new file called voicemail.conf.new in the same directory and then copies it into place, so worth checking the permissions on the directory as well, that asterisk can write to it. Asterisk is running as asterisk: 24560 ?Ssl 409:34 /usr/sbin/asterisk -U asterisk I see your asterisk is running -U asterisk but this ps output is ambiguous. What does ps xaguwww show? if it really is running as UID asterisk, you should see something like: asterisk 8506 0.0 0.6 443672 12912 ? Ssl Oct02 31:46 /usr/sbin/asterisk -U asterisk -G asterisk Nothing generated from voicemail is showing up in the asterisk logs, nor does the console show any error after changing a password. Otherwise, it could be some sort of odd file locking issue where multiple things are trying to write to the same file at once? Or perhaps you have a blank voicemail.conf.new that it can't erase, sitting about somewhere? Maybe try running asterisk under strace to see what happens when you try to change a password. Rob ___ -- 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] busylevel question
I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringing. As the GXP280 is a single line phone it does not ring the second call. I would have expected the call to get a busy as I have busylevel set to one. I have tried setting busylevel to two as well with the same result. Can someone let me know what I should look at to see why I am not getting a busy instead of ringing? Here are the definitions in sip.conf for the GXP280 and zoiper: [dickenson] type=friend context=empl nat=yes host=dynamic secret=password callerid=Jim Dickenson 108 [EMAIL PROTECTED] [GXP280] type=friend context=empl host=dynamic secret=password callerid=GXP280 109 [EMAIL PROTECTED] busylevel=1 TIA -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Posting Delay
On Oct 31, 2008, at 11:13 AM, Douglas Garstang wrote: We have a situation where it's sometimes taking Asterisk 17-19 minutes to post CDR's, both over the AMI, and over the MySQL socket. It seems however that they are logged locally to /var/log/asterisk/ cdr-csv/Master.csv right after the call is terminated. Anyone got any idea what's causing this? It's a problem for us because we (badly IMHO) are using CDR's to maintain call state (if a user is in a call for example). Doug. You may consider including additional information for others so that they may assist in debugging. Version of code? CDR.conf basic configs? Without more data, the first thing that comes to mind is batch=no in cdr.conf JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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] Ztdummy and Asterisk
On Sat, Nov 01, 2008 at 11:03:09PM -0200, Aldo D. Sudak wrote: Loading zaptel hardware modules: ztdummy. Running ztcfg: done. There is no need to run ztcfg if you just want to use ztdummy. The call to ztcfg is probably just part of some standard init script. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- 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] Call quality issue across VPN- POTS vs SIP
Just as an interesting follow-up/additional information, if I place a call to Site 2 on a POTS line, someone at Site 2 answers the call (using one of the Cisco phones) and then transfers it to me across the VPN the call sounds fine. So I think Bob's question was on the right track with it being a CODEC issue, but I'm not sure how I need to deal with that for the ZAP channel type. Thanks again, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln King-Cliby Sent: Monday, November 03, 2008 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is done for you but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote: It's conceivable, but how would I verify this and how would I change it if that was the problem? There's a few things you can do here. 1) Check the sip.conf on both sides to see what is defined there for the trunk. Look for some disallow and allow statements. If they are there, that will tell Asterisk what codecs to use on that trunk. 2) You could also check the codec that is in use during a call by looking at the sip channel. From the asterisk CLI, start with show channel SIP/ and tab it out to complete the command showing the trunk between your two systems. I believe the codecs are listed here as NativeFormats and ReadFormat. You could check this under both of your scenarios to see if there is a different codec in use. 3) If you'd like to try and force the use of a compressed codec such as GSM between your two sites, you would just need to make sure that both sides had the following lines in the definition for the trunk in sip.conf and then do a 'reload chan_sip.so from the Asterisk CLI: disallow=all allow=gsm ___ -- 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] VoIP traffic shaping
Kristian Kielhofner wrote: On 11/1/08, OCG Technical Support [EMAIL PROTECTED] wrote: This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? It was based off Wondershaper originally, enhanced for VoIP traffic and gives the option to use HFSC or HTB. Not only do I use it myself for AstLinux and Star2Star, most of the reports I've (we've) had have been favorable. Try it out! Has anyone tried this on OpenWRT? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ztdummy and Asterisk
Hi again, Thanks for your answer, Stefan. In fact, ztcfg is automatically running from the startup script. Anyhow this seems harmless and not the cause of the problem. I have confirmed this with two other Asterisk servers, both having a zaptel card installed in them, but one running the same Debian Etch OS as the troublesome machine, and the other one running Fedora Core 6. In both cases I restricted the zaptel modules loaded to just ztdummy and modified zaptel.conf in order not to configure anything. Then I restarted zaptel and attempted to start Asterisk. In the first case I obtained the same error message as I reported before. In the case of the machine running Fedora, Asterisk started normally (and it's worth to say that ztcfg does also automatically run in it). So it is apparent that the problem involves Debian. Greetings. Aldo Sudak On Sat, Nov 01, 2008 at 11:03:09PM -0200, Aldo D. Sudak wrote: Loading zaptel hardware modules: ztdummy. Running ztcfg: done. There is no need to run ztcfg if you just want to use ztdummy. The call to ztcfg is probably just part of some standard init script. -- Stefan Tichy ( asterisk2 at pi4tel dot de )___ -- 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] asterisk src=dst
Hi all I saw in the CDR stocked in mysql as well as those in the csv file that some time, the src field is the same as the dst field which is the extension. When does it happens. Here, we have 4 dgits extensions and most of the time the dst field is the extension and the src field is the 10 digit customer phone number. Do you know when does this happens ?? Thanks Ruddy Gbaguidi http://www.astblog.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blank Voicemail.Conf after Password Change
On Monday 03 November 2008 12:19:31 Leah Newmark wrote: The directory is owned by asterisk and permissions seem fine there too: drwxrwxr-x 9 asterisk asterisk 3072 2008-11-03 13:12 . I never see a voicemail.conf.new created. I have done a locate on the whole server and don't see a blank one there either which might be causing any confusion. In 1.4, it no longer creates any such file. Instead, it creates a randomly named file, designed to be unique, starting with voicemail.conf. and ending with 6 random characters. ps xaguwww confirms asterisk is running as UID asterisk: asterisk 24560 1.0 3.1 50648 32692 ?Ssl Oct02 506:35 /usr/sbin/asterisk -U asterisk I see that it's been up for a while, and I'm wondering if that coincides with when we started noticing this behavior. I'd have to restart asterisk to use strace, but restarting for all I know might help... Any other input before I do so? Please try the following patch: http://bugs.digium.com/view.php?id=13831 -- 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] Question regarding keywords in sip.conf/users.conf
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Hillis wrote: Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra configuration included in sip.conf becomes a problem, I'll move it to another text-based config file - not my preferred option (since I'd like to keep everything close-to-hand) but not a major problem since I'm likely to need a separate config file for global configuration options anyway. You could still store them in sip.conf, just make each line a comment. e.g.: ;[myentry]keyword=value ;[myentry]keyword=value You could then search for ;[myentry] for your keywords and strip them off when you write out the real entries. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJD4ToCFu3bIiwtTARAiD2AJ4/5xKHZ3dSt1/J7GdrP0Bfqi5y6gCfQIBE RScZG7kDyLQOpHGTLeCHsIQ= =OVRk -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf
Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Hillis wrote: Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra configuration included in sip.conf becomes a problem, I'll move it to another text-based config file - not my preferred option (since I'd like to keep everything close-to-hand) but not a major problem since I'm likely to need a separate config file for global configuration options anyway. You could still store them in sip.conf, just make each line a comment. e.g.: ;[myentry]keyword=value ;[myentry]keyword=value You could then search for ;[myentry] for your keywords and strip them off when you write out the real entries. I did think of that, but the idea of using something that's actually a comment as configuration seems fraught with danger - not to mention it being an awful hack. ___ -- 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] Question regarding keywords in sip.conf/users.conf
Historically Asterisk's config file parser ignored unknown keywords. This is useful for exactly the things you are trying to do. I hope 1.6 did not remove this feature. Rob Hillis wrote: Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Hillis wrote: Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra configuration included in sip.conf becomes a problem, I'll move it to another text-based config file - not my preferred option (since I'd like to keep everything close-to-hand) but not a major problem since I'm likely to need a separate config file for global configuration options anyway. You could still store them in sip.conf, just make each line a comment. e.g.: ;[myentry]keyword=value ;[myentry]keyword=value You could then search for ;[myentry] for your keywords and strip them off when you write out the real entries. I did think of that, but the idea of using something that's actually a comment as configuration seems fraught with danger - not to mention it being an awful hack. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- 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
