Re: [asterisk-users] help...i cant do more...
On Fri, Apr 25, 2008 at 4:38 AM, Bruno Pereira [EMAIL PROTECTED] wrote: Thanks for the answers. I need to say that this command is executed from another machine, with the command ssh because in ocalhost is all ok, with sudo or with root. I will try that trace to see if it helps me, but the bg probem is start the service from another machine with ssh . Did anyone ever find a solution to this issue. I have the same problem when trying to start asterisk from another computer via SSH. It starts fine on the local box, but over SSH it just hangs forever. I am using root as the user, and issuing the command: ssh 10.0.0.10 '/etc/init.d/asterisk start'. Thanks! Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help Needed - Error when playing wav files in 1.4.11
I get the following error when trying to play wav files for my IVR menu. Does anyone know what this means or how to fix it? [Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt Thanks! David ___ --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] Voicemail gain option NOT working in 1.4.11?
Hi Everyone, I cannot seem to get the voicemail gain option g(#) work in Asterisk 1.4.11. I am using it like so... Voicemail([EMAIL PROTECTED],bg(10)) ; for busy announce and 10dB record gain This has absolutely NO affect on the resulting voicemail wav file. I have also tried using format=wav instead of wav49 in voicemail.conf to increase the volume as well. This also has no affect on the volume of the resulting wav file. Any help would be greatly appreciated, as asterisk is unusable without louder voicemail files. My boxes are SIP only. Centos 5, x86_64 Pentium-D 3.0 with 4GB of ram. Asterisk 1.4.11 was compiled from source with all default options. Thanks! David ___ --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] Voicemail gain option NOT working in 1.4.11?
I'm not sure how the gain option works as an argument to Voicemail(), but I know that the volgain option for e-mail attachments requires that you have sox installed in order to work properly. If you don't already have it installed, I would suggest installing sox and seeing if that helps. Mark Michelson Yes, volgain works fine, unfortunately it is only useful for emailed wav files. That's why I was trying to get the g(#) option to work. I suppose I could try to hack together a shell script using sox and call it from externnotify. Does anyone know if there are drawbacks to this approach? Any other thoughts or options? Thanks, David ___ --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] Configuration assistance needed.
A start would be to get the contact information and actually CONTACT the person about it. Come on now. Maybe I'm confused... Isn't that what Dovid did when he replied to Tim's post? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replicating SIP Registrations Across Asterisk Servers
On 4/2/07, John C. Wolosuk Jr. [EMAIL PROTECTED] wrote: Does any one know if there's an mechanism (internal to asterisk or otherwise) to replicate dynamic SIP device registrations across a pool of asterisk servers? I'm in the process of creating a asterisk cluster using a SIP hardware load balancer and so far this is one of the challenges I'm facing. One thought I'm currently investigating is to use openSER to intercept and replicate the incoming SIP REGISTER packets to all servers... The other thought in the back of my mind is to completely removing the task of handling registrations from asterisk and give it to SER directly or other registrar server. any other ideas? solutions? Using SER or OpenSER should work for this task. I have not found any other options to accomplish this. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtsavesysname not working in 1.4
On 3/2/07, Bruce Reeves [EMAIL PROTECTED] wrote: Try renaming you column in the peers table to regserver Thanks for the suggestion Bruce, unfortunately it did not help. Any other thoughts? Does the systemname in asterisk.conf and regserver in field mysql need to be an IP address, FQDN, hostname, or what is the proper format? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to disable MOH completely?
I need to disable MOH completely. We are using all SIP extensions and do not want Asterisk to invoke MOH when flash or hold is pressed on the phone. Anyone know how to configure this? Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtsavesysname not working in 1.4
Thanks again Bruce! That was indeed the problem. I added displaysystemname=yes and it started working. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable MOH completely?
On 3/5/07, C F [EMAIL PROTECTED] wrote: Just comment everything in your musiconhold.conf Funny thing is, I don't have a musiconhold.conf and res_musiconhold.so is not loaded, however when I press flash or hold on my phone (connected to an ATA), on the CLI I see Asterisk try to execute music on hold. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
On Tue, 06 Mar 2007 05:12:03, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and http://www.virtualphoneline.com/did for current available numbers. Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 Let me know how it goes for you, Rehan Ahmed Come on Rehan... Do you think we're really going to fall for that trick. We all know you represent virtualphoneline.com. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable MOH completely?
On 3/5/07, C F [EMAIL PROTECTED] wrote: Could be its trying but does it actualy play the music? It's not actually playing anything. I guess it just seems odd that Asterisk re-invites the media back to itself when a call is put on hold (when MOH is disabled), instead of simply disconnecting the media until the call is retrieved. I guess I was hoping for a config option that would simply turn MOH off to achieve this behavior. Does such a config option exist in 1.4? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rtsavesysname not working in 1.4
I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname = mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called sysname in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something I'm missing? Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak I'm not sure if it's related, but we are doing only SIP to SIP calling with Asterisk 1.4 and experience the same thing. The signaling shows up instantly, but it takes 5-7 seconds before ringback is heard. Watching the CLI it does look like it takes a long time for the channel to pick up an dial. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Packek2Packet Bridging vs. Native Bridging
I am just wondering if someone can explain the difference between Packek2Packet Bridging vs. Native Bridging in Asterisk. I'm basically tyring to make sure the media travels end-to-end and I've see both of these bridging types mentioned on the asterisk console. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax2 prun realtime peer only can't prune user
On 1/24/07, JR Richardson [EMAIL PROTECTED] wrote: Hi All, I'm running 1.2.9.1. I can prune sip realtime peers and users and iax realtime peers but no command to prune iax realtime users. Was this implemented in a later version? Thanks. JR From what I could dig up, it looks like you can do peers or all, but not users. The code in iax2.c has a function prune_users(); but I could not find anything pointing to a CLI command to prune iax users. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Open Source Hosted PBX
Does anyone know if there exists an Open Source Hosted PBX platform based on asterisk? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Realtime Voicemail Password Change Not Working
On 1/17/07, JR Richardson [EMAIL PROTECTED] wrote: I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? I updated to asterisk 1.2.14 and add-ons 1.2.5 with no luck. I still don't see any update statement in the mysql.log when I change a password. I built a vmbox in the voicemail.conf file and can change that password just fine. Any suggestions? JR, I'm just pulling things out of the air here, but if realtime voicemail works like realtime users/peers, loading everything into memory from MySQL, then there would need to be some type or prune command to force the re-read of the voicemail table, this is asuming you change the password via MySQL and not on the handset. Maybe something like DBput would work to update astdb as well. Again just throwing out ideas... It sounds like you are using the handset to update the password. Is this correct? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AbsoluteTimeout with canreinvite=yes
Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN in Asterisk 1.4
Browsing through the developers documentation and 1.4 source, I see references to STUN in the code and documentation. Does 1.4 have support for STUN, if so how is it configured? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rt db lookup
On 1/17/07, Tim Connolly [EMAIL PROTECTED] wrote: Okay. That doesn't help. What forces * to look at the DB rather than waiting on a registration ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Monday, January 15, 2007 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rt db lookup On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ regards David To my knowledge, the only two ways to do this is... 1.) To create a SIP or IAX trunk between each box that needs to communicate then add the login to your dialplan in extensions.conf to use those trunks when the call cannot be completed locally. 2.) To create a SIP or IAX trunk between each box that needs to communicate then configure DUNDi to handle the extension location. As far as registration and Realtime is concerned... have a look at the rtcachefriends option in sip.conf iax.conf. Hope this helps. - David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rt db lookup
On 1/17/07, David Thomas [EMAIL PROTECTED] wrote: On 1/17/07, Tim Connolly [EMAIL PROTECTED] wrote: Okay. That doesn't help. What forces * to look at the DB rather than waiting on a registration ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Monday, January 15, 2007 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rt db lookup On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ regards David To my knowledge, the only two ways to do this is... 1.) To create a SIP or IAX trunk between each box that needs to communicate then add the login to your dialplan in extensions.conf to use those trunks when the call cannot be completed locally. 2.) To create a SIP or IAX trunk between each box that needs to communicate then configure DUNDi to handle the extension location. As far as registration and Realtime is concerned... have a look at the rtcachefriends option in sip.conf iax.conf. Hope this helps. - David What I meant to say on # 1 is add the logic to your dialplan in extensions.conf - David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Absolute Timeout or Dial Limit option???
I need a method of limiting the duration of calls when RTP media does NOT travel through Asterisk. I know that the Dial() command limit option L requires Asterisk to carry the media, but what about Set(TIMEOUT(absolute)=XX)? Are there any other apps/options that might work for this? Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rt db lookup
On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ regards David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ANY ADVICE ON THIS????
On 1/15/07, Lars Knopf [EMAIL PROTECTED] wrote: Hello List, I am stuck with this problem for several days... anybody can give me a hint on this?? I know many of you dealt with problems similar to this, how did you address this?? Thanks in advance!!! -lars -- Forwarded message -- From: Lars Knopf [EMAIL PROTECTED] Date: Jan 11, 2007 1:12 PM Subject: realtime sipusers and rtcachefriends... big headache!! To: asterisk-users@lists.digium.com hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. Randomly, when trying to dial SIP/x (a customer's account), especially those behind NAT, I get in the console the error no route to Sometimes, too, they can't even register with asterisk. It seems to happen mostly when using realtime. I was digging into the bug tracking system, and I see two bugs that seems to be related, but I can't figure how can I fix it or what step I am supposed to do. The bugs are: http://bugs.digium.com/view.php?id=4687 http://bugs.digium.com/view.php?id=4832 So please, anything than can bring me some light on this... is very appreciated. I think you will need to prune the user/peer after changes. I believe the syntax is something like sip prune realtime user_or_peer where user_or_peer is the actual username. - David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
This is by far the most volotile list I have ever been on. I'm not sure that's exactly the reputation Digium/Asterisk is shooting for, but even so it does provide some much needed comedy relief. After seeing the G.729 pricing direct from SIPRO, I now take the shut-up and be thankful position. I think Digium has done us a great service by working out favorable pricing with SIPRO. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) As I understand it video will NOT work if you use an IAX trunk between * boxes, it must be SIP. Just food for thought in case you are planning on using video. David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail personalised greetings using DB/IMAP backend?
On 1/5/07, Ray Jackson [EMAIL PROTECTED] wrote: Hi Kevin, Thanks for your response. That answers a few questions I had. I am very happy to get involved in this area if I can help. Using IMAP and REALTIME I have a really nice VM solution with MWI, Webmail access etc. and it scales horizontally - I just add a new server into the mix when necessary. Until we get a generlized storage subsystem in place, I may look at a 'hack' to get the personalised greetings going... Do you think a shared NFS mount is risky for this? Should I do an rsync periodically perhaps to keep greetings on all servers up to date with each other? In the DUNDi * cluster we're designing phones can register with any of our asterisk boxes. Actually sometimes phones are registered to multiple boxes. I'm wondering if the new IMAP/MWI would have any problems with this type setup. Any experiences here? Regards, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 Random disconnects
On 12/28/06, Jason Adams [EMAIL PROTECTED] wrote: Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Yes, same thing here. This seems to be the only problem we have with 1.4. We are using only SIP connections. David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and MiniITX setups
On 12/29/06, Mark Greene [EMAIL PROTECTED] wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f - Mark I can get around 15-20 simultaneous SIP-2-SIP calls with no transcoding on a VIA EPIA-V 1.0 GHz, so it really depends on how many simultaneous calls you require and if there will be any transcoding involved. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Warnings
I get the following warning when starting Asterisk 1.4. Does anyone know what these mean, and/or how I can get rid of them? [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show cache' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show channels' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show firmware' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show netstats' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show peers' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show registry' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show stats' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show threads' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show users' already registered (or something close enough) Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is ZTDUMMY still required with Asterisk 1.4?
Is ztdummy still required with Asterisk 1.4 when no zaptel cards are available to use for timing? In all the beta releases I used to get a warning when Asterisk started up, saying that no timing device was found. The warning seems to have gone away with the full release of 1.4, which prompts the question... Is it still required? Does 1.4 do something different for timing? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is ZTDUMMY still required with Asterisk 1.4?
On 12/27/06, Carlos Alperin [EMAIL PROTECTED] wrote: Do you have a zap section on the CLI? Just do ? And check if you have that. I have zaptel working on two machines with wtc1xxp and ztdummy. The one with the card doesn't show zap section, the other one with ztdummy does. I thought that both should show the section on the CLI. No, I do not have zap listed when I type ? at the CLI. Does this indicate a problem? David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about sip account format
On 12/20/06, Rilawich Ango [EMAIL PROTECTED] wrote: I have 2 sip accounts with name 1234 and abcd respectively. Account abcd can make call to 1234 but not visa versa. When I change account abcd to 1abcd, both of them can make call to each others. In the case, the format of sip account should be start with number. I wonder whether we can use a sip account using only characters. Anyone can tell me how? Is it possible? Yes, as I recall you cen use alpha, numeric or both. I am guessing you do not have the proper character matching in whatever context the calls are going to in extensions.conf. I think something like this should work: exten = _[A-Za-z0-9].,1,Answer Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
Are you kidding? Lighten up people! Al made a friendly recommendation based on the comments regarding TrixBox. Go have a beer... take a load off... enjoy the holidays. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Match a Numer - then continue with dialplan
On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug. Have you tried using the SetVar cmd? I haven't tried it but it sounds like it might work for this. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Commercial Grade Service Provider?
Please do not take this as a flame against cyberdyne-ip.com. That is not the intention. I am just wondering how businesses like this expect to stick around when they are charging rates this low. You can find a whole list of other providers that thought this model would work at: http://www.voip-info.org/wiki/view/RIP+VOIP The fact is... if you want good quality, reliable service, and reasonable support, I think you should expect to pay a little more. I would be very cautious. Just my $0.02. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.0-beta4 Released
Can anyone help with the compile error? 1.4 beta-3 compiled fine under Fedora 6 w/IMAP storage. I get this on beta-4. [CC] app_voicemail.c - app_voicemail.o In file included from /usr/src/imap-2006c1/c-client/osdep.h:63, from /usr/src/imap-2006c1/c-client/c-client.h:42, from app_voicemail.c:65: /usr/src/imap-2006c1/c-client/env_unix.h:71: warning: function declaration isnât a prototype app_voicemail.c: In function âforward_messageâ: app_voicemail.c:3920: warning: unused variable âdurationâ app_voicemail.c: In function âplay_record_reviewâ: app_voicemail.c:7882: error: âvmsâ undeclared (first use in this function) app_voicemail.c:7882: error: (Each undeclared identifier is reported only once app_voicemail.c:7882: error: for each function it appears in.) app_voicemail.c:7882: warning: passing argument 8 of âimap_store_fileâ makes integer from pointer without a cast make[1]: *** [app_voicemail.o] Error 1 make: *** [apps] Error 2 [EMAIL PROTECTED] asterisk-1.4.0-beta4]# app_voicemail.c:7882: error: âvmsâ undeclared (first use in this function) Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.0-beta4 compile errors on Fedora 6
1.4 beta-3 compiled fine under Fedora 6 w/IMAP storage. I get this on beta-4. Any Ideas? [CC] app_voicemail.c - app_voicemail.o In file included from /usr/src/imap-2006c1/c-client/osdep.h:63, from /usr/src/imap-2006c1/c-client/c-client.h:42, from app_voicemail.c:65: /usr/src/imap-2006c1/c-client/env_unix.h:71: warning: function declaration isnât a prototype app_voicemail.c: In function âforward_messageâ: app_voicemail.c:3920: warning: unused variable âdurationâ app_voicemail.c: In function âplay_record_reviewâ: app_voicemail.c:7882: error: âvmsâ undeclared (first use in this function) app_voicemail.c:7882: error: (Each undeclared identifier is reported only once app_voicemail.c:7882: error: for each function it appears in.) app_voicemail.c:7882: warning: passing argument 8 of âimap_store_fileâ makes integer from pointer without a cast make[1]: *** [app_voicemail.o] Error 1 make: *** [apps] Error 2 [EMAIL PROTECTED] asterisk-1.4.0-beta4]# app_voicemail.c:7882: error: âvmsâ undeclared (first use in this function) Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk 1.4.0-beta4 compile errors on Fedora 6
I forgot to mention this is an x86_64 Pentium-D system. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime +Mysql +Failover
On 12/13/06, Rob Schall [EMAIL PROTECTED] wrote: Hoping someone out there has run into this or has some ideas for us. We currently have asterisk set up with Realtime (using mysql) for its extensions,sip and voicemail files. The problem we are trying to solve, is one of a failover mechanism. What if our mysql server went down. Can Realtime be set up with a secondary mysql server to get its data from. We can set up mysql to sync with its fellow server, and maybe when it goes down, it couldn't make any changes (write), but either way, you could still get the extension info, etc, so your phones would still survive a mysql outage. Any ideas? Thanks, Rob I don't think Realtime can be setup with a secondary server (someone please correct me if I'm wrong). Two possibilities come to mind... 1. You can run MySQL in an HA arangement with on box as the hot standby. 2. If you can allow for ocassional asterisk reloads, you could use Realtime Static Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRV Entries
On 12/13/06, Rob Schall [EMAIL PROTECTED] wrote: I saw on a mailing list for digium that back in March, they were looking to get SRV working properly. Was this ever repaired? If so, is it just a matter of adding 2 entries to tinydns data file, and then point the res_mysql.conf file to point to the new hostname (astmysql.yournet.com)? Trying any way possibly for redundancy. Rob Asterisk will do SRV lookups, it just does not fail to the next record if the first is unavailable as SRV was intended. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Input on Dundi
On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? We have been playing with DUNDi in a configuration similar to JR's whitepaper. Everything seems to be working fine but we have encountered a couple hurdles. Maybe others on the list have encountered these as well. 1.) When a registration server fails there doesn't seem to be an easy way to have clients automatically register to a new server. (our clients are mostly other asterisk boxes.) To solve this we are considering using DNS failover. 2.) If you plan to do any direct routing using the fullcontact address like what is shown in JR's whitepaper, you may find that fullcontact sometimes contains private network addresses. This makes it impossible to route inbound calls directly to the client. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Input on Dundi
On 12/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 11:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Input on Dundi On 12/12/06, Al Bochter [EMAIL PROTECTED] wrote: Ok, I am looking for some input on using dundi. Is anyone using dundi? And how is it working out? We have been playing with DUNDi in a configuration similar to JR's whitepaper. Everything seems to be working fine but we have encountered a couple hurdles. Maybe others on the list have encountered these as well. 1.) When a registration server fails there doesn't seem to be an easy way to have clients automatically register to a new server. (our clients are mostly other asterisk boxes.) To solve this we are considering using DNS failover. Wow. I remember when I raised this as an issue I was accused of being a Asterisk heretic. The solution suggested was to, increased load not-withstanding, bring your phone registration period right down. Thanks for the tip Doug. Even when registration periods are set low, the clients don't know which new server to register to. It would be great to use SRV, however most of our clients are also Asterisk boxes, and as you know Asterisk does not support multiple SRV lookups. We have used DNS failover on other services in the past and have thought to try this with asterisk. Basicly our clients would register to FQDN like reg1.mydomain.com, then when that box fails we'd have DNS re-direct that name to a the IP of reg2.mydomain.com. This seems to work with web and ftp, but I'm not sure how asterisk will respond. Any thoughts? David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Input on Dundi
On 12/12/06, JR Richardson [EMAIL PROTECTED] wrote:
1.) When a registration server fails there doesn't seem to be an easy
way to have clients automatically register to a new server. (our
clients are mostly other asterisk boxes.) To solve this we are
considering using DNS failover.
When registering with an Asterisk server to an Asterisk cluster of
servers, for the purpose of traversing a NAT or something else (to
solve a problem where direct contact cannot be performed), I would
suggest doing multiple registration to two registration servers, using
different names.
Like
registration [name1] to registration server 1
registration [name2] to registration server 2
in the outgoing dilaplan
exten = _NXXNXX,1,Dial(IAX2/server1..|j)
exten = _NXXNXX,102,Dial(IAX2/server2..
so if server one is not there the call will jump to the next server
or
exten = _NXXNXX,1,Dial(IAX2/server1IAX2/server2.
first server to answer will get the call.
you can do something similar calling from the cluster to the end Asterisk server
dundi lookup for [name1] if not available lookup [name2]
2.) If you plan to do any direct routing using the fullcontact
address like what is shown in JR's whitepaper, you may find that
fullcontact sometimes contains private network addresses. This makes
it impossible to route inbound calls directly to the client.
I recently started pulling the ipaddress and port from the database
instead of using the fullcontact field. Aaron Daniels helped me to
get the realtime query working instead of using the mysql connect
statements.
[lookupmysql]
include = invalid
exten = _X.,1,RealTime(sippeers|name|${EXTEN}|DN_)
exten = _X.,2,GotoIf($[${DN_ipaddr} = ]?${EXTEN},105:${EXTEN},3)
exten = _X.,3,Set([EMAIL PROTECTED]:${DN_port})
exten = _X.,4,Dial(SIP/${directdial},15,rj)
exten = _X.,5,Macro(sendtovm,${EXTEN})
exten = _X.,6,Hangup
exten = _X.,105,Macro(sendtovm,${EXTEN})
exten = _X.,106,Hangup
The RealTime command pulls all the entire record from the database and
prepends all the fields with the last argument (here is have DN_) so
when the record is pulled, all the records info is available as a
variable like DN_port and DN_ipaddr.
This is a really cool command. Hope this helps.
Wow, thanks for the examples JR. This is exactly what I needed. I was
not aware of the RealTime command. That will be very useful.
David
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[asterisk-users] IAX2 to SIP protocol translation overhead?
Just wondering if there is much CPU overhead in the translation from IAX2 to SIP, and how taxing this function is as compared to transcoding. We're trying to build an efficient system and would like to avoid taxing the CPU as much as possible. Our upstream service provider is 100% SIP, however we'd like to use IAX2 in our network as well, if it does not cause too much overhead. Not sure if it matters, but we will be running aprox 100 simultaneous calls. Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to communicated Both SIP and IAX2 each other ?
Yes, as long as Asterisk is in between the two, it can perform the protocol translation. regards David On 12/8/06, raviprakash sunkara [EMAIL PROTECTED] wrote: Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)---OpenSER-- Asterisk-- UA(IAX2) . UA(IAX2) --- Asterisk --- OpenSER -- UA (SIP ). other wise we can like that.. UA(SIP ) --- Asterisk-UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Switching from FreeBSD to Linux - which distro?
If you are new to CentOS or redhat based OS's, I would recommend using yum, as it will resolve any dependencies automatically. If you wish to install RPMS directly, you can download them from any CentOS mirror. See the CentOS website. Note: a default install of CentOS installs a bunch of unnecessary services that you will want to turn off using chkconfig service_name off. David On 12/8/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs great on it. Debian is good too. They have Asterisk packages, but they're generally a little bit old. Source installations work fine. Both have large, active developer and user communities. Hi Carla! Can you tell me from where do you download rpm's for Cent OS 4? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Switching from FreeBSD to Linux - which distro?
On redhat based OS's I would do this... You can run the following command to see what services are enabled: chkconfig --list | grep 3:on Then disable whichever ones you dont need... The services may vary a bit depending on hardware or what packages you have installed. I often disable everything except network, iptables sshd; like this... chkconfig acpid off chkconfig atd off chkconfig autofs off chkconfig cpuspeed off chkconfig cups off chkconfig gpm off chkconfig haldaemon off chkconfig isdn off chkconfig mdmonitor off chkconfig messagebus off chkconfig netfs off chkconfig nfslock off chkconfig pcmcia off chkconfig portmap off chkconfig rawdevices off chkconfig rpcgssd off chkconfig rpcidmapd off chkconfig anacron off chkconfig crond off chkconfig kudzu off chkconfig sendmail off chkconfig smartd off chkconfig syslog off chkconfig xinetd off chkconfig irqbalance off chkconfig microcode_ctl off chkconfig sshd on chkconfig iptables on chkconfig network on then reboot. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
Aaron, Could you please send me the scripts as well. Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime fullcontact field contains nat device private ip
I have noticed this as well. I have seen a few configs like your DUNDi setup, that use the fullcontact URI to directly contact a phone. I was always puzzled how everyone was making this work with NAT. I have not looked into it much yet, but I wonder if the new netfilter SIP conntrack/NAT extension might help overcome this issue? Regards, David On 12/3/06, JR Richardson [EMAIL PROTECTED] wrote: Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This eliminates the ability to use the fullcontact URI to directly contact the nat'ed phone. Works great for non-nat'ed devices. Is this by purpose or an oversight the way Realtime pulls the correct contact info in the sip registration header from the device? Does anyone know how to correct this behavior? It is the same with nat=yes or nat=no. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel-1.4.0-beta2 Getting it to compile on Fedora Core 6 _64bit
On 12/5/06, Steve Gladden [EMAIL PROTECTED] wrote: I keep running into the dead end that it can't find config.h in the source tree. I ran into this problem yesterday trying to compile ztdummy on FC6 i586. The latest Digium tarball gave me the config.h error. I was able to compile an SVN checkout of zaptel-1.4.0-beta2, but I was still unable to load ztdummy. At this point I received Unknown symbol. FYI: I did have full kernel sources available I'll try again later today... hopefully Tzafrir's quick-and-dirty fix will do the trick. David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk: SIP Gateway or Proxy
Asterisk can actually act as a Gateway and a SIP Proxy. This is where a lot of confusion comes in. It can do pretty much any voip function you throw at it. Definitely search the archives if you still have questions. try site:lists.digium.com keyword in google to search the mail archives. David On 12/1/06, yusuf [EMAIL PROTECTED] wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balance Asterisk servers?
On 11/14/06, Stelios Koroneos [EMAIL PROTECTED] wrote: JR Richardson gave a very nice presentation at Astricon on how to do that with DUNDI As I understand it JR Richardson's DUNDi solution does not support IAX. It uses regcontex which I believe is only available with SIP. (please correct me if I'm wrong) Also JR notes that... Associated SIP Users, business customers, require same registration and failover to the same servers so unless this is for a residential setup, it may not be of much use to you. Nevertheless it is great documentation, and may get you further than you are now. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balance Asterisk servers?
On 11/14/06, Aaron Daniel [EMAIL PROTECTED] wrote: Incorrect :) IAX2 most definitely does support regcontext. Also, I think what he means is the phone specific information must be exactly the same from system to system or the failover won't be as seamless as you expect. A lot of phones support some sort of SRV records, so in the event of a failure, the phones will automatically find the next available server. The other option there is to set up an HA environment so the failover is even transparent to the phones, they just start talking to the new IP address immediately. Another thought, in any failover situation, if you have any sort of automated failover, you must make sure phones that need specific features fail to the same server (i.e. hinting and such) as those features don't work cross server. Aaron On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: On 11/14/06, Stelios Koroneos [EMAIL PROTECTED] wrote: JR Richardson gave a very nice presentation at Astricon on how to do that with DUNDI As I understand it JR Richardson's DUNDi solution does not support IAX. It uses regcontex which I believe is only available with SIP. (please correct me if I'm wrong) Also JR notes that... Associated SIP Users, business customers, require same registration and failover to the same servers so unless this is for a residential setup, it may not be of much use to you. Nevertheless it is great documentation, and may get you further than you are now. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There I go spreading mis-information again. :) Thanks Aaron for clearing up the IAX2 regcontext question. That is good to know. In the DUNDi scenario, are there any other features that would be affected or become unavailable in the event of a failure? It seems like normal call processing would continue once the client re-registered to a different box, but I haven't tried it yet. I assume one could use DNS-Round-Robin to load balance registrations between the boxes in the cluster, then pull the failed box out of DNS to prevent registration attempts while the box is dead. Is there a better way to do this ??? David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi Asterisk Cluster
We use only IP connections to our asterisk boxes. Given this our origination/termination providers usually send/receive traffic to/from our network on a single IP or limited number of IPs. In a DUNDi Asterisk Cluster, would each of the boxes need to be able to connect to our origination/termination providers directly, or would we need to setup a common gateway box to forward calls to/from our providers? How is this type of routing best handled in an all IP environment? Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk iax2 monitoring
On 11/8/06, Thomas Blanchin [EMAIL PROTECTED] wrote: Hi David. I read your post on : http://lists.digium.com/pipermail/asterisk-users/2006-September/167456.html I am in the same situation as you are. I'm looking for a way to monitor iax2 connexions on asterisk. I'm using sipsak for sip connexions. I'm looking for a very simple tool, like sipsak, because I'm using BigBrother for global monitoring, so I just need an app that returns something or an exit code, and then BB do the rest. If you founded something, please let me know, I'm interested. Cheers. Thomas Hi Thomas, I have not found a good solution yet. I did find a tiny app called iaxping. http://rpm.pbone.net/index.php3/stat/4/idpl/3029621/com/iaxping-0-1mdv2007.0.i586.rpm.html This worked well to test the connection, but I could not get it to exit cleanly. I was thinking I might try to fix the code, but I'm not much of a C programmer. The source code (iaxping.c) is available online. I did not actually try to recompile the code so the problem might just be with the rpm. After you look at it, let me know your thoughts. If nothing else, maybe we get together and hire a decent programmer to fix the app to exit properly with a return code. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 SIP Monitoring Solution for Asterisk
Greeting Everyone, Just wondering if anyone has come up with a reliable method for remotely monitoring Asterisk boxes. I need to be able to check if Asterisk is actually providing service (registering clients, processing calls), not just answering to pings. In the past I have used sipsak in a cron script to do SIP registrations, but I haven't been able to find anything similar for IAX2. Any thoughts? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone for Windows Mobile 5?
Try SJphone, it works for me. http://www.sjlabs.com/sjp.html The latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] Softphone for Windows Mobile 5?
Yes, use it on WM5. Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, Christian On 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.html The latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] Softphone for Windows Mobile 5?
Sorry, poor reply. Yes I use it on WM5, and have not seen any problems. I admit I don't use it a lot, but it does seem to work fine. regards, Dave On 8/15/06, David Thomas [EMAIL PROTECTED] wrote: Yes, use it on WM5. Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, Christian On 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.html The latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g.711 Codec Question
Greeting Everyone, I don't have access to Asterisk box right now or I'd check this myself... If my client phone uses g.711 (alaw) and my outbound trunk leaving asterisk uses g.711 (ulaw), will asterisk have to transcode? If so is the processing overhead much? regards, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime SIP Registrations
Doug, If you'd be willing to share the patch and AGI, I would be happy to help test your solution. I know that myself and several others have been looking for a way to make Asterisk do this for quite some time. regards, David On 6/29/06, Doug G [EMAIL PROTECTED] wrote: Well, to dial a peer direclty the only thing that is missing in realtime is the status of the sip peer. (registered, Unregistered, unknown, reachable). If you dial a peer via ip and it is unavaliable you get dead air. So you need to know the status of the peer before dialing it. The change basicly updates realtime with the peers status. I did the same thing for IAX as well.. Doug From: [EMAIL PROTECTED] on behalf of Mike Lynchfield Sent: Thu 6/29/2006 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations can you elaborate on modify sip to update the status on the sip friends in realtime thanks On 6/29/06, Doug G [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial ( SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status in realtime data before you dial. This allows MANY Asterisk servers to share the same SIP data.I then load balance with DNS SRV.. Yes I have tested in failover it works. I too have been told that by many that this will not work. So I keep expecting to hit some problem with it, but to date I have not... Doug From: [EMAIL PROTECTED] on behalf of David Thomas Sent: Thu 6/29/2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP Registrations I think lots of us know about it... We're just not sure how to go about fixing it. :-( I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have never actually tested their system in failure scenarios, or they are working in a controlled environment without NAT and such... regards, David On 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.theclubvoip.com Making it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime SIP Registrations
I think lots of us know about it... We're just not sure how to go about fixing it. :-( I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have never actually tested their system in failure scenarios, or they are working in a controlled environment without NAT and such... regards, David On 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime SIP Persistency
Try googling the archives using the keywords rtcachefriends mwi. You should find more info about this. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback from VON expo! Info on * HAandPolycomphone!!
The only thing is I want to be sure I understand the statement above because the only time I can see Asterisk needing to do an SRV lookup is if it is handing a call to a carrier for termination. Gabe, that is what I was talking about. Asterisk really needs the ability to make use of the termination providers' SRV records for failover. When one doesn't respond... use the next record. This work exceptionally well when connecting to directly to a provider with an ATA that supports SRV. It is a shame that * doesn't do this. I have been tempted to grab the SRV lookup code from the Jabber project and try to merge it into Asterisk. http://www.jabberstudio.org/cgi-bin/viewcvs.cgi/cvs/jabberd/jads2s/srv.c?view=markup I kinda understand the Jabber code, but I really have no idea where to begin in Asterisk. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING
Brad, How are you able to overcome the Call-ID stickiness problem when loadbalancing with Ultramonky? As I understand it LVS does not properly support SIP in that it doesn't always use the same source path. regards, David On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote: At the moment I'm out of the office, but when I return I'll be certain to do that. Note that my solution is different from what you are working on with regexten, though I suspect some of the challenges that I've faced and overcome are not. I'm actually using UltraMonkey for load-balancing and failover of the Asterisk boxes, and my dialplan is set up so that it need not be changed when extensions are added or removed. I've been meaning for some time to do a write-up of how it all works, both in the hopes of giving my knowledge back to the community as well as learning some things that may help me improve the solution. I've had to make a couple of (minor) tweaks to both app_voicemail.c (to ensure proper password synchronization) and pbx_dundi.c. The latter is for larger sites(more than one cluster since my clusters are designed as buliding blocks of two Asterisk systems), adding a count flag to return the number of matching records and the ability to return a specific record. The former, I suspect, would be unnecessary if I were using Realtime for voicemail. But I'm not using Realtime for anything at the moment. Anyway, I apologize for not being able to answer you fully right now, but I've got this set to remind me to follow up on Monday. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, March 17, 2006 12:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: DUNDi Halfway and CLUSTERING Doug, I feel your pain. I have, since 3 days ago, all but giving up on dundi in a enterprise/carrier core scalable environment, mostly due to no ability to summarize dial plan routes across several servers that may or may not have contiguous extensions registered across the cluster. Example server 1 has exten 1234, 1235, 1001, 1002 registered and server 2 has 1236, 1237, 1003, 1004. But also in a failure or load balancing event, server 1 could have 1236, 1001, 1235, 1004 and server 2 could have 1237, 1234, 1002 and 1003 registered. The dynamic nature on maintaining extension state across many registration servers in real time, is not something dundi, realtime or regcontext/regexten can handle right now. At least I haven't figured it out yet. Brad, Please be so kind to publish your dundi.conf configs and also dundi snips from extension.conf for server 1, server 2, server3.. Thanks Guys. JR -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 16, 2006 9:42 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 20, Issue 114 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com Could you perhaps post your dundi.conf for both boxes? I'm afraid this message doesn't mean anything to me, but I have about a dozen boxes doing DUNDi peering so I know what the config should look like. But it's basically always worked for me. Regards, - Brad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sticky Problem SER/Asterisk
I'm sure there is some briliant person out there that can figure this out, but I have gone over this so many times it's just a headache. I was able to make a decent setup using OpenSER w/Asterisk only as voicemail, but I really needed to support IAX as well so I'm back to Asterisk again. Depending on the call features you need, SER/Mediaproxy can do just about everything Asterisk can in regards to SIP. You do miss out on a lot of the traditional PBX functionality though. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote: Do you mean the peristence of connecting a specific phone to a specific server? If so, then it's relatively easy. The ldirectord has a persistence setting that does that. If I'm misunderstanding you, then could you explain further what you mean? Have a look at this doc from the developers of LVS... http://www.austintek.com/LVS/LVS-HOWTO/HOWTO/LVS-HOWTO.services_that_dont_work_yet.html This has been discussed a few times in the SER list. The issue is specific to UDP which includes the majority of SIP endpoints. Currently LVS does not always send reply packets from the addresses that they were received on. This breaks SIP dialog. If you have found a way around this, that would be great. Please let me know. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote: I understand what you're saying now. While I have absolutely no proof of this, I have to believe that it's something they've solved. I've got several production systems (since early December of last year) using the type of cluster that I'm talking about, and I've yet to hear of any issues that could be related to this. I also did extensive testing both in the lab and at the first production site with a lot of debugging information and never saw this sort of thing. Sorry I don't have any magic for you, but for me it just worked. That was actually the easiest part of the whole solution. That is encouraging! What type of environment are you working with? Are your SIP clients behind NAT? Is this an Asterisk only setup? I guess I will have to do more testing to see what has changed. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycom phone!!
In regards to HA... SER is definitely a good option, but it does require the extra hardware to have at least 2 boxes that can failover on each other. I would user OpenSER however (better documentation and mor features). I couldn't agree more that Asterisk should FULLY support DNS-SRV. The solution seems to work great for phones and ATA's. This would be a good item to create a bounty for. I have only two boxes right now, so it seems like my only HA options are the dreaded DUNDi setup or a active/passive failover with linux-HA. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI, SER and asterisk
There is a patch to chan_sip on voip-info.org that I use. It seems to work very well. I believe it is on the Astrisk at large page on the voip-info.org wiki. regards, Darvid On 3/7/06, Sharon [EMAIL PROTECTED] wrote: I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI, SER and asterisk
Method 3 is the one I was speeking of. As long as you plan to continue to have SER in front of Asterisk it should be fine. David On 3/8/06, Christian B [EMAIL PROTECTED] wrote: Hi Sharon! This is pretty difficult, i was not able to implement it so far(though my ser-skills are pretty basic). At http://www.voip-info.org/wiki-Asterisk+at+large you'll find some howto's, method 2 seems to be the most promising to me... regards christian On Tue, 7 Mar 2006 15:36:57 -0600 Sharon [EMAIL PROTECTED] wrote: I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware Requirements for 1M minutes
I'm doing an install for a client with the following requirements. - 1 Million minutes of outbound calling - Calls come in to asterisk via SIP/IAX and terminated to third party provider via SIP - Codec usage will be about 70% g711 30% g729 (there should be no transcoding) - 100% IP setup with no voice cards in the box They have a box on hand with a single 3.2ghz P4 w/Hyper-threading, 2GB RAM Dual 10/100 card. The question is... Will their current system be OK for them? If not, what would you recommend? I realize I may be leaving out some needed info, hopefully this is enough to go on. regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Requirements for 1M minutes
Sorry, I saw that right after I posted. It is per month. And almost all during business hours. regards, David On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: On Mar 3, 2006, at 9:49 AM, David Thomas wrote: I'm doing an install for a client with the following requirements. - 1 Million minutes of outbound calling Per what? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conditional canreinvite
That is the way way SER works. I too am very interested to know if this can be done with Asterisk. David On 1/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho, correct and logical), but, is possible to combine both, so do reinvite only within e.g. one context and disable reinvite when connecting phones between two context, or any better option exist/planned how to solve? thanks PJ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 - the party's over :-(
I heard it was actually Longhorn Embedded VoIP version with free automatic updates if buy 500 CALS and software assurance. -D On 1/23/06, Cory Andrews [EMAIL PROTECTED] wrote: Anyone have a conspiracy theory or two to roll into this thread? Cisco is actually coming out with a new line of gateways that only support IAX, and will be porting their entire Callmanager platform to IAX. The best thing about these gateways is that they will actually be running an embedded version of Microsoft Windows 98 SE (Second Edition). Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-users-list asterisk-users@lists.digium.com Sent: Monday, January 23, 2006 6:50 PM Subject: RE: [Asterisk-Users] SPA-3000 - the party's over :-( I can't speculate as to why their sales of Linksys/Sipura products have been restricted, but as a Linksys VAD I can say we are not under any such restriction at present. its pretty obvious, linksys/sipura are shifting to selling primarily to service providers who would provide service-locked ATAs to end users. sipura telegraphed their intent a long while back by withholding auto-provisioning documentation from anyone except service providers, and now they have completed the move by no longer allowing sales to end users at all. That seems to be right in line with Chamber's objective to be a major player in the home market. He's certainly not going approach that objective by selling one/two devices at a time, so it makes sense he'd change the sales/marketing approach to focus/lock-in higher volume customers/resellers regardless of what the rest of us think. That certainly isn't the last shoe to drop in the voip market; wait till the next level(s) of announcements from Cisco. If I were going to bet a couple bucks on this, I'd suggest the spa3000 will disappear alltogether, and a replacement in the form of a linksys box with a faster processor is not far behind. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SER for dummies ?
Are, Are you using a common database for SER and Asterisk? How are you keeping the accounts synced? Does this setup cause any complications with AstBill? regards, David On 10/25/05, Are [EMAIL PROTECTED] wrote: Good Question. We have tested it with any combination we can think about and it is working safely. There is no way (we know about) that you can pass toll free calls. :-) Basically SER is configured to only accept clients that have the same callerid as account numbers so SER refuse to pass the call if you try to be smart. Asterisk only passes the call if you have a valid account and the request is handed over from the SER server. Asterisk determine the max length of the call based on the Users Account balance in AstBill. Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com On 10/25/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2005-10-25 at 08:27 +0100, Are wrote: The authentication in Asterisk is done using ANI/CLI. Same way as broadvoice, wonder if using that setup if I set my caller id to someone else will it cause the INVITE that broadvoice does (broadvoice will invite the person registered as that account if you try to make a call on their CID, asterisk ignores that invite, I am not so sure if all devices will) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQBDXeNg+1olxlzQw5cRApWJAJ4sXCutFLLuAk26jzumrS/ioMiZ3ACfa8zZ IBWJRwuEQ1RN9EqRvajQG/c= =DzJ5 -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA failover between datacenters
Hi Everyone, Does anyone know of any ATAs that can do proxy failover without using SRV. I don't want to rely on dns if at all possible. Basically, I have Asterisk boxes in two different data centers and I need ATAs to be able to uses the server at DC2 if DC1 goes down. The servers are already in a HA setup at each datacenter. I am looking for added protection if one of the datacenters becomes unreachable. The perfect solution I believe, would be an ATA that would failover to an alternate proxy if the first was unavailable, then failover to POTS if no proxies were available. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk SIP architecture question
Does asterisk have support for SIP session timers? David On 11/24/05, Olle E. Johansson [EMAIL PROTECTED] wrote: Matt Riddell wrote: Kevin P. Fleming wrote: Matt Riddell wrote: So how does Asterisk know that the media stream has been disconnected between the two remote hosts? It doesn't... nor does any other SIP softswitch. See my other reply for a possible solution. I agree that you could code a fix, but saying my advice is bogus because you could code a fix for Asterisk to avoid it is slightly wrong. The fact remains, if you need *very* accurate cdr's then you either don't do canreinvite=yes for the peer or you code something so that Asterisk notices that the rtp has stopped. The fact remains that without these, the most accurate CDR is going to come from the provider. If the audio goes through asterisk without re-invites, you could use the rtptimeouts to detect a dead phone. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP architecture question
When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn. Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? regards, David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk SIP architecture question
Thanks for the information Matt! Does asterisk store any SIP dialog cdr info in mysql like Call-ID Cseq? With This info I could at least detect runaway calls and fake a BYE to the pstn gateway with an external app. regards, David On 11/23/05, Matt Riddell [EMAIL PROTECTED] wrote: David Thomas wrote: When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn. Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk SIP architecture question
Kevin, Is the CDR accounting done based on SIP signaling? If a UA is talking (RTP) to a third party PSTN gateway, isn't it at risk if say the UA loses power. How will asterisk know the call has ended if it is not involved in the media path. The idea is this.. I want to use canreinvite =yes to force users to talk end-to-end to preserve bandwidth, but I can see the potential for hung calls if asterisk never get the BYE from a UA in the event the ATA gets unplugged or somehow loses power. regards David On 11/23/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Matt Riddell wrote: No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs Sorry, this advice is bogus :-( SIP re-INVITEs do _not_ affect the CDRs in any way, period. They only affect the media streams. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk DNS SRV lookups
Does asterisk fully support DNS SRV lookups yet, or does it still only read the first SRV entry? Info on the wiki looked quite old, so I thought I better ask. regards David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PAP2-NA and SRV
Has anyone had success using the SRV functionality in Linksys PAP2-NA's? Regards, David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
