[Asterisk-Users] sip invite timeouts
Is there a way in asterisk to configure a sip invite timeout ? It seems to be about 30 seconds right now which is too long. I would like to have asterisk return congestion if a host does not respond to an invite within 5 seconds. ___ --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-biz] http://www.87810.com/
Well.. you have to be a member of www.goiax.com for someone to call the 87820 number that is assigned. Of course it is free, so that isn't a huge barrier. It's basically like Free World Dialup, although I allow regular US dids to be assigned for free as an option. yours, matthew Rehan Ahmed AllahWala - Super Technologies I wrote: So how do u route them ? I mean if i want to route calls to you, what does one do ? Rehan I tried to use those numbers as an option for my www.goiax.com project. However, Telcordia wants $3.00 per month per number for a number that most people can't even call... so I made up my own number [87820]. :-D The ITU can put that in their pipe and smoke it. Rehan Ahmed AllahWala - Super Technologies I wrote: Have any one used http://www.87810.com/ or has a +87810x number running? Whats the news on this ? Rehan Super Technologies Inc., Pensacola, Florida http://www.supertec.com - Technologies from tomorrow, TODAY! http://www.VirtualPhoneLine.com - Get A US, UK, EU Number, Forward it to PSTN, SIP or IAX2 number, or Asterisk Superb Web Controls. http://www.PhoneOpia.com - SIP Based OPEN Phone Services. http://www.MySuperPhone.com - The NEXT Generation of Telephone. Http://www.ip-pabx.com - Ip Centrex System, with global service. Http://www.superPBX.net - One World, One Number, One Pabx, Physical. http://www.didX.org - World's First DID Number Exchange / Peering Service. ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz Super Technologies Inc., Pensacola, Florida http://www.supertec.com - Technologies from tomorrow, TODAY! http://www.VirtualPhoneLine.com - Get A US, UK, EU Number, Forward it to PSTN, SIP or IAX2 number, or Asterisk Superb Web Controls. http://www.PhoneOpia.com - SIP Based OPEN Phone Services. http://www.MySuperPhone.com - The NEXT Generation of Telephone. Http://www.ip-pabx.com - Ip Centrex System, with global service. Http://www.superPBX.net - One World, One Number, One Pabx, Physical. http://www.didX.org - World's First DID Number Exchange / Peering Service. ___ --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 high load high availability servers
anyone using a high availability server set up for Asterisk ? I saw IBM had some kind of solution at VON but was too busy exhibiting to check it out. :( ___ --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] free dids on goiax.com
GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --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] more dids added to goiax.com
To address the issue about the Firefox browser, I develop [if you want to call it that, lol] the sites on Opera so it should work fine in Firefox. Nothing fancy at all just HTML 4.0 forms, etc. The other poster hit the nail on the head, it was out of DIDs. I just loaded another 200 random dids in so we'll see how long those last. I'm going to add code to confirm the email addresses that people signed up for to avoid DID hogging and also going to reclaim DIDs that go totally unused for more than probably a 30 day period to keep DIDs from going stale and unused. And about the free outgoing, I don't want people to jump through hoops at all. That's not what this was supposed to be about. I don't want people to have to use something like Paypal either, that's irritating and a lot of people used the service from out of the country [like some US Marines calling home from Iraq] that couldn't use paypal anyway. I do like the Web of Trust idea. Maybe I'll do it google style and give out some invites to people [say 5] and then those 5 people will get a couple invites, and so on. Each invite would point back to the person who gave it so if I trust Bob and he trusts Sam who trusts Sue but Sue turns out to be an abuser, I can see that Sam give it to Sue and revoke Sue's account and make her untrustworthy to the system [blacklisted]. If Sam invites several people that all end up black listing, than Sam's ability to get invites will get removed [but he would not get blacklisted since he is not abusing, just passing out invites to abusers]. This would have to all be done by machine to scale ... not sure I have the time to put the code together but if it worked it might be useful for other free services out there. What do you guys think? yours, matthew ___ --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: goiax expanded with free us domestic calling
Joe Stewart wrote: On Fri, Sep 23, 2005 at 11:12:24AM -0700, Matthew Simpson wrote: I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Thank you very much for setting up this service. I've successfully made calls, but unlike my other iax trunks the callerid isn't passed on so the call comes in from areacode 202. Any hints to get this working? The caller ID thing is intended behavior. Passing the 87820-xxx number doesn't usually show up so it will come up as the 202 number. Currently the site hands out a virtual 87820-xxx number but I intend to add the ability to get a free United States DID [possibly other countries as well] as well. Please test it out. You can use an IAXy, asterisk, or an IAX softphone like iaxcomm. I've only used asterisk. If I have a chance I'll try a softphone. Any chance of g729? I know that since this is iax your options would be more limited as far as licensing. GSM is available. It takes up far less CPU than G729 and is about equal in quality and bandwidth usage. just wanted to send you a note and say thanks, Joe ___ --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
Re: [Asterisk-Users] goiax caller ID
Kevin Scott wrote: I'm not sure what he/she was sending as the caller ID information, what I was trying to do, was send a normal 10 digit number as caller ID. Is there any solution to this? Or anything planned? There are no plans to allow just any caller ID to be sent. Once US dids are available than the DID cid would be sent instead. ___ --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
Re: [Asterisk-Users] Re: goiax expanded with free us domestic calling
Steven wrote: Can I ask how you are providing calls to us domestic numbers for free? goiax.com is backed by TxLink [www.txlink.net]. We terminate a lot of minutes. ___ --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] goiax expanded with free us domestic calling
I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Currently the site hands out a virtual 87820-xxx number but I intend to add the ability to get a free United States DID [possibly other countries as well] as well. Please test it out. You can use an IAXy, asterisk, or an IAX softphone like iaxcomm. I intend to also modify asterisk to allow some QoS checking to avoid the problems IaxTel had with scalability. I would like some developer input on this. My idea right now is to add an app to asterisk that listens on a port for a status packet that will be sent from a softphone or IAX-compatible device. Asterisk would then reply with a packet that would contain current CPU load, current mem load, and number of channels up at that point. I would also like to have the asterisk app know the status of other IAX servers on that network and be able to reply with the IP address of the best available server at that time. ___ --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
Re: [Asterisk-Users] custom ring tone
Yeah. It's a brilliant idea because I believe they would probably return answer supervision to play these custom ring tones therefore creating more revenue from the incoming calls. Marko Rakar wrote: Few weeks back local telco introduced option of custom ring tones. I am not talking about your phone ring tones but about ring tones you hear in your headset while phone is ringing on the other end. If I understand correctly, ringing tone is generated localy on asterisk if you are connected to phone network with E1/T1 connection. Which means that instead of regular beep-beep tone we could send something else to the caller in PSTN (like mp3 music). Is there a way of customizing ring tone in asterisk and if yes how? Long, long ago in a galaxy far away, in General Hospital born I was, and quite happy were my parents, but when a youngling still I was, moved we did. mailto:[EMAIL PROTECTED] http://printel.hr ___ --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
Re: [Asterisk-Users] AstriCon 2006 Location
Actually, we have thought about that too. The problem is scheduling so as not to conflict with any of the other shows that our major sponsors are doing (remember, they help keep the cost of the conference down). Within roughly week of AstriCon on one side or ther other we have the CompTel/ASCENT show, WISPCON, ISPCON, Telecom '05, Phreaknic, ITU Americas, and TMC's Internet Telephony. VON is only 3 weeks prior. In general the fall is chock-full-o telephony shows. I'm skipping Comptel this year to go to Astricon. I tried to get out to Orlando for just one day but the flights from LAX to MCO are ridiculously long time wise, and they all landed outside of exhibition hours. ___ --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
Re: [Asterisk-Users] AstriCon 2006 Location
If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) quoted for truth. $79 flights from Dallas. ___ --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
Re: [Asterisk-Users] Re: Who is going to AstriCon (The Asterisk Conference)?
So far we've had at least one person indicate that they would not want to travel to the US at this time. All politics aside, how many out there feel the same way? I do NOT want to start a polical flame-war, but I am curious at the number of people who simply won't come to the States due to the current Administration, the Iraq war, or some other reason. That's just ridiculous. LOL. ___ --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
Re: [Asterisk-Users] AstriCon 2006 Location
Actually, we're based out of Kansas City, Missouri (NOT Kansas - we believe in science) so Omaha would be pretty convenient for us -- it's the other 98% of the community which would have four connections. ;-) Atlanta is hub for Delta and Airtran Dallas is hub for American Chicago is hub for ATA All good central locations with cheap non stop flights. ___ --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] free IAX calling platform
Hello all, I have set up a free IAX calling platform similar to FreeWorldDialup/IAXtel. You can access it at http://www.goiax.com/ The website is still very beta but it will allow you to sign up for a virtual phone number, and you can make outgoing calls to US toll-free numbers. There is also a conference bridge set up. Codecs are G.711 and GSM. I intend to allow users to choose united states DIDs for free in the future after I get the website polished up a little bit. The backend is using Asterisk Realtime. If anybody has any ideas for improvements let me know. yours, Matthew TxLink ___ --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
Re: [Asterisk-Users] Who is going to AstriCon (The Asterisk Conference)?
I will be there. Matthew Simpson TxLink-Commpartners Steven Sokol wrote: Hi, I'm taking a straw-poll to see who out there is planning on going to AstriCon. I would like to hear from both new members of the community and gurus. What kinds of things would you like to see at an Asterisk Conference? What topics are good BOF (Birds Of a Feather - informal discussion group) fodder? What parts of Asterisk require the most attention? FYI - AstriCon is October 12 - 14 in Anaheim. For more information on what we currently have planned, see the web site (listed below). Thanks, Steve ___ --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] realtime caller id extension matching
I'm trying to copy the functionality of something like this in
extensions.conf [extension matching on callerid]:
exten = _NXXNXX/7065557230,1,NoCDR
exten = _NXXNXX/7065557230,2,Dial(Zap/g1/${EXTEN})
exten = _NXXNXX/7065557230,3,Busy
in realtime:
| 16 | fb | _NXXNXX/7065557230 |1 | NoCDR | |
| 17 | fb | _NXXNXX/7065557230 |2 | Dial | Zap/g1/${EXTEN} |
| 18 | fb | _NXXNXX/7065557230 |3 | Busy | |
however realtime seems to match on only the _NXXNXX, making this code
execute on all extensions, ignoring the caller id matching. Do I need to
use something other than / to delimit the CID field ? Or is this not
supported in realtime?
[i'm using mysql driver]
- matthew
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[Asterisk-Users] re: realtime caller id extensions matching
answered my own question from: pbx_realtime.c The realtime table currently does not support callerid fields. ___ 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] cisco 7960 question
I have a stupid question. How do you remove line presentations on a cisco 7960 ? I have 3 line presentations I don't use anymore and I can't figure out how to remove them. ___ 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] zap a sip channel
Is there anyway to destroy a sip channel ? I get hung up channels like this in sip show channels: 67.153.9.20 2145558260 33ae28f6088 00102/0 unknow 67.153.9.20 2145558260 52be8085005 00102/0 unknow 67.153.9.20 2145558260 4653d937578 00102/0 unknow 67.153.9.20 2145558260 2f9bc49d03a 00102/0 unknow ___ 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] dialplan question
Hello, I have a dial plan that tries to place a call over several different outbound gateways, like this: exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1X.,2,Dial(SIP/[EMAIL PROTECTED]) exten = _1X.,3,Dial(SIP/[EMAIL PROTECTED]) exten = _1X.,4,Dial(SIP/[EMAIL PROTECTED]) exten = _1X.,5,Hangup it works fine if one of the gateways is busy [rolls to the next dial statement]. However, if the phone number itself on the proxyX gateway is PSTN-busy, then it correctly returns 486 busy here, but execution continues, which wastes trunks trying a busy number on each gateway. What is the best way to handle this? Inserting +101 extensions with the Hangup command ? Will that still properly signal 486 busy here back? Should I be using Congestion instead of Hang up ? ___ 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] TE-405P freezing, anyone else?
Hello list, I have about 20 Digium TE-405Ps out in the field, and I started having trouble with one just recently. The card had worked fine for a month with 4 PRIs in NFAS configuration, and then all of a sudden I started getting a disappearing D channel. A restart of asterisk / ztcfg /module unload-load did not fix the problem, but a reboot [not power off, just a restart] would bring it back online. The card would then work for another day and then poop out again, no D channel, nothing coming in from a pri intense debug. Asterisk itself was working fine, nothing strange in the verbose or debug logs. I RMA'd it back but just wondering if anyone else had this problem ? One of our customers told me over the phone that he had the same issue and that he had heard that it was an issue with the firmware on the newer TE-4x0P cards. BTW, I did a trace on what the card was sending with a protocol analyzer... card was sending out all 1s. A new TE-405P seems to be working okay. Matthew Simpson TxLink Communications http://www.txlink.net/ + SIP and IAX origination and termination + Unlimited incoming toll-free $20/LATA + Texas origination and termination for $0.005/min + US origination and termination $0.005 to $0.012/min ___ 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] echo cancelation on Digium T1 cards
Hello all, I am getting console debug messages about tone detected on channel XX, disabling echo cancelation on channel XX when using echocancel=yes with a Digium T1 card. does this mean that DTMF breaks the echo can? Does Asterisk permanently disable the echo can or is it for that channel instance only? ___ 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] Verisign SIP7 sip--ss7 service
Is anyone using the Verisign SIP7 SIP -- SS7 service with Asterisk? Does anyone have a Verisign contact? yours, Matthew Simpson TxLink Communications www.txlink.net/ IAX and SIP Origination and Termination Services ___ 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 and level3
We are in the process of mating our Lucent softswitch to Level3. I am also wanting to put some Asterisk equipment in this configuration. Has anyone done interop testing with Level3 and Asterisk? Any issues ? I am about to start next week. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call files
Hello, I am having trouble with call files. I want my call files to attempt only 1 time, and never retry. I am trying to bridge two calls together, one call to my office [9726172877] and the other call to my cell [2022463521] My call file looks like this: Channel: IAX2/outgoing/19726172877 SetVar: ringtime=30 Callerid: 8668954650 MaxRetries: 0 RetryTime: 0 WaitTime: 0 Context: outgoing Extension: 12022463521 Priority: 1 Everything works properly, but I get these errors: Nov 15 23:16:04 WARNING[180235]: pbx_spool.c:156 apply_outgoing: Invalid retrytime at line 5 of /var/spool/asterisk/outgoing/2d4f5de784381a423b13480003a39a6434d63f96.call Nov 15 23:16:04 WARNING[180235]: pbx_spool.c:161 apply_outgoing: Invalid retrytime at line 6 of /var/spool/asterisk/outgoing/2d4f5de784381a423b13480003a39a6434d63f96.call and sometimes Asterisk tries to call my office again while the call is bridged. What is the proper way to accomplish this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] answer on # key?
I thought I read somewhere on the Wiki that one could give Dial() an argument that would first dial the extension, but not bridge the connection until the called party hit the # key. It must have been during one of those late night coding sessions because now I can't find anything to do with that other than options to allow hangup of the call by hitting *. Does such an option exist? If not, is anyone using a Macro to do that? I have a system that attempts to do a Dial out to a cell phone number with a 15 second timer as a find me type of application. If the cell phone is off or out of range, the 15 seconds of ring time isn't reached and the caller gets connected to the cell phone's voicemail instead of the Asterisk voicemail like I want. Having the # to connect option would fix this problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream bt-486 can only dial with #
I have a grandstream BT-486 in the lab running 1.0.5.11 firmware. For the past three days I've had no trouble dialing out without hitting #. I had the setting for using # as dial key to no in the config. Today the BT wouldn't pass outgoing calls. I turned on # as dial key and it works now if I hit # at the end. I have changed nothing on the BT-486 and nothing on the * box it is connected to. Anybody seen this happen before? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] new ATA box for sale by Linksys
Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a User Guide meant for end-users that is very well written [no Engrish] and explains the calling features and install well. I imagine that the wholesale price of these ATAs will be very attractive if they are selling for $60 retail! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new ATA box for sale by Linksys
From your experience, could you give us the merits and demerits of these ATA devices as well as the IAXy. They are essentially a Sipura SPA-2000. One of my customers uses the Sipura exclusively for his customers and they work very well. Setup is easy, and they support the CLASS type features superbly. Thanks to everyone who cleared up the PAP2 versus PAP2-NA. I am not sure if the one bought at Fry's is the NA version or not. I didn't buy it, my customer did. If it's the Vonage-locked version I'm sure he'll return it. I do know that the only version Frys had on the shelf had a Vonage sticker pasted on it, but the side of the box seems to indicate that it will work with any SIP phone service provider. Matthew N. Simpson TxLink Communications SIP/IAX VoIP Origination and Termination Minutes as low as 0.005/minute www.txlink.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 transfer and CDRs
Does IAX2 properly update call records for transferred calls to another IAX2 server? Or should I still be using notransfer=yes ? Example: SERVER1 calls SERVER2 which transfers call to SERVER3 If Call records are pulled from Server2 will that call have proper CDRs? The Wiki says no. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stale voicemail messages / greeting
I'm using Asterisk to read voicemail users out of a SQL database. I am
assigning users real phone numbers as their voicemail box. The problem is
that if I re-assign a phone number (say, 972-245-0001), the new user is
stuck with the old user's greeting and saved messages. What is the best way
to resolve this?
I don't want to use unique mailbox ids because my dialplan looks like this
in the incoming DID context
[incomingdids]
exten = _972245,1,setvar(boxnum=${EXTEN})
exten = _972245,2,VoiceMail(u${EXTEN})
exten = _972245,3,Hangup
exten = a,1,VoiceMailMain(${boxnum})
exten = a,2,Hangup
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[Asterisk-Users] new Asterisk resources site
hello everyone, I have compiled a few scripts (PERL AGI and PHP management) I have written for Asterisk together and put them up at: http://www.txlink.net/asterisk.php Just click on Resources to skip directly to the scripts. The rest of that page is meant to be an Asterisk introduction to potential new Asterisk users, and I plan to add some additional tutorials to it. If anyone has any new-user type links that I do not have on the page already, please email them to me for addition to the page. If you would like your company to be listed on the Case Studies section, please respond off-list to me as well. yours, Matthew Simpson TxLink Communications www.txlink.net/ Asterisk PSTN Origination and Termination -- Connect your * to the public telephone network ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is chan_skinny broken?
I am trying to use chan_skinny but when loading the module I get: [ Booting../usr/lib/asterisk/modules/chan_skinny.so: undefined symbol: ast_pickup_call I am using CVS 07/23 I can't get chan_sccp2 to compile, it gives me parse errors, or I'd be using that. :-/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hang up when going to voicemail
Doh! The reason it changed when I upgraded is because I was compiling VM with Mysql, and I had the mailbox definitions in the voicemail.conf flat-file. I put the definition in the SQL database and it works fine, now. :-/ thanks for kicking me into the right direction :) yours, matthew Are you sure you have a mailbox for this number ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Simpson Sent: 23 July 2004 16:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] hang up when going to voicemail I have a little menu set up where hitting 1, 2, or 3 places the call through to a cellular phone over IAX. That works. However, if caller hits 4 to go into voicemail, the system hangs up. Am I doing something wrong in the dial plan, or is this a CVS change? I had no trouble with this until I upgraded to about 07/21 CVS, and I'm on 07/23 [latest] now with same results. My dial plan: [txlink] exten = s,1,Answer exten = s,2,Background(/txlink/txlink-main) exten = 1,1,Dial(IAX2/:[EMAIL PROTECTED]/12149490280) exten = 1,2,Hangup exten = 2,1,Dial(IAX2/:[EMAIL PROTECTED]/14693373687) exten = 2,2,Hangup exten = 3,1,Dial(IAX2/:[EMAIL PROTECTED]/18174017579) exten = 3,2,Hangup exten = 4,1,VoiceMail(s2147649296) exten = 4,2,Hangup exten = t,1,Goto(txlink,s,2) exten = i,1,Playback(invalid) [didin] exten = 2147649296,1,Dial(SIP/2147649296,15) exten = 2147649296,2,Goto(txlink,s,1) exten = 2147649296,3,Hangup Here is console output: -- Executing Goto(SIP/2147649296-fb41, txlink|s|1) in new stack -- Goto (txlink,s,1) -- Executing Answer(SIP/2147649296-fb41, ) in new stack -- Executing BackGround(SIP/2147649296-fb41, /txlink/txlink-main) in new stack -- Playing '/txlink/txlink-main' (language 'en') == CDR updated on SIP/2147649296-fb41 -- Executing VoiceMail(SIP/2147649296-fb41, s2147649296) in new stack -- Executing Hangup(SIP/2147649296-fb41, ) in new stack == Spawn extension (txlink, 4, 2) exited non-zero on 'SIP/2147649296-fb41' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hang up when going to voicemail
I have a little menu set up where hitting 1, 2, or 3 places the call through to a cellular phone over IAX. That works. However, if caller hits 4 to go into voicemail, the system hangs up. Am I doing something wrong in the dial plan, or is this a CVS change? I had no trouble with this until I upgraded to about 07/21 CVS, and I'm on 07/23 [latest] now with same results. My dial plan: [txlink] exten = s,1,Answer exten = s,2,Background(/txlink/txlink-main) exten = 1,1,Dial(IAX2/:[EMAIL PROTECTED]/12149490280) exten = 1,2,Hangup exten = 2,1,Dial(IAX2/:[EMAIL PROTECTED]/14693373687) exten = 2,2,Hangup exten = 3,1,Dial(IAX2/:[EMAIL PROTECTED]/18174017579) exten = 3,2,Hangup exten = 4,1,VoiceMail(s2147649296) exten = 4,2,Hangup exten = t,1,Goto(txlink,s,2) exten = i,1,Playback(invalid) [didin] exten = 2147649296,1,Dial(SIP/2147649296,15) exten = 2147649296,2,Goto(txlink,s,1) exten = 2147649296,3,Hangup Here is console output: -- Executing Goto(SIP/2147649296-fb41, txlink|s|1) in new stack -- Goto (txlink,s,1) -- Executing Answer(SIP/2147649296-fb41, ) in new stack -- Executing BackGround(SIP/2147649296-fb41, /txlink/txlink-main) in new stack -- Playing '/txlink/txlink-main' (language 'en') == CDR updated on SIP/2147649296-fb41 -- Executing VoiceMail(SIP/2147649296-fb41, s2147649296) in new stack -- Executing Hangup(SIP/2147649296-fb41, ) in new stack == Spawn extension (txlink, 4, 2) exited non-zero on 'SIP/2147649296-fb41' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: incoming calls on Cisco 7960
From: Randy Bush [EMAIL PROTECTED] [214] disallow=all allow=ulaw type=friend secret= host=dynamic nat=no dtmfmode=rfc2833 canreinvite=no incominglimit=1 mailbox=214 where is the context= to send it to an incoming context? In the general part I have context=from-sip I don't have separate contexts for each SIP device due to the way I have this configuration set up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming calls on Cisco 7960
Hello list, I have a Cisco 7960 with SIP Image 7.1. I can make calls outgoing through Asterisk, but I'm having problems with incoming calls from Asterisk. The phone is on a public IP address, no NAT, no firewall. The phone is registered and shows up in sip show peers. If I place a call to the phone, Asterisk sends invites to the phone in vain, and then gives up. I can use my soft phone and place a call to the phone IP to IP [EMAIL PROTECTED] Are there any known issues with this firmware? I'm using latest CVS, and it also does not work with July 1 CVS. Here is my sip.conf: [214] disallow=all allow=ulaw type=friend secret= host=dynamic nat=no dtmfmode=rfc2833 canreinvite=no incominglimit=1 mailbox=214 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DISA and AGI: authenticate by caller ID? (resolved)
Here is some code to do authentication by caller ID for DISA through AGI.
My original code had a bug in the Mysql query code, and there was a hangup
in the wrong place
[that's what I get for coding something at 2:00am], but the attached code
works correctly.
Take note of the REGEXP for the CallerID variable. When I tested the code
from the PSTN
it worked because there was no name component, but it broke from SIP. If
you're calling SIP--SIP you'll have the name in
that variable as well as the number, so I added code to snip everything but
the 10 digits. Adjust accordingly if you have more or less
than 10 digits. Also, I've thought of a bug already, if your caller ID name
has digits in it, it'll break the regexp. Adjust accordingly
if that is true about your installation.
Yours,
Matthew Simpson
TxLink Communications
IAX/SIP Termination and Origination
Wholesale Dialup Services
[EMAIL PROTECTED]
972-617-2877
http://www.txlink.net
You'll need a context called ldincoming [or equivalent] for the AGI to
transfer access to DISA like:
[ldincoming]
exten = 1011,1,DISA(no-password|disa)
exten = 1011,2,Hangup
You'll need a context called disa [or equivalent] with what you want to
allow the authenticated callers to access, mine looks like:
[disa]
include = tollfree
include = localonly
and then just call the agi in your dialplan with something like:
1234,1,AGI(cidauth.agi)
1234,2,Hangup
Here is the Mysql table:
mysql describe cids;
++--+--+-+-++
| Field | Type | Null | Key | Default | Extra |
++--+--+-+-++
| id | int(11) | | PRI | NULL| auto_increment |
| cid| char(10) | YES | | NULL||
| active | int(11) | YES | | NULL||
++--+--+-+-++
Insert each CID into the table that you want to have access. Active = 0 for
disabled, active = 1 for enabled.
Here is the perl code:
#!/usr/bin/perl
#
use Asterisk::AGI;
use DBI;
$db = dbname;
$host = dbhost;
$port = 3306;
$userid = dbuser;
$password = dbpass;
$connectionInfo = DBI:mysql:database=$db;$host:$port;
$dbh = DBI-connect($connectionInfo,$userid,$password);
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer();
if (my $callerid = $input{'callerid'}) {
$callerid =~ m/(\d{10})/; # cut off the name
part of CID, numbers only
$callerid = $1;
$query = SELECT active FROM cids WHERE cid='$callerid';
$sth = $dbh-prepare($query);
$sth-execute();
$active = $sth-fetchrow_hashref();
if ($active-{active})
{
$AGI-set_context('ldincoming');
$AGI-set_extension('1011');
$AGI-set_priority(1);
exit;
}
}
# if we got here, there was no match found [auth failed], so play a message
saying so
# you could also log all auth failed [with caller ID ! :) ]
# you could also transfer caller to an operator
$AGI-stream_file('invalid');
$AGI-hangup();
exit;
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[Asterisk-Users] DISA and AGI: authenticate by caller ID?
I'm having trouble getting an AGI exec command to spawn app_disa. The
script executes properly, but does not spawn DISA. The CLI gives no helpful
clues. Am I doing the exec incorrectly?
I want to have a way to authenticate callers to the extension by Caller
ID... if their caller ID is in my database and set to active, they can call
out. [like a calling card but auth'd by CID instead of PIN].
Here is my dialplan:
1234, 1, agi(ldusers.agi)
1234, 2, Hangup
Here is my code:
#!/usr/bin/perl
#
use Asterisk::AGI;
use DBI;
$db = dbname;
$host = hostname;
$port = 3306;
$userid = dbuser;
$password = dpasswd;
$connectionInfo = DBI:mysql:database=$db;$host:$port;
$dbh = DBI-connect($connectionInfo,$userid,$password);
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer();
if (my $callerid = $input{'callerid'}) {
$AGI-say_digits($callerid);
$query = SELECT active FROM cids WHERE cid=$callerid;#
active should be 1 if the caller ID is found and set active
$sth = $dbh-prepare($query);
$sth-execute();
$sth-bind_columns(undef, \$active);
$sth-fetch();
if($active)
$AGI-exec('DISA','no-password|disa');
}
$AGI-hangup();
exit;
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Re: [Asterisk-Users] Hyperthreading?
I have a Linux 2.6.6 box with Hyperthreading with a Digium 4 port T1 board [TE-405P ?] Intel P4 3.2 w/ HT and the board is an Intel 875 w/ HT support. So far no issues. I did have a hard-lock six hours after first booting the box, but so far it has been up since then [uptime 5 days 16:11], and actually has higher load than when it locked [had no load then]. Call quality is perfect. I will reply again to this thread if any problems do crop up. Message: 10 Date: Tue, 08 Jun 2004 08:44:19 +0800 From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Hyperthreading? Reply-To: [EMAIL PROTECTED] Chris Bond wrote: Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. A lot of people report no problems with HT turned on, but you have to look at these reports carefully. A lot of people have no zaptel hardware in their system. That seems OK with HT on. Some people with zaptel hardware use it in very simple ways. That also seems OK. However, if you try things like setting loopback on a TE410P card with HT turned on, the machine locks solid. So, there are HT issues, but not everyone hits them. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialplan experts needed
In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set up so that 1000 and 2000 are lines in hunting on incoming extension 555. I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here is what I have now and it seems to work okay: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,VoiceMail2(u3278) exten = 555,104,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup Is this correct? What if there were a third SIP device 3000 ? Would it look like: exten = 555,1,Dial(SIP/1000,30) exten = 555,102,Dial(SIP/2000,30) exten = 555,103,Dial(SIP/3000,30) exten = 555,104,Voicemail2(u3278) exten = 555,105,Hangup exten = 555,2,VoiceMail2(u3278) exten = 555,3,Hangup That doesn't seem correct. Also, quick note, the user does not want to have a different busy and unavailable message, so that is why I have it set up to always be the unavailable message for voicemail. thanks for the help! Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip device discussion and reviews
Good evening. I just wanted to take a minute and review my experiences with some of the SIP devices out there on the market. I hope this post will help newbies or someone considering a certain device. I would appreciate any other input on either the devices I am reviewing or other devices that I didn't! These devices are deployed in our primary line and small PBX replacement service provider offering. Hard Phone: Grandstream 101 Good: Call quality, Feature support, Ease of setup, Price, Firewall support Bad: Buttons are crap, cheap looking, speakerphone Price: About $60-70 US I use this phone to do all my testing and as my personal SIP device. My biggest gripe of all is that if you attempt to speed-dial, you will never succeed. The buttons require a firm and definite press to register. I fail 1 out of 10 calls on this device simply because one of the digits was ignored. On the plus side the call quality is very good and loud if you are on the handset. The price is right, and the phone is easy to setup. It also supports STUN, and plays nicely with NAT/firewalls, and has good codec support. The mic is too close to the speaker on speakerphone, and there is noticeable echo if speakerphone is used. The speakerphone is also too quiet, even at maximum volume. Customers call it the walmart phone -- it is the cheapest looking SIP device out there. Hard Phone: Cisco 7960 Good: Great looks, speakerphone Bad: Price, Cisco stupidity Price: About $250-300 US I still use my personal 7960 phone in Skinny mode because it refuses to load the SIP load that I had to PAY for from Cisco. Cisco's directions to load the SIP load do not work and are outdated. After tinkering with a friend for half a night I finally got it to load the SIP image from the TFTP server... which it then promptly rejected with a checksum failure. I can handle buggy upgrades if I can access different firmwares for free, but to pay for a firmware that won't even load? No thanks. The speakerphone is by far the best speakerphone I have ever had on any phone, ever. I use this phone when I do training conferences and meetings that are broadcast via teleconference, and it performs like a champion. The phone has good looks and is definitely an eye catcher. I have a couple of them set up in our administrative offices as eye-candy. The later SIP versions work well. ATA: Sipura SPA-2000 Good: Configuration, Functionality, Stability Bad: Unimpressive Codec Support, Doesn't handle firewalls well Price: About $85-95 US Of all the SIP devices we have in the field, we have the most in Sipura SPA-2000s. The configuration is clean and straightforward. The Sipura definitely has the functionality that we need, and all features and functions seem to work well, and properly. The units are stable and do not need constant rebooting or maintenance.Built in echo cancelation works well. Unfortuantely, the lack of any decent low bit-rate codec is making me look hard at the Handytone 286/486 units [they support iLBC]. The code in my Sipura's does not support GSM or iLBC. They also do not handle firewalls well at all. NAT support seems to work okay with Register's set frequently, but with no Stun support [they may have added Stun support in later firmware releases], real firewalls can interfere with the device receiving incoming calls. Soft Phone: SJ Phone by SJ Labs Good: Interface Bad: Configuration, No Echo Cancelation Price: Free to Try I've installed and quickly uninstalled many soft phones, but the SJ Phone stays on my test machine. The interface is clean and works well. Unfortuantely, the SJ phone is not very straightforward to configure the service provider information with its nested profile setup. There is also no echo cancelation code. I can't blame the SJ Phone for the echo, it's caused by a cheap sound card interface in my laptop, but it would sure be nice to find a soft phone that would help me attempt to cancel it out. The sound controls on the SJ phone are also poor, which exacerbates the echo problem. Interface Card: Digium TE-405P Good: Price, Intel-based, Features Bad: ??? Price: $1500 Not really a SIP device, but I am very pleased with the TE-405P's that we have been buying from Digium. We interface them with our PRI and CT1 circuits and they are working without any problems. The price is right, they work with Linux, and they support every form of T1/E1 out there. What more could one ask for? Coming soon: Review of Cisco ATA-186 [if I can ever get it to work correctly] and the Grandstream Handytone 486 [have some on the way]. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] miserable time with Cisco ATA186
If I turn allow=ulaw on only, asterisk tries to use it a=rtpmap:0 PCMU/8000 but the ATA says it doesn't have it: Answering/Requesting with root capability 4 Answering with non-codec capability 0x1(G723) If I turn allow=alaw on only or with allow=ulaw, asterisk sends it, the ATA says it has it [alaw], but it still won't negotiate it. I think the stupid ATA is just determined to use G723 no matter what... I have LBRCodec set to 3 which should have it try to use G729, but it still tries to use G723. The AudioMode setting has a parameter bit to Enable G711 only, but I'm not sure how that bit thing works. Either the default 0x00150015 or the recommended 0x00140014 fails. [btw, bit 1 should be 1 to enable G711 only, if someone can help me there]. I'm seriously about to punt this thing into the garbage. Help! thanks, matt From: Timothy R. McKee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] miserable time with Cisco ATA186 Date: Fri, 4 Jun 2004 00:04:22 -0400 Reply-To: [EMAIL PROTECTED] Noticed that he has ALAW set as the preferred codec on the ATA. I'd suggest testing with allow of ulaw only, then try turning on other codecs. We know that one works well. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, June 03, 2004 23:36 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186 Perhaps, but *I* at least had decent luck with 2.16.1. I suspect he has allow=all and the codec that ends up being used is G723.1 and then, of course, everything goes to hell. On Thu, 2004-06-03 at 22:59, brian k. west wrote: because 2.16.1 has some bugs.. you need 2.16.2 or higher. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] miserable time with Cisco ATA 186
think I figured out the binary bit thing, so I am posting to list to hopefully help someone else out bits 15-8 are all 0 and are reserved bit 7:value 0:numeric 8 reserved bit 6:value 0:numeric 4 reserved bit 5:value 0:numeric 2 dtmfmethod bit 4:value 1:numeric 1 dtmfmethod bit 3:value 0:numeric 8 [CNG fax tone] bit 2:value 0:numeric 4 [CED fax tone] bit 1:value 1:numeric 2 [g711 codec only: 1 true 0 false] bit 0:value 0:numeric 1 [silence suppression 1 on 0 off] so that makes 0012 for each line, or 0x00120012 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] miserable time with Cisco ATA186
I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses right next to each other on the same subnet. SIP Debug shows [munged being the IP address]: Answering/Requesting with root capability 4 Answering with preferred capability 0x8(ALAW) Answering with capability 0x1(G723) Answering with capability 0x2(GSM) Answering with capability 0x10(G726) Answering with capability 0x20(ADPCM) Answering with capability 0x40(SLINR) Answering with capability 0x80(LPC10) Answering with capability 0x100(G729A) Answering with capability 0x200(SPEEX) Answering with capability 0x400(ILBC) Answering with capability 0x800(UNKN) Answering with capability 0x1000(UNKN) Answering with capability 0x2000(UNKN) Answering with capability 0x4000(UNKN) Answering with capability 0x8000(UNKN) Answering with non-codec capability 0x1(G723) 12 headers, 20 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP munged:0;branch=z9hG4bK304da88f From: munged To: munged Contact: munged Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Jun 2004 02:26:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 461 v=0 o=root 284 284 IN IP4 munged s=session c=IN IP4 munged t=0 0 m=audio 14466 RTP/AVP 0 8 4 3 2 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - This Retransmits several times and then the call is scheduled for destruction. The CANCEL sip messages seem to fail also, as they are retransmitted many times. I'm running the ATA conf from: http://www.fnords.org/~eric/asterisk/ata-186.shtml Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] free sip termination
help me test load a box! I have a new box with four PRIs on a TE405P I will terminate US Toll-free traffic (1-800, 888, 877, 866) for free via SIP to anyone who wants to test. Just email me at [EMAIL PROTECTED] if you would, to let me know that you're testing, and with any comments about quality, etc. I have ulaw, alaw, and GSM codecs enabled. To use, just send your call via SIP to 67.153.209.214 with the username of free secret free yours, Matthew Simpson TxLink Communications 972-617-2877 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki TOS - worrying for an open source project?
From: Steven Critchfield [EMAIL PROTECTED] The other part is that a wiki is really unmirrorable using normal methods of mirroring a site. You need to just run the same software and have the database behind it mirrored. I'm sure if the wiki is running a new enough version of mysql, and the admin is willing, you could set up a mirror of the database and then set up a full on replication. Mysql supposedly supports replication, might want to put it to some use. -- I don't know who is hosting the Wiki right now, but we are willing to either host the Wiki as a mirror, or be a mysql replication mirror. We are using mysql replication right now to replicate amongst three servers for our RADIUS and other hosted apps and it works very well. We also do daily backups of the master mysql server to an offsite location. We would do this free of charge, of course. We are using asterisk as a media gateway with Digiums TE405P cards and we appreciate the work that is going into Asterisk. Contact [EMAIL PROTECTED] or 972-617-2877 yours, Matthew Simpson TxLink Communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] strange problem with SIP/voicemail
I'm having a very strange problem I've been fighting with all day. It's 2:30am, and I'm stuck. I think the problem may lie with one of my SIP providers, but I'm not sure. I have two ways to call into my test Grandstream. I can call a PSTN 360 area code number that will forward to my FWD number, which in turn is registered with my * box on extension 2030. If I call the 360 number, everything works, my Grandstream rings, and if I don't answer, it goes to voicemail and voicemail works. I also have a PSTN 972 area code number that forwards directly to my * box. If I call the 972 number, my Grandstream will ring, but if I don't answer, it will give me silence for a bit, then I hear a click, my CLI interface says that it is recording a message, but then it says: Apr 19 02:21:20 WARNING[15373]: app_voicemail.c:1261 play_and_record: No audio available on SIP/66.147.170.34-0811abe8?? Here is my exten map [actual phone number munged]. I have removed the Grandstream from the exten for this example. It makes no difference whether the Grandstream gets rang or not: exten = 9725551212,1,Answer exten = 9725551212,2,Voicemail2(u1000) exten = 9725551212,3,Hangup Also, just for testing, I have added this extension: exten = 2501,1,Voicemail2(u1000) exten = 2501,2,Hangup If I dial 2501 from my grandstream, voicemail works that way, too. My questions: 1) Should I have the Answer in there or not? It doesn't help to add or remove it. On the FWD number, I do not have an Answer. 2) I can get voicemail to work on the incoming 972 number if I change the dialplan around and then do a restart gracefully. Example: exten = 9725551212,1,Answer exten = 9725551212,2,Playback(transfer) exten = 9725551212,3,Voicemail2(u1000) exten = 9725551212,4,Hangup It will work once, maybe twice, and then it won't work any more after that until I fiddle with the dialplan again and do another restart. On Saturday when I thought I had all of this working, I dialed in at least ten times and had no problems. I originally was running a CVS from 03-14-04 now I am running 04-19-04, and still have the same issue. Anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] what is the best codec for low bandwidth? for quality?
The number of codecs is overwhelming to me. What do ya'll consider the best codec for conserving bandwidth? [I realize at the cost of quality] Secondly, what do you think the best codec for voice quality is? Yours, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
