[Asterisk-Users] Re: RE: virtual extension per user ?
You can do this with agents, no need for a queue. Define agents in agents.conf In your dialplan, instead of Dial(SIP/bedroom) use Dial(Agent/200) Let the phones login as agent :) OK, I know I have to Dial(Agent/200), but how will I login agents if I don't use queue? If phone log's in as agent, then I didn't do anything, because that agent will always be on that phone (and that is something I would like to avoid - because of that I started to use agents in first place). Maybe I didn't understand something right. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: RE: virtual extension per user ?
AMP doesn't do miracles! Look at its dialplan. I believe he doesn't, but I don't have AMP installed. Next week I think I'll have enough free time to try it. Will [EMAIL PROTECTED] do the trick? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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 add stun functionality in asterisk
Hi friends ! I want to add stun functionality in asterisk. can anybody give me some hint that how can i start that. thanks in advance Deepak Dhiman ___ --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: RE: virtual extension per user ?
On 09:02, Fri 17 Feb 06, Tomislav Par?ina wrote: You can do this with agents, no need for a queue. Define agents in agents.conf In your dialplan, instead of Dial(SIP/bedroom) use Dial(Agent/200) Let the phones login as agent :) OK, I know I have to Dial(Agent/200), but how will I login agents if I don't use queue? If phone log's in as agent, then I didn't do anything, because that agent will always be on that phone (and that is something I would like to avoid - because of that I started to use agents in first place). You have to use AgentCallbackLogin for that. If a phone logs in that way, it's reachable as Agent/200 You can also use AgentCallbackLogin to logout the agent. You don't have to worry about an agent that forgets to logout on phone X when they walk to phone Y, cause AgentCallbackLoging will overwrite asterisk database entry for that agent so it's only reachable on the phone where they last login (asuming they didn't logout there) Maybe I didn't understand something right. When I get home later today I will put an example in my system and post it here. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] RE: What ATA should I buy?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... We have got some ATA for only $55 if you are interested? Sam Yes Sam, I'm interested. If they work with FAX I'll definitely buy one of them for testing. -- Tomislav [EMAIL PROTECTED] ___ --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] problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)
On 2/15/06, Michael Collins [EMAIL PROTECTED] wrote: Nik, Looks like you're making some progress. When I first started using [EMAIL PROTECTED] I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten = 555,1,Dial(Zap/1/5595551212) I chose a specific Zap channel and the exact digits that I wanted to send to the telephone company. This helped me figure out what to dial. The other thing you can do is log on to the CLI and turn on PRI debugging: pri debug span 1 This will cause PRI debug messages to display on the console. It might take a while but you will learn to read those debug messages. You can also post them to the list and we'll help you to interpret them. -MC ok, thanks for your support, now i've enabled debug on span 1, and i've make a new entry in extension.conf: exten = 444,1,Dial(Zap/0/mynumber) when i call 444 i get in the logs: Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Setting NAT on RTP to 0 Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Checking SIP call limits for device 102 Feb 17 03:50:59 DEBUG[3607] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Feb 17 03:50:59 VERBOSE[4262] logger.c: -- Executing Dial(SIP/102-2079, Zap/0/mynumber) in new stack Feb 17 03:50:59 NOTICE[4262] app_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown) Feb 17 03:50:59 VERBOSE[4262] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Feb 17 03:50:59 DEBUG[4262] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. it seems that the only information it gives mi is: app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. so it seems that i don't have channel for outgoing calls? how can i check it? maybe there is another logfile more detailed? thanks a lot for your help... ___ --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] one way / irratic voice over iax and g729
Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP) Asterisk1 -(IAX) Asterisk2 -(SIP) Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there is one way voice whereby the B party cannot hear the A party, however the A party can hear the B party fine. Sometimes there is no audio for the B party, other times the B party can hear the A party but it is very broken up and stuttery, with only parts of the words coming through. The calls also work fine when using g711 from the A party. Asterisk2 is running a couple of TDM04B's so there is a physical timing device on that side and Asterisk1 is running ztdummy on a 2.6 kernel - so there is timing on that side also (??) Have done a fair bit of searching on this one, and as it only happens with g729 (both systems have the licensed codecs installed) it is a bit of a head scratcher - has anyone else experiencved this? Or does anyone have any feedback? Cheers, Ben ___ --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] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi Chuck, my solution may be considered a bit strange but I chose it after trying asterisk code without success, trying to use Tzafrir patch but I had to change asterisk user umask too The right solution could be something like a voicemail_dir_permissions parameter in voicemail.conf so anyone could change permissions without modifying asterisk code. The externnotify parameter solution I used was the faster and less invasive. If you want to make a script to install asterisk, it is better to copy voicemail.conf and a script file than patching. Giorgio Incantalupo Chuck Bunn wrote: Hi, Could you post the updated patch for 1.2.4 Thanks Ben Klang wrote: On Thursday 16 February 2006 11:47, you wrote: Just so I am clear this patch will work with 1.2.4 and requires manual updating to files and then a recomplie of Asterisk source correct?? This patch was written against trunk a couple weeks ago. Last night I applied it to 1.2.4 and there were only two small conflicts (easily resolved). Recompile and install Asterisk. You may need to manually poke existing files to get the perms the way you like but all new files should be created correctly. If you're having trouble getting it to apply to 1.2.4 let me know and I'll send you my rebuild patch. If you happen to be a SuSE user I've got Asterisk 1.2.4 RPMs built for SuSE 10.0. /BAK/ ___ --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 -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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
[Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Hi List Is someone out there using one or more GSMgateway(s) from CyberTelecom ? Me and some friends are interested in buying some of them, but before we would like to ask, how the experiences are others have made. e.g. How easy to setup ? How reliable ? How's the voice quality ? etc. Any input/feedback is welcome. Greets Adibar Hi Adibar, I have one since a couple of weeks. It works for me. Basically you just plug it into an analog interface after installing the GSM chip. The voice quality is good even in my office; a sort of radio waves-black hole. Normally most cellphones just disappear when they are there.. The only problem I have so far is that the TDM400 FXO module does not seem to read the caller id. A regular phone shows it, if I switch connections. It might be a problem of configuration of the TDM card; I have looked in the wiki and googled around, but I do not know how I can change the way a zaptel card reads the callerid. I will try to upgrade to 1.2.x asap to see if this helps. Best regards Aldo ___ --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: What ATA should I buy?
Since you have no Digium hardware (and thus no connection to POTS or PRI)... are you routing your phone calls via VoIP? If so, it is not recommended to run FAX via VoIP. The two don't mix. FAX is not able to handle packet loss like VoIP. Also, any codec other than uLaw will not even come close to working, as the codecs are designed to compress voice. Hi Ron! Thank you for your mail. I know there could be some issues, but if I use ulaw, most of FAX should pass true. In few years people won't send faxes anymore, but till then I need something that will work with 90% success. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: What ATA should I buy?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... AFIK, fax is supported and installed with with app_txfax app_rxfax If this proves to be true why would you need the ATA? I'm working on this one. I have to install app_rxfax but I have failed. Soon, I'll try again (hopefully next week). Anyway, I'll need ATA even then. Because it isn't just receiving FAX, but sending it. It is problem to scan paper then send it by mail or app_txfax. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] aastra v1.3.1 firmware
Hi there, Is it possible with the new aastra firmware to have distinctive ring support? (the wiki says: There doesn't seem to be any way to have the server request a distinctive ring.) The rest of the features make this sound like a good phone. (price/quality) cheers ___ --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: Re: asterisk logger - urgent!!!
Why don't you simply rotate the logs with logrotate ? How to do that? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] sipdiscount
2006/2/8, adibar [EMAIL PROTECTED]: Hi That does the job (dialout only) I'm trying with this configuration but I receive the same result. Checking with ethereal I see the answer from sipdiscount: asterisk-sipdiscount Request: INVITE sip:the number@sip1.sipdiscount.com sipdiscount-asterisk Status: 401 Unauthorized It seems the fromdomain option is not being used... I'm using Asterisk 1.2.4 and my sip.conf (really sip_additional.conf has this). [sipdiscount] username=test type=peer secret=test qualify=yes nat=yes host=sip1.sipdiscount.com fromuser=test fromdomain=stun.sipdiscount.com dtmfmode=inband disallow=all canredirect=no allow=gsm allow=ulaw allow=alaw -- Alejandro Vargas ___ --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] one way / irratic voice over iax and g729
2006/2/17, Ben Dinnerville [EMAIL PROTECTED]: Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP) Asterisk1 -(IAX) Asterisk2 -(SIP) Carrier if you are calling asterisk-to-asterisk, you should try speex compression. In my tests, speex had the better quality even with low bandwidth or bandwidth very occuped by other applications (p2p). -- Alejandro Vargas ___ --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: one way / irratic voice over iax and g729
The carrier does not support speex, only g729, 723 and 711, so to minimise codec coversions etc, and due to the fact that licensing 723 is so expensive and 711 is a bit fat on bandwidth (asterisk 1 is connecintg over 128k ISDN) we are kind of stuck with g729 (not that it has ever proved to be a problem anywhere else) Alejandro Vargas wrote: if you are calling asterisk-to-asterisk, you should try speex compression. In my tests, speex had the better quality even with low bandwidth or bandwidth very occuped by other applications (p2p). -- Alejandro Vargas ___ --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: asterisk t.38 pass
yes, with last patch works well. thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adolfo R. Brandes Sent: Thursday, February 16, 2006 10:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: asterisk t.38 pass turby wrote: is there recomended source files for t.38 pass? latest cvs does not work for me. is it possible publish working src? You mean T.38 passthrough? I've just uploaded an asterisk-1.2.4 backport of the lastest svn asterisk/trunk T.38 code to the bugtracker, and it works swell for me. Go here: http://bugs.digium.com/view.php?id=5090 Adolfo ___ --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] sipdiscount
Hi I hope you dont try to dial with the user test and the password being also test, that would definitly end up in an Unauthorized ;-) Don't forget the register-stuff inside sip.conf, e.g.: register = YOURLOGIN:[EMAIL PROTECTED] otherwise it does not work for me either ;-) If you do a sip show registry you should see something like this: sip1.sipdiscount.com:5060 YOURLOGIN 3585 Registered ...and if you do a sip show peers something like this: sipdiscount/YOURLOGIN194.120.0.201N 5060 OK (35 ms) Hope that helps. Greets Adibar On Fri, Feb 17, 2006 at 10:45:57AM +0100, Alejandro Vargas wrote: 2006/2/8, adibar [EMAIL PROTECTED]: Hi That does the job (dialout only) I'm trying with this configuration but I receive the same result. Checking with ethereal I see the answer from sipdiscount: asterisk-sipdiscount Request: INVITE sip:the number@sip1.sipdiscount.com sipdiscount-asterisk Status: 401 Unauthorized It seems the fromdomain option is not being used... I'm using Asterisk 1.2.4 and my sip.conf (really sip_additional.conf has this). [sipdiscount] username=test type=peer secret=test qualify=yes nat=yes host=sip1.sipdiscount.com fromuser=test fromdomain=stun.sipdiscount.com dtmfmode=inband disallow=all canredirect=no allow=gsm allow=ulaw allow=alaw -- Alejandro Vargas ___ --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 -- This message has been scanned and dangerous content has been removed by our MailScanner. This includes all executables. If the transfer of executables is desired please consider to send them as a zip-file, which is allowed to pass the checks, but which will be scanned for viruses. Please be sure to keep your local Antivirus up-to-date, as this message is no guarantee that all viruses have been removed. -- ___ --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
FW: [Asterisk-Users] AGI onAnswer function: does it exist?
Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? Best regards, Vlasis Hatzistavrou. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlasis Hatzistavrou Sent: Thursday, February 16, 2006 3:43 PM To: asterisk-users@lists.digium.com Cc: 'Vlasis Hatzistavrou' Subject: [Asterisk-Users] AGI onAnswer function: does it exist? Hello, I am trying to write an AGI in Perl and I need to execute a function upon answer of a call. I know that there is the possibility to use the M() option in the Dial command in order to do what I need, but that would mean that I would have to incorporate the function's work in a different AGI program, and I need to avoid this. So, I would like to know if such an option is available in AGI like an onanswer() function or something equivalent that I can use. Any help would be really appreciated, as I've been searching www.voip-info.org and the Asterisk mailing lists for days now, without any success. Best regards, Vlasis Hatzistavrou. ___ --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] FXO port on TDM400P hangs!!
The UPD I'm using for the * is actually an UPS I used for a much biger Windows machne, complete with monitor etc. The coleague who used the UPS was aut of the office when I installed the system and took hid UPS :-) I'm sure the UPS is good. I'm saying I'll change the PSU because I've had problems with the PSU in an other big machine into our office. The machine was randomly rebooting. I changed the PSU to a thermaltake Active PFC (hope I remamber the name corectly) PSU and that cured it all. I decided to use an Active PFC PSU for that machine because the power line conditions here are owfull, we never get power at nominal capacity and the power is allways oscilating. The Active PFC PSU is supposed to be better in such conditions. Beware anyone in the rural area of Romania! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jared Armstrong Sent: Thursday, February 16, 2006 9:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FXO port on TDM400P hangs!! In a dedicated fax server with brooktrout fax cards (analogue), and when I first setup my * without a UPS. We were noticing that the lines became un-initialized which required the fax/phone software/drivers to require re-initialization. On our windows based fax server this required restarting the fax service and on * it required doing a zaptel/asterisk restart. Since we moved both of these systems to new/larger UPS's the issue appears to have disappeared. This is only a suggestion since it appears to me that there might be a correlation. I can't say if a larger PSU would help, but I don't see how it could hurt. Jared Armstrong -Original Message- From: Cosmin Prund [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 10:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FXO port on TDM400P hangs!! POWER FLUCTUATIONS I have in abundance! My * is on a modest machine (Duron 3000+, 512RAM, a good Gigabyte MB and a cheap PSU). I've got a TDM400P card with one FXS and three FXO. The UPS is as good as I'm willing to put into the box. If power fluctuations are known to cause such problems I'll have to upgrade the PSU to something good. Anyone else had such problems because of power fluctuations? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jared Armstrong Sent: Thursday, February 16, 2006 4:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FXO port on TDM400P hangs!! If this is anything like the issue I have seen on Brooktrout fax cards it is related to power fluctuations. Is your * system on a properly sized UPS for the system? What card do you have installed and what motherboard/PSU are you using? Jared Armstrong -Original Message- From: Cosmin Prund [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 5:36 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FXO port on TDM400P hangs!! Hello everyone. This is a message I've sent before on Sunday, no one replied so I'm reposting it (guess not everyone's at work 7/7) I've got this really annoying and beyond-my-knowledge-to-debug problem. The line connected to my FXO port gets marked out of order by my telco operator. I don't know how to explain this further. If I dial my own number from a different phone I get a message the called number is out of order. This is only rarely happening (twice on Sunday, once today) but when it does happen the * requires a reboot! The worst part is that we usually find of this problem from a customer calling our other number or a mobile saying the main number can't be reached! If anyone has any idea where to look or what to look for in the log files, please advice. If anyone has any workaround for this problem, again, please advice. At the moment I'm working on a really ugly solution: I'm planning to create a call file once every 5 minutes and have * call the hanging number from the other number. If the call makes it back to the * I'll set a global var. If the call doesn't make it back to * the global var will not get set and, when the Dial command times out, I'll know it's time to System(/sbin/reboot)! Unfortunately this is really ugly and I'm not sure I'll be able to make it work, but I will try! ___ --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:
Re: [Asterisk-Users] Re: Re: asterisk logger - urgent!!!
check out the man page for logrotate The logrotate script is usually started daily by the cron daemon ( see /etc/cron.daily/logrotate on redhat boxes) hth, cristi On 2/17/06, Tomislav Parčina [EMAIL PROTECTED] wrote: Why don't you simply rotate the logs with logrotate ? How to do that? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] sipdiscount
On 08/02/06, Alejandro Vargas [EMAIL PROTECTED] wrote:
Sipdiscount has replaced their asterisk servers for another thing.
Then, no more iax. Ok, but I can't make calls using sip also... I'm
getting a forbidden error when using sip1.sipdiscount.com. Anybody
got it working?
A pretty simple setup works for me:
sip.conf:
[sipdiscount]
type=peer
host=sip1.sipdiscount.com
username=xxx
secret=yyy
canreinvite=no
dtmfmode=info
extensions.conf:
[sipdiscount-out]
exten = _6.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _6.,2,Hangup
(I use a prefix of '6' to reach sipdiscount)
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
___
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RE: [Asterisk-Users] RE: RE: virtual extension per user ?
AMP doesn't do miracles! Look at its dialplan. I believe he doesn't, but I don't have AMP installed. Next week I think I'll have enough free time to try it. Will [EMAIL PROTECTED] do the trick? Yes, I was referring to AAH Mimmus ___ --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] Bridge Calls with G()
On 02/16/06 04:45 Prakash Rao Kanthi said the following: This works but the calling party hears 'prompt02' and the called party hears 'prompt04' the two parties are NOT connected foa conversatoin - just like the wiki describes Does anyone know when the 'G()' flag will be fixed or any potential work-arounds? i'm not really sure what the original rationale was in transferring the called party to priority+1 and the calling party to priority, but i've opened an issue on this and provided a small patch which makes it act the way it's described. it's available at http://bugs.digium.com/view.php?id=6523 -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Anyway to pass CIC in sip header
On 02/17/06 08:51 BJ Weschke said the following: On 2/15/06, Kevin Hanson [EMAIL PROTECTED] wrote: I am using an Asterisk box as a mini-softswitch and have run into a minor (hopefully) road block. The far end switch requires CIC (Carrier Identification Code) in the SIP invite like: INVITE sip:+18001234567;[EMAIL PROTECTED];user=phone SIP/2.0 ^^^ Is there a way to configure Asterisk to send this in the SIP invite? Any help would be *greatly* appreciated. Not w/o a code change, but if you're game to do that, you certainly could do it. just curious, but is the cic tag added to the INVITE method a valid recommendation under the SIP RFCs ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] zoom FXS/FXO gateways
On Thu, 16 Feb 2006, Martin Joseph wrote: I have ordered the wellgate 3701A to see if that helps me any... It's about twice the price of the SPA3000 ($199), but I know my 2 wire loop is 15000ft+ so I figured the SPA3000 isn't going to help me. I'll be very interested in your review of this device. Where did you buy it from? -Dan ___ --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 t.38 pass
turby wrote: yes, with last patch works well. thanks. Glad to be of service! Adolfo ___ --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] Aasterisk large-scale deployment w/analog phones
Didnt think of that. Thanks for the insight. Dovid --- Matt Florell [EMAIL PROTECTED] wrote: Hello, We had over 100 ATA adapters in production 3 years ago. Now we have less than 20. They use more power overall than Channelbanks, they are not designed to be used 12-16 hours a day every day. You must configure and test every one. They do not last as long as channelbanks. Using Channelbanks has saved us time and cost in fixing/replacing equipment, We have not had to replace a single one of the ten channelbanks we've had in place for the last 18 months. In the first year of ATA usage, we had to replace 20% of our ATAs(overheating, random dying, coffee, soda, other abuse). As for initial purchase cost, ATA adapters are actually slightly cheaper per port than channelbanks(if buying new). MATT--- On 2/15/06, Dovid Bender [EMAIL PROTECTED] wrote: I may be missing something here but why wouldnt ATA's work ? (other than cost). --- maka [EMAIL PROTECTED] wrote: hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar setuyp/planning - Do you think is it better to distribute resources among multiple (more than two), lower-port-density asterisk servers? Or is it better to use a channelbank for that purpose? Cheers ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: FW: [Asterisk-Users] AGI onAnswer function: does it exist?
On 12:17, Fri 17 Feb 06, Vlasis Hatzistavrou wrote: Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? It's a -user question to begin with. Have your agi connect to the manager interface and get the answer info from there :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] IE Display in SETUP (pri_cpe)
Hi, we send an SETUP message to an SIEMENS (German) Provider. Our Equipment is pri_cpe, so we may NOT send an IE Display to the Carrier. Call Ref: len= 2 (reference 15072/0x3AE0) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 9b] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 27 ] [28 0e 53 74 65 66 61 6e 20 53 63 68 6d 69 64 6c] Display (len=14) [EMAIL PROTECTED]@ ?º'[EMAIL PROTECTED]@ÈF@[ Test Test ] [6c 0c 21 81 39 31 32 32 31 37 31 33 33 36] Is there a way to eliminate Display in SETUP Message? zapata.conf: language=nl pridialplan = international prilocaldialplan = national switchtype = euroisdn signalling = pri_cpe group = 1 context = default overlapdial=yes channel = 1-15,17-31,32-46,48-62,63-77,79-93 Asterisk: 1.2.4 Zaptel: 1.2.3 Libpri: 1.2.2 Best Regards Markus ___ --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] sipdiscount
2006/2/17, adibar [EMAIL PROTECTED]: Hi I hope you dont try to dial with the user test and the password being also test, that would definitly end up in an Unauthorized ;-) But... the page of sipdiscount says you can use the user test with password test to do free one minute calls for testing purposes. Don't forget the register-stuff inside sip.conf, e.g.: register = YOURLOGIN:[EMAIL PROTECTED] otherwise it does not work for me either ;-) sip show registry HostUsername Refresh State sip show peers sipdiscount/test 80.239.235.200 N 5060 OK (60 ms) -- Alejandro Vargas ___ --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] sipdiscount
2006/2/17, Peter Bowyer [EMAIL PROTECTED]: A pretty simple setup works for me: The problem may be the username/password. But the page says this: SIP Discount offers the possibility to test our service right away, for free! No need to sign up: just enter the account details below in your favorite softphone or ATA and start calling! You can call all destinations marked with * in our rate list . (Trial calls are limited to a maximum duration of 1 minute). To enjoy unlimited calls, simply sign up for SIP Discount. User Name: test Password: test Domain/Realm: sipdiscount.com SIP Proxy/registrar:sip1.sipdiscount.com SIP Outbound Proxy (optional): sip1.sipdiscount.com STUN server (optional): stun.sipdiscount.com The only problem I say is asterisk is not sending stun.sipdiscount.com or sipdiscount.com as domain. It is sending sip1.sipdiscount.com. -- Alejandro Vargas ___ --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] SOLVED - Channel bank woes - no outbound calls
On 02/17/06 10:13 James Texter said the following: static int vpmdtmfsupport = 1; Change this to static int vpmdtmfsupport = 0; i'm guessing that this would only be relevant if you were using the newer TE4XXP cards with the VPM boards attached. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] free tollfree termination
http://www.trxtel.com/index.php?page=Tollfree_Termination This is a free service, I am not selling anything with this service. I just thought that individuals that read this list may enjoy getting tollfree access free this way (yet another way) given that it lets you send your caller id and some of the other free providers dont let you do that. Starting a test service now, for individuals free north american tollfree termination. For carriers that do large quantities of minutes (a not really defined term as yet, more a negotiated value) we will share revenue with you for sending calls to us. If you set up IP PBX systems for customers, add a route in and make residuals off those customers. Run a ITSP? Get revenue for each minute that a customer dials a north american toll free. If anyone has any problems using the service I would appreciate hearing about it, the service will remain free even after the test period, however to get compensation requires an account so that it can be uniquely tracked. Granted tollfree traffic isnt usually the bulk of a provider, but at least now you can provide it free to your customers without losing on costs like bandwidth :) -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] codec negotiation with SPA-3K
I'm having trouble with Asterisk-1.2.4 negotiating codecs with a Sipura 3000 which is running the latest v3 firmware. The SPA-3K seems to use the preferred codec only and doesn't negotiate? The SPA is set to no in use only preferred codec. Does anyone know if Sipura will support gsm at some point? I this a bug with the SPA or codec negotiation stuff? Thanks Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
Re: [Asterisk-Users] Aasterisk large-scale deployment w/analog phones
Matt, Whose channel banks did you end up going with? Hello, We had over 100 ATA adapters in production 3 years ago. Now we have less than 20. They use more power overall than Channelbanks, they are not designed to be used 12-16 hours a day every day. You must configure and test every one. They do not last as long as channelbanks. Using Channelbanks has saved us time and cost in fixing/replacing equipment, We have not had to replace a single one of the ten channelbanks we've had in place for the last 18 months. In the first year of ATA usage, we had to replace 20% of our ATAs(overheating, random dying, coffee, soda, other abuse). As for initial purchase cost, ATA adapters are actually slightly cheaper per port than channelbanks(if buying new). MATT--- On 2/15/06, Dovid Bender [EMAIL PROTECTED] wrote: I may be missing something here but why wouldnt ATA's work ? (other than cost). --- maka [EMAIL PROTECTED] wrote: hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar setuyp/planning - Do you think is it better to distribute resources among multiple (more than two), lower-port-density asterisk servers? Or is it better to use a channelbank for that purpose? Cheers ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ---End of Original Message- ___ --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] Aasterisk large-scale deployment w/analog phones
We ended up going with Zhone channelbanks because they are available very cheaply and are very small. Just make sure you get the B version, they are much easier to program than the A version. I must note that Zhone channelbanks are not made anymore so you must buy them on the secondary market. Other than that, I have talked with people using rhino, adit and carrier access with asterisk with no issues. MATT--- On 2/17/06, Rich Adamson [EMAIL PROTECTED] wrote: Matt, Whose channel banks did you end up going with? Hello, We had over 100 ATA adapters in production 3 years ago. Now we have less than 20. They use more power overall than Channelbanks, they are not designed to be used 12-16 hours a day every day. You must configure and test every one. They do not last as long as channelbanks. Using Channelbanks has saved us time and cost in fixing/replacing equipment, We have not had to replace a single one of the ten channelbanks we've had in place for the last 18 months. In the first year of ATA usage, we had to replace 20% of our ATAs(overheating, random dying, coffee, soda, other abuse). As for initial purchase cost, ATA adapters are actually slightly cheaper per port than channelbanks(if buying new). MATT--- On 2/15/06, Dovid Bender [EMAIL PROTECTED] wrote: I may be missing something here but why wouldnt ATA's work ? (other than cost). --- maka [EMAIL PROTECTED] wrote: hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar setuyp/planning - Do you think is it better to distribute resources among multiple (more than two), lower-port-density asterisk servers? Or is it better to use a channelbank for that purpose? Cheers ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ---End of Original Message- ___ --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] sipdiscount
Hi I would sugest, that you just register without balancing your account. Than use the supplied username/password and it will work. I doubt that the test/test works. Greets Adibar On Fri, Feb 17, 2006 at 12:31:17PM +0100, Alejandro Vargas wrote: 2006/2/17, Peter Bowyer [EMAIL PROTECTED]: A pretty simple setup works for me: The problem may be the username/password. But the page says this: SIP Discount offers the possibility to test our service right away, for free! No need to sign up: just enter the account details below in your favorite softphone or ATA and start calling! You can call all destinations marked with * in our rate list . (Trial calls are limited to a maximum duration of 1 minute). To enjoy unlimited calls, simply sign up for SIP Discount. User Name: test Password: test Domain/Realm: sipdiscount.com SIP Proxy/registrar: sip1.sipdiscount.com SIP Outbound Proxy (optional):sip1.sipdiscount.com STUN server (optional): stun.sipdiscount.com The only problem I say is asterisk is not sending stun.sipdiscount.com or sipdiscount.com as domain. It is sending sip1.sipdiscount.com. -- Alejandro Vargas ___ --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 -- This message has been scanned and dangerous content has been removed by our MailScanner. This includes all executables. If the transfer of executables is desired please consider to send them as a zip-file, which is allowed to pass the checks, but which will be scanned for viruses. Please be sure to keep your local Antivirus up-to-date, as this message is no guarantee that all viruses have been removed. -- ___ --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: aastra v1.3.1 firmware
No, distinctive ring isn't supported in 1.3.1. You only have the option of setting the ring-tone on a per-line basis. Gareth stoffel wrote: Hi there, Is it possible with the new aastra firmware to have distinctive ring support? (the wiki says: There doesn't seem to be any way to have the server request a distinctive ring.) The rest of the features make this sound like a good phone. (price/quality) cheers ___ --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] sipdiscount
2006/2/17, adibar [EMAIL PROTECTED]: I would sugest, that you just register without balancing your account. Than use the supplied username/password and it will work. I doubt that the test/test works. Thanks. This worked. I already had a sipdiscoutn account without credit, but It never worked before (always needed to use test). -- Alejandro Vargas ___ --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] [OT] List messages and end user outages
Sorry, this is off topic to asterisk itself, but is about the list server. I had a power failure lastnight at home, where my email server resides, and my network was down for about 20 minutes, that was after 45 minutes of uptime on UPS. Since power was restored, around 9:45 PM EST on 2/16, I have not received a single post from the users, biz, or dev lists. Normally when this has happened in the past, it has taken 24 hours for the list server to start sending to my email server again. My question is why so long? I am on other lists and it might take an hour or so for the messages to start showing up, but why 24 hours for a 20 minute loss of contact with my email server? Robert P.S. - If there is somewhere else this question should be directed, that would be constructive, please feel free to let me know. ___ --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] sipdiscount
On Fri, 2006-02-17 at 14:32 +0100, Alejandro Vargas wrote: 2006/2/17, adibar [EMAIL PROTECTED]: I would sugest, that you just register without balancing your account. Than use the supplied username/password and it will work. I doubt that the test/test works. Thanks. This worked. I already had a sipdiscoutn account without credit, but It never worked before (always needed to use test). they may have recently disabled the test account given that if everyone is using it abuse would be high. While a free account does little to stop abuse, it does add a very small hurdle to it, which can slow people down and potentially add for slightly better tracking of problem users. -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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: Firmware version 1.3.1 released for Aastra IP phones
The follow should work from the configuration files (aasta.cfg/MAC.cfg), although I haven't tried it... audio mode: mode Where mode is a number between 0 and 3 0 = speaker 1 = headset 2 = speaker/headset 3 = headset/speaker Gareth Lee Archer wrote: Any chance of getting a config option in that allows you set headset/speaker in the audio menu? Lee ___ --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: Bridge Calls with G()
In article [EMAIL PROTECTED], Dinesh Nair [EMAIL PROTECTED] wrote: On 02/16/06 04:45 Prakash Rao Kanthi said the following: This works but the calling party hears 'prompt02' and the called party hears 'prompt04' the two parties are NOT connected foa conversatoin - just like the wiki describes Does anyone know when the 'G()' flag will be fixed or any potential work-arounds? i'm not really sure what the original rationale was in transferring the called party to priority+1 and the calling party to priority, but i've opened an issue on this and provided a small patch which makes it act the way it's described. it's available at http://bugs.digium.com/view.php?id=6523 I think it is more useful to transfer to the two separate priorities, but the documentation should reflect that. If you want to distinguish between the called and calling parties in your dialplan, you can do something like Set(CALLING=yes) at priority and then fall through to priority+1. You could even put a Goto at priority and have two completely different sequences of commands for caller and called. If both legs of the call go to the same priority, it might be more fiddly to distinguish between them. If you want both legs to do the same, just put a NoOp at priority. However, even if you get them both to the same priority, they will NOT be bridged together! The option is specifically to UNbridge them and put both legs into the dialplan independently. The G option is most useful for transferring both legs of the call into a MeetMe conference. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] MOH from RCA jack?
Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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] 79xx's and call queues
If you figure it out, please let me know. I would actually love to _enable_ such a beep for my agents... (If it isn't there already...) Bob McDowell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary RichardsonSent: Thursday, February 16, 2006 5:23 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] 79xx's and call queues Hey,I'm testing out some call queues. I have 7940's and 7960's with the SIP 7.4 image.I have a queue that looks something like:[testqueue]strategy = rrmemorytimeout = 15retry = 5weight = 0 announce-frequency = 0joinemtpy = yesreportholdtime = yesI dynamically add a phone or two to the queue (AddQueueMember, not agents). When a caller calls in, connections are made and everything is fine. When a second person calls in, each queue member that is currently in a call gets a call waiting beep in a round robin fashion. Is this how it happens on non-Cisco phones, or is there something with how Cisco does line appearances causing this? This happens when 1 or more line appearances are configured on the phone.Thanks. ___ --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] MOH from RCA jack?
I have not done this but I could probably send you in the right direction. * MOH uses a he standard out of an audio program (ie mpg123) you should be able to add a custom mohtype in the musiconhold.conf file. All you need is to 'play' the audio from the line in on your MB and put it on STDOUT. Otherwise you can 'record' the message via line-in, edit it for length, and convert to MP3. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, February 17, 2006 8:42 AM To: Asterisk-users-list Subject: [Asterisk-Users] MOH from RCA jack? Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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: Bridge Calls with G()
On 02/17/06 21:50 Tony Mountifield said the following: I think it is more useful to transfer to the two separate priorities, but the documentation should reflect that. this makes sense. however the help text for 'show application dial' should then be updated to reflect this. i know this sounds pedantic, but it's got to work the way it's described without someone having to look through the code to find out why. i've updated the bug issue at http://bugs.digium.com/view.php?id=6523 with a new patch which retains the existing functionality but amends the helptext to be more accurate. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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: aastra v1.3.1 firmware
On 2/17/06, Gareth Owen [EMAIL PROTECTED] wrote: No, distinctive ring isn't supported in 1.3.1. You only have the option of setting the ring-tone on a per-line basis. hm, okay. is it a feature that will be built-in in the future? or can you say for sure it will not? thanks, cheers ___ --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] Quintum Tenor AX 24 Port SIP FXS Unsupported Media Type
Has anyone worked with one of these boxes and Asterisk? I have the Tenor AX registering 24 extensions just fine with asterisk but when I try to call one of the configured FXS extensions on the Tenor AX, I get Got SIP response 415 Unsupported Media Type back from xx.xx.xx.xx. I have tried various codecs and get the same. I am not having much luck on google, the Quintum manual nor voip-info. Maybe someone here has a quick answer? Thanks, Steve Totaro ___ --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] Playing sound File using GotoifTime function
This is my own GotoifTime section, which works swimmingly I might add: exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,Set(TIMEOUT(digit)=5) exten = s,4,Set(TIMEOUT(response)=10) exten = s,5,Background(fedwelcome) exten = s,6,GotoIfTime(*|*|1|jan?afterhours,s,1) ; New Year's Day exten = s,7,GotoIfTime(*|mon|25-31|may?afterhours,s,1) ; Memorial Day exten = s,8,GotoIfTime(*|*|4|jul?afterhours,s,1) ; 4th of July exten = s,9,GotoIfTime(*|mon|1-7|sep?afterhours,s,1) ; Labor Day exten = s,10,GotoIfTime(*|thu|22-28|nov?afterhours,s,1) ; Thanksgiving exten = s,11,GotoIfTime(*|*|25|dec?afterhours,s,1) ; Christmas Day exten = s,12,GotoIfTime(08:00-17:00|*|*|*?mainmenu,s,1) exten = s,13,Goto(afterhours,s,1) exten = s,14,Hangup- For your situation I would do something like: exten = s,12,GotoIfTime(08:00-17:00|*|*|*?playsoundfile,s,1) [playsoundfile] exten = s,1,Playback(soundfile) - I personally use a record extension to get my files on the server, striaght out of the wiki, sort of: [205record]; Record voice file to /tmp directory exten = 9205,1,Wait(2) ; Call 205 to Record new Sound Files exten = 9205,2,Record(/tmp/asterisk-recording:gsm) ; Press # to stop recording exten = 9205,3,Wait(2) exten = 9205,4,Playback(/tmp/asterisk-recording) ; Listen to your voice exten = 9205,5,wait(2) exten = 9205,6,Hangup- If I were to try and move the file to * from another PC, I'd probably upload said file to my webserver then 'wget' it down to *. Hope this helps. There's nothing too special about it, but I'd really proud of what we are going to be able to do with the awesome system. If anyone sees any faux pas in my config, please let me know. Bob McDowell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faisal InamSent: Thursday, February 16, 2006 11:33 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Playing sound File using GotoifTime function I want to play a sound file using GotoifTime function.1) What should be the appropriate format of this type of sound file?2) Is there any method to copy this file into the destination directory using the browser of a PC other than the asterisk PC (currently i am using cp to copy the file in /var/lib/asterisk/sounds on asterisk PC)???Waiting for ur kind reply !! Yahoo! MailUse Photomail to share photos without annoying attachments. ___ --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: BRI Newbie - What Hardware, PCI, in the US?
Brent Torrenga [EMAIL PROTECTED] wrote: We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am very interested in this too. My main motivation is to get the improved signaling. You got a number of answers, but it wasn't clear to me which of them were actually in the US. My vague recollection was that the Junghanns cards weren't supported in the US. Anyone know if this is still the case? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.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
[Asterisk-Users] Re: SPA-941 stutter tone
Kerry Garrison [EMAIL PROTECTED] wrote: I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? I can't help, but I do understand your pain. I tried to turn this off with the SPA-2000 with no luck. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.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
Re: [Asterisk-Users] Quintum Tenor AX 24 Port SIP FXS Unsupported Media Type
I did some config with one of these. When I got that error it was because I had only the G729 codec selected on the quintum and did not have the g729 license for the asterisk. I switch alaw on the quintum and it worked. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Steve Totaro wrote: Has anyone worked with one of these boxes and Asterisk? I have the Tenor AX registering 24 extensions just fine with asterisk but when I try to call one of the configured FXS extensions on the Tenor AX, I get "Got SIP response 415 "Unsupported Media Type" back from xx.xx.xx.xx. I have tried various codecs and get the same. I am not having much luck on google, the Quintum manual nor voip-info. Maybe someone here has a quick answer? Thanks, Steve Totaro ___ --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] MOH from RCA jack?
My * pc has an integrated soundcard, should be ok for this type of application, I'd get an RCA-to-line jack cable (radioshack should have those ;) I know * can play hold music from a streaming server, and I know some streaming servers can stream from a line in, so the combination of the two should do the trick I think. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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: Firmware version 1.3.1 released for Aastra IPphones
Nice one it works. Is there a complete list of all the options you can use in the config files? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Owen Sent: 17 February 2006 13:39 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IPphones The follow should work from the configuration files (aasta.cfg/MAC.cfg), although I haven't tried it... audio mode: mode Where mode is a number between 0 and 3 0 = speaker 1 = headset 2 = speaker/headset 3 = headset/speaker Gareth Lee Archer wrote: Any chance of getting a config option in that allows you set headset/speaker in the audio menu? Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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
[Asterisk-Users] SPA-941 hint
Hi Have someome a solution to use the hint function to have the signalling of the status of a extension on the SPA-941 phone ? Matteo ___ --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] A unique 'click to call' project - Could use some advice
Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a daily basis we place numerous (50-100) 'test' calls to various locations in the US; these 'test' calls are routed using one of three different phone systems: 1) The PSTN 2) Broadband phone platform one 3) Broadband phone platform two I have an Asterisk server configured that can terminate out three platforms listed above. Our support agents are behind a Televantage ACD using D-TermSeries E NEC phones. Each agent has a DID and are permitted to receive inbound calls on that DID. Here is my goal: Create a web application that will allow the agent to enter the following information into a form: 1) The agents DID 2) The platform the agent wishes to terminate a test call through (either 1,2,3 above) 3) The number the agent wishes to terminate to My thought is this form will generate a .call file in /var/spool/asterisk/outgoing that will then ring the agents station, pause, and terminate to the selected DID using the selected platform. I also thought about interacting directly with the AGI. I can successfully generate the .call files, and ring a station on the Asterisk server - the problem is the agents are not on the Asterisk server. Is there a way to use Asterisk to initiate these test calls? Is it possible to create a forwarding context to handle this? Any thoughts? Thanks for the help! Cheers, -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- -- -- ___ --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] Cheap BRI card
Hi, I'm asking to myself what's the main problem in using cheap BRI cards (30-60Euro, as these HFC-based) vs. great active cards as Eicon DIVA. Any help? -- Mimmus ___ --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] MOH from RCA jack?
Any idea how difficult it might be to get an integrated sound card to work properly with asterisk? (That seems to be the limiting factor or more time consuming part of doing this. Adapting the cables and audio levels is easy.) My * pc has an integrated soundcard, should be ok for this type of application, I'd get an RCA-to-line jack cable (radioshack should have those ;) I know * can play hold music from a streaming server, and I know some streaming servers can stream from a line in, so the combination of the two should do the trick I think. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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 ---End of Original Message- ___ --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] MOH from RCA jack?
Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? Rich, First, you will need an RCA to 1/8 cable from Radio Shack or something. Next, you will need a sound card in the machine. USB audio interfaces are cool too (and they usually have a high SN ratio). Then you need to setup a custom MOH class and use arecord: http://linuxcommand.org/man_pages/arecord1.html I haven't done this, but I was intrigued by your question and thought I'd look into it. Let us know how it turns out! -- Kristian Kielhofner ___ --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] SIP Problem Fedora Core 4 and Asterisk 1.2.4
Try turning off iptables (firewall) service. MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abhimanyu RapriaSent: Friday, February 17, 2006 2:19 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP Problem Fedora Core 4 and Asterisk 1.2.4 Fedora:Linux abcde 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/LinuxAsterisk: 1.2.4SIP Problem1. Asterisk sends SIP messages to Softphone. 2. Softphone receives SIP messages and replys back.3. Asterisk doesn't receive these replies and keeps on sending.Asterisk:Reliably Transmitting (no NAT) to 192.168.1.4:5060:OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK17299c89;rport From: "asterisk" sip:[EMAIL PROTECTED];tag=as1ab4b0c6To: sip:192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70 Date: Fri, 17 Feb 2006 07:13:32 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0---Retransmitting #1 (no NAT) to 192.168.1.4:5060 :OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK17299c89;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as1ab4b0c6To: sip:192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 17 Feb 2006 07:13:32 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0vicidial3*CLI---Retransmitting #2 (no NAT) to 192.168.1.4:5060:OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK17299c89;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as1ab4b0c6To: sip: 192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 17 Feb 2006 07:13:32 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0vicidial3*CLI ---Retransmitting #3 (no NAT) to 192.168.1.4:5060:OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK17299c89;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as1ab4b0c6To: sip: 192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 17 Feb 2006 07:13:32 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0vicidial3*CLI ---Retransmitting #4 (no NAT) to 192.168.1.4:5060:OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK17299c89;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as1ab4b0c6To: sip: 192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 17 Feb 2006 07:13:32 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0vicidial3*CLI ---Destroying call '[EMAIL PROTECTED]'Softphone:RECEIVE TIME: 7187271 RECEIVE 192.168.1.10:5060OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060 ;branch=z9hG4bK26e7bd24;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as6f689e4cTo: sip:192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 OPTIONS User-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 17 Feb 2006 07:18:12 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0RECEIVE TIME: 7188270RECEIVE 192.168.1.10:5060OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK26e7bd24;rport From: "asterisk" sip:[EMAIL PROTECTED];tag=as6f689e4cTo: sip:192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70 Date: Fri, 17 Feb 2006 07:18:12 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0RECEIVE TIME: 7189270RECEIVE 192.168.1.10:5060 OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK26e7bd24;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as6f689e4cTo: sip:192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 17 Feb 2006 07:18:12 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0RECEIVE TIME: 7190270RECEIVE 192.168.1.10:5060OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK26e7bd24;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as6f689e4cTo: sip: 192.168.1.4Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 17 Feb 2006 07:18:12 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0RECEIVE TIME: 7191269 RECEIVE 192.168.1.10:5060OPTIONS sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.10:5060 ;branch=z9hG4bK26e7bd24;rportFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as6f689e4cTo:
Re: [Asterisk-Users] A unique 'click to call' project - Could use some advice
On 2/17/06, Aloi, Christopher [EMAIL PROTECTED] wrote: Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a daily basis we place numerous (50-100) 'test' calls to various locations in the US; these 'test' calls are routed using one of three different phone systems: 1) The PSTN 2) Broadband phone platform one 3) Broadband phone platform two I have an Asterisk server configured that can terminate out three platforms listed above. Our support agents are behind a Televantage ACD using D-TermSeries E NEC phones. Each agent has a DID and are permitted to receive inbound calls on that DID. Here is my goal: Create a web application that will allow the agent to enter the following information into a form: 1) The agents DID 2) The platform the agent wishes to terminate a test call through (either 1,2,3 above) 3) The number the agent wishes to terminate to My thought is this form will generate a .call file in /var/spool/asterisk/outgoing that will then ring the agents station, pause, and terminate to the selected DID using the selected platform. I also thought about interacting directly with the AGI. I can successfully generate the .call files, and ring a station on the Asterisk server - the problem is the agents are not on the Asterisk server. Is there a way to use Asterisk to initiate these test calls? Is it possible to create a forwarding context to handle this? There's a couple ways to go about getting the calls to your agents. The first, and probably easiest would just be to dial out through the PSTN on Asterisk to the DIDs on the other system, but that obviously is a waste cost and resources wise. The second, would probably be to do a PRI DS1 tie line between the NEC and Asterisk and then just dial the DID on a Zap channel that belongs to that span to get to your agents. But your objectives sound like they are definitely achievable with some work. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [Asterisk-Users] asterisk silence suppression?
The patch you saw is not for the stable branch. Salu2 Jsalas Right, but try using this, i adapted it, no guarantees, i have not made tests, just modified it to apply properly, it would be great if some one can test it: http://chewbacca.ivsol.net/asterisk-1.2.1-silence-suppression-4.patch Regards On 2/17/06, Rob Lith [EMAIL PROTECTED] wrote: That a phone setting you must set to not supress silence - i.e. in X-Lite/eyeBeam in the advanced settings/audio there is a silence setting.Same for the SNOMs, most phones should have it.RegardsRob On 2/15/06, Dan Elder [EMAIL PROTECTED] wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads? Thx as always___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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] vISDN with Asterisk and HFC passive cards.
Has anyone dared to go down the visdn road. www.visdn.org I want an alternative to zaphfc for the passive HFC-PCI card. I managed to get the snapshot version of 17th Feb to compile against *1.2.4 et al. Now I am battling to get it configured. Could anyone with a working visdn.conf for 2 HFC cards and a good Dial/VISDN?? extensions.conf entry share this with me? Regards Allan Gee Phone: +27 21 4644400 Ext. 103 www.equation.co.za ___ --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] MOH from RCA jack?
Barix Instreamer takes RCA in and MP3 or ulaw stream out. Asterisk can use either for MOH. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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] MOH from RCA jack?
My Intel board's card works great with * for paging... I haven't ever tried it the other way. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, February 17, 2006 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MOH from RCA jack? Any idea how difficult it might be to get an integrated sound card to work properly with asterisk? (That seems to be the limiting factor or more time consuming part of doing this. Adapting the cables and audio levels is easy.) My * pc has an integrated soundcard, should be ok for this type of application, I'd get an RCA-to-line jack cable (radioshack should have those ;) I know * can play hold music from a streaming server, and I know some streaming servers can stream from a line in, so the combination of the two should do the trick I think. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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 ---End of Original Message- ___ --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] Intrado / VoIP E911
Ok, So... we've been looking at Intrado as a solution for national E911. They claim to be able to offer FCC compliant E911 services for VoIP companies. However, as I look into things further, they don't seem to have links to all the PSAPs for E911. Now, I understand if the PSAP is not capable of receiving E911 information, the VoIP provider is under no obligation to provide it. However, a few PSAPs that I have tested ARE able to receive E911, but Intrado is still dialing a 10-digit number to get to them, resulting in no ALI information being passed. Does anyone, more experienced with Intrado then I, have any thoughts on this? How can Intrado claim to be able to provide services that make a VoIP provider FCC Legal and Compliant, and yet still send calls to E911 equiped PSAPs without the ALI information?! ___ --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 unique 'click to call' project - Could use some advice
You
create a context in your dialplan that accepts the DID to call as a variable
using the SetVar: syntax in your .call file. You then set up the context to call
your agent, and when they pick up, the context takes the variable you set in
your .call file asthe dialstring argument for a subsequent Dial(). Once
the DID picks up, the calls are bridged together. Whatever web scripting
language you use writes the .call file, and you use POSTed arguments or
querystrings:
http://foo.com/call?context=MyContextAgent=SIP/DID=1551212
You
can see this in action at www.landmarkhomes.ca - click on any of
the pretty buttons that say "Call us now"
However, I have noticed that * 1.2.x will not wait for the caller to pick
up before executing the rest of the directives in the context- it keeps
executing regardless of the calling party's pickup status. Using * 1.0.x the
context will wait for the caller to pick up before placing the call to the
callee (i.e. executing the rest of the directives in the
context)
.call
file (shortened to relevant)
Channel: SIP/ (if you are using SIP
phones)
SetVar: DID=XXX
Context: MyContext
[MyContext]
exten
= s,1,Dial(ZAP/g0/${DID})
hth
-Original Message-From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]Sent: Friday, February 17, 2006 8:07
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] A unique 'click to call'
project - Could use some advice
Hello List,
I work for an IP communication provider in upstate NY
as the engineer assisting our technical support team.
We provide a number of different Telco systems to
residential subscribers; and in an effort to more effectively trouble shoot
termination problems I came up with the idea of creating a click to call
system that will allow our agents to effortlessly place test
calls.
On a daily basis we place numerous (50-100) 'test'
calls to various locations in the US; these 'test' calls are routed using one
of three different phone systems:
1) The PSTN
2) Broadband phone platform one
3) Broadband phone platform two
I have an Asterisk server configured that can
terminate out three platforms listed above.
Our support agents are behind a Televantage ACD using
D-TermSeries E NEC phones.
Each agent has a DID and are permitted to receive
inbound calls on that DID.
Here is my goal:
Create a web application that will allow the agent to
enter the following information into a form:
1) The agents DID
2) The platform the agent wishes to terminate a test
call through (either 1,2,3 above)
3) The number the agent wishes to terminate to
My thought is this form will generate a .call file in
/var/spool/asterisk/outgoing that will then ring the agents station, pause,
and terminate to the selected DID using the selected platform. I also
thought about interacting directly with the AGI.
I can successfully generate the .call files, and ring
a station on the Asterisk server - the problem is the agents are not on the
Asterisk server.
Is there a way to use Asterisk to initiate these test
calls?
Is it possible to create a forwarding context to
handle this?
Any thoughts?
Thanks for the help!
Cheers,
--
Christopher T. Aloi USA Datanet - Technical Support
Engineer 318 South Clinton Street Syracuse, NY 13202
C: (315) 569
4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- --
--
___
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Re: [Asterisk-Users] MOH from RCA jack?
Actually, with my suggestion, you would be using the soundcard with whatever streaming mp3 server you choose to use, some kinda shoutcast server I guess, so there shouldn't be any asterisk-soundcard interaction.. * just takes its moh from the streaming server. On Fri, 2006-02-17 at 09:18 -0600, Rich Adamson wrote: Any idea how difficult it might be to get an integrated sound card to work properly with asterisk? (That seems to be the limiting factor or more time consuming part of doing this. Adapting the cables and audio levels is easy.) My * pc has an integrated soundcard, should be ok for this type of application, I'd get an RCA-to-line jack cable (radioshack should have those ;) I know * can play hold music from a streaming server, and I know some streaming servers can stream from a line in, so the combination of the two should do the trick I think. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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 ---End of Original Message- ___ --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 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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 unique 'click to call' project - Could use someadvice
Why dont you use Local and router functionality to find a route to PSTN based agents? W From: Aloi, Christopher [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] A unique 'click to call' project - Could use someadvice Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a daily basis we place numerous (50-100) 'test' calls to various locations in the US; these 'test' calls are routed using one of three different phone systems: 1) The PSTN 2) Broadband phone platform one 3) Broadband phone platform two I have an Asterisk server configured that can terminate out three platforms listed above. Our support agents are behind a Televantage ACD using D-TermSeries E NEC phones. Each agent has a DID and are permitted to receive inbound calls on that DID. Here is my goal: Create a web application that will allow the agent to enter the following information into a form: 1) The agents DID 2) The platform the agent wishes to terminate a test call through (either 1,2,3 above) 3) The number the agent wishes to terminate to My thought is this form will generate a .call file in /var/spool/asterisk/outgoing that will then ring the agents station, pause, and terminate to the selected DID using the selected platform. I also thought about interacting directly with the AGI. I can successfully generate the .call files, and ring a station on the Asterisk server - the problem is the agents are not on the Asterisk server. Is there a way to use Asterisk to initiate these test calls? Is it possible to create a forwarding context to handle this? Any thoughts? Thanks for the help! Cheers, -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- -- -- ___ --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] MOH from RCA jack?
Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? Rich, First, you will need an RCA to 1/8 cable from Radio Shack or something. Next, you will need a sound card in the machine. USB audio interfaces are cool too (and they usually have a high SN ratio). Then you need to setup a custom MOH class and use arecord: http://linuxcommand.org/man_pages/arecord1.html I haven't done this, but I was intrigued by your question and thought I'd look into it. Let us know how it turns out! The usb audio interface sounds very cool! Have you played with any that has a line-in jack or have any specific device recommendations? (Wondering now if such a device could be made to work as an asterisk overhead paging system cabled to an amplifier, etc.) ___ --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-941 stutter tone
I just double checked my SPA-841. You can change the dial tone in the Web config on the Regional page. I just copied the Dial Tone: to the MWI Dial Tone field and it didnt stutter after that. I'm not sure if its the same with the 941, but i've heard the phone configs are similar. Hope this helps. Kerry Garrison wrote: I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.techdatapros.com http://www.techdatapros.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] How do I install speex for asterisk?
Jesus, If you recompile Asterisk and still have problems, take a look in /codecs/Makefile. It'll tell you where Asterisk expects to find stuff in order to trigger the building of the speex-related objects. If the build goes as planned, the /codecs directory will contain three speex-related files: - codec_speex.c - codec_speex.o - codec_speex.so On my box, a yum install speex-devel took care of everything for me. I only did this to ensure some changes that I made to codec_speex.c would compile, so I have no experience with using the actual codec. Nonetheless, I hope this information is helpful to you. Good luck, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Mark Phillips wrote: If you did a make install with speex then everythings where it should be. Just do a make; make clean with asterisk and all will be fine. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --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 ___ --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] Grandstream GXP-2000
Hi, I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones. - Is it a good choice (budget limit of 100 Euro/phone is mandatory)? - Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk? Thanks in advance -- Mimmus ___ --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 unique 'click to call' project - Could use some advice
I'm using the telnet manager interface with the 'originate' command, just a little perl script that connects and has asterisk dial the selected number. It rings the extension first, if they pick up, it'll dial the remote number. It's one of the showcase features of the new phonesystem for us :) and it was surprisingly easy to implement. -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --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] using AMP custom extensions
Hi all, I'm trying to setup a custom extension in AMP (yes i can code it by hand but the on-site admin that does moves changes cannot). I've tried the following add cutom extension 600 in the dial box i have Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) this doesnt work as these lines are added to extensions_additional.conf exten = 600,1,Macro(exten-vm,novm,600) exten = 600,hint,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) anyone know why the hint line is there? does anyone have custom IAX extensions configured thru AMP? Thanks in advance Bails ___ --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 unique 'click to call' project - Could usesomeadvice
Hello, I'm not sure what you mean, could you elaborate? Thanks, -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- -- -- From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 10:47 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] A unique 'click to call' project - Could usesomeadvice Why dont you use Local and router functionality to find a route to PSTN based agents? W From: Aloi, Christopher [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 10:07 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] A unique 'click to call' project - Could use someadvice Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a daily basis we place numerous (50-100) 'test' calls to various locations in the US; these 'test' calls are routed using one of three different phone systems: 1) The PSTN 2) Broadband phone platform one 3) Broadband phone platform two I have an Asterisk server configured that can terminate out three platforms listed above. Our support agents are behind a Televantage ACD using D-TermSeries E NEC phones. Each agent has a DID and are permitted to receive inbound calls on that DID. Here is my goal: Create a web application that will allow the agent to enter the following information into a form: 1) The agents DID 2) The platform the agent wishes to terminate a test call through (either 1,2,3 above) 3) The number the agent wishes to terminate to My thought is this form will generate a .call file in /var/spool/asterisk/outgoing that will then ring the agents station, pause, and terminate to the selected DID using the selected platform. I also thought about interacting directly with the AGI. I can successfully generate the .call files, and ring a station on the Asterisk server - the problem is the agents are not on the Asterisk server. Is there a way to use Asterisk to initiate these test calls? Is it possible to create a forwarding context to handle this? Any thoughts? Thanks for the help! Cheers, -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- -- -- ___ --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] [Fwd: using AMP custom extensions]
OK I'm answering my own question but if i add a custom extension in AMP with no dial string. Then add a dialstring in extensions_custom.conf like exten = 600,1,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) it works Bails ---BeginMessage--- Hi all, I'm trying to setup a custom extension in AMP (yes i can code it by hand but the on-site admin that does moves changes cannot). I've tried the following add cutom extension 600 in the dial box i have Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) this doesnt work as these lines are added to extensions_additional.conf exten = 600,1,Macro(exten-vm,novm,600) exten = 600,hint,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) anyone know why the hint line is there? does anyone have custom IAX extensions configured thru AMP? Thanks in advance Bails ---End Message--- ___ --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] problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion)
Nik, This definitely helps! Please check your dial command. You've got Dial(Zap/0/mynumber) and I think you might possibly want it to be something like this: Dial(Zap/1/mynumber) or Dial(Zap/g0/mynumber) I don't recall there being a zap channel zero, but it is common to have a group zero. I would recommend trying Zap channel 1 - Dial(Zap/1/mynumber) - before trying the group. Again, please get the debug info. The CHANUNAVAIL message made it easier to diagnose this issue. Don't give up! The education you are getting will help you in the long run and in a few months you'll be able to help a * newbie with the same issues! -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600 Sent: Friday, February 17, 2006 12:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion) On 2/15/06, Michael Collins [EMAIL PROTECTED] wrote: Nik, Looks like you're making some progress. When I first started using [EMAIL PROTECTED] I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten = 555,1,Dial(Zap/1/5595551212) I chose a specific Zap channel and the exact digits that I wanted to send to the telephone company. This helped me figure out what to dial. The other thing you can do is log on to the CLI and turn on PRI debugging: pri debug span 1 This will cause PRI debug messages to display on the console. It might take a while but you will learn to read those debug messages. You can also post them to the list and we'll help you to interpret them. -MC ok, thanks for your support, now i've enabled debug on span 1, and i've make a new entry in extension.conf: exten = 444,1,Dial(Zap/0/mynumber) when i call 444 i get in the logs: Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Setting NAT on RTP to 0 Feb 17 03:50:59 DEBUG[3607] chan_sip.c: Checking SIP call limits for device 102 Feb 17 03:50:59 DEBUG[3607] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Feb 17 03:50:59 VERBOSE[4262] logger.c: -- Executing Dial(SIP/102-2079, Zap/0/mynumber) in new stack Feb 17 03:50:59 NOTICE[4262] app_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown) Feb 17 03:50:59 VERBOSE[4262] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Feb 17 03:50:59 DEBUG[4262] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. it seems that the only information it gives mi is: app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. so it seems that i don't have channel for outgoing calls? how can i check it? maybe there is another logfile more detailed? thanks a lot for your help... ___ --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] simple iaxmoden configuration
Hi everyone, I am trying to get iaxmodem up and running. This is a very basic setup, which at this moment should only answer incoming faxes. What I did: zapata.conf (rest of it should be fine): faxdetect=incoming group = 1 channel = 1-2 context=from-pstn iax.conf: [200] username=200 type=friend callerid=Fax 200 secret=dooo host=dynamic notransfer=yes allow=all context=from-pstn extensions.conf: [from-pstn] exten = fax,1,Dial(IAX2/200) /etc/iaxmodem-cfg.ttyIAX: device /dev/ttyIAX port4569 refresh 300 server 127.0.0.1 peername200 secret dooo cidname 200 cidnumber 200 codec slinear When trying fo fax, all I get is: Extension '265399' in context 'default' from '0123456789' does not exist. Rejecting call on channel 0/1, span 1 When changing the extensions.conf to: [default] exten = 265399,1,Dial(IAX2/200) it is working perfectly. Where are the mistakes? 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
RE: [Asterisk-Users] How do I install speex for asterisk?
Mark: I did so, but that did not make asterisk to integrate speex. Do I have to tweak something in speex after installation? This is some of asterisk output when I try to use speex: -- Accepting AUTHENTICATED call from 192.168.2.32: requested format = speex, requested prefs = (), actual format = speex, host prefs = (speex|ilbc|gsm), priority = mine -- Executing Macro(IAX2/ext2-2, outbound|14802012944) in new stack -- Call accepted by 66.234.228.160 (format speex) -- Format for call is speex -- IAX2/66.234.228.160:4569-5 is circuit-busy -- Hungup 'IAX2/66.234.228.160:4569-5' Feb 17 09:20:42 WARNING[1811]: chan_iax2.c:1717 attempt_transmit: Max retries exceeded to host 66.234.228.166 on IAX2/66.234.228.166:4569-9 (type = 6, subclass= 1, ts=8, seqno=0) -- Hungup 'IAX2/66.234.228.166:4569-9' == No one is available to answer at this time (1:0/0/0) Feb 17 09:20:52 WARNING[2508]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'internal' -- Hungup 'IAX2/ext2-2' -- Registered IAX2 'ext1' (AUTHENTICATED) at 192.168.2.31:4569 Feb 17 09:30:42 NOTICE[1811]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'ext1' to 60 seconds (requested 300) -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 17:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How do I install speex for asterisk? If you did a make install with speex then everythings where it should be. Just do a make; make clean with asterisk and all will be fine. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --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 ___ --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 unique 'click to call' project - Could usesome advice
Colin,
Thanks for your assistance.
Reading over your advice I seem to still be a bit
confused.
My agents are not on the Asterisk server; it appears in
your advice that my the call will travel this path:
WWW interface -- agent enters their DID, platform
to use, and termination DID -- AST calls agent -- Agent calls
termination DID
If my agents are not on the Asterisk server (believe
me, I wish there were) :) how will this work?
I need a way to pass both the desired termination DID
and the origination DID.
Maybe I missed something
Thanks,
--
Christopher T. Aloi USA Datanet - Technical Support
Engineer 318 South Clinton Street Syracuse, NY 13202
C: (315) 569
4033 O:
(315) 579 7074 E: [EMAIL PROTECTED] -- --
--
From: Colin Anderson
[mailto:[EMAIL PROTECTED] Sent: Friday, February 17,
2006 10:42 AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] A unique 'click to call'
project - Could usesome advice
You
create a context in your dialplan that accepts the DID to call as a variable
using the SetVar: syntax in your .call file. You then set up the context to call
your agent, and when they pick up, the context takes the variable you set in
your .call file asthe dialstring argument for a subsequent Dial(). Once
the DID picks up, the calls are bridged together. Whatever web scripting
language you use writes the .call file, and you use POSTed arguments or
querystrings:
http://foo.com/call?context=MyContextAgent=SIP/DID=1551212
You
can see this in action at www.landmarkhomes.ca - click on any of
the pretty buttons that say "Call us now"
However, I have noticed that * 1.2.x will not wait for the caller to pick
up before executing the rest of the directives in the context- it keeps
executing regardless of the calling party's pickup status. Using * 1.0.x the
context will wait for the caller to pick up before placing the call to the
callee (i.e. executing the rest of the directives in the
context)
.call
file (shortened to relevant)
Channel: SIP/ (if you are using SIP
phones)
SetVar: DID=XXX
Context: MyContext
[MyContext]
exten
= s,1,Dial(ZAP/g0/${DID})
hth
-Original Message-From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]Sent: Friday, February 17, 2006 8:07
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] A unique 'click to call'
project - Could use some advice
Hello List,
I work for an IP communication provider in upstate NY
as the engineer assisting our technical support team.
We provide a number of different Telco systems to
residential subscribers; and in an effort to more effectively trouble shoot
termination problems I came up with the idea of creating a click to call
system that will allow our agents to effortlessly place test
calls.
On a daily basis we place numerous (50-100) 'test'
calls to various locations in the US; these 'test' calls are routed using one
of three different phone systems:
1) The PSTN
2) Broadband phone platform one
3) Broadband phone platform two
I have an Asterisk server configured that can
terminate out three platforms listed above.
Our support agents are behind a Televantage ACD using
D-TermSeries E NEC phones.
Each agent has a DID and are permitted to receive
inbound calls on that DID.
Here is my goal:
Create a web application that will allow the agent to
enter the following information into a form:
1) The agents DID
2) The platform the agent wishes to terminate a test
call through (either 1,2,3 above)
3) The number the agent wishes to terminate to
My thought is this form will generate a .call file in
/var/spool/asterisk/outgoing that will then ring the agents station, pause,
and terminate to the selected DID using the selected platform. I also
thought about interacting directly with the AGI.
I can successfully generate the .call files, and ring
a station on the Asterisk server - the problem is the agents are not on the
Asterisk server.
Is there a way to use Asterisk to initiate these test
calls?
Is it possible to create a forwarding context to
handle this?
Any thoughts?
Thanks for the help!
Cheers,
--
Christopher T. Aloi USA Datanet - Technical Support
Engineer 318 South Clinton Street Syracuse, NY 13202
C: (315) 569
4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- --
--
___
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Re: [Asterisk-Users] MOH from RCA jack?
Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? Rich, First, you will need an RCA to 1/8 cable from Radio Shack or something. Next, you will need a sound card in the machine. USB audio interfaces are cool too (and they usually have a high SN ratio). Then you need to setup a custom MOH class and use arecord: http://linuxcommand.org/man_pages/arecord1.html I haven't done this, but I was intrigued by your question and thought I'd look into it. Let us know how it turns out! The usb audio interface sounds very cool! Have you played with any that has a line-in jack or have any specific device recommendations? (Wondering now if such a device could be made to work as an asterisk overhead paging system cabled to an amplifier, etc.) Rich, I had one a while back that worked fine. I think it was someting for a mac called iMic by Griffin. It worked with ALSA under Linux. P.S. - You need ALSA to use arecord anyways :). Looking at the other posts, it really looks like this would be the best, most direct solution... -- Kristian Kielhofner ___ --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] Quintum Tenor AX 24 Port SIP FXS UnsupportedMedia Type
That was it. There is way more configuration in these things than I need and I guess I have to RTFM. VERY impressive box. I just want to use it as an FXS Gateway. I set the codecs to ulaw and alaw. I configured the SIP useragents and as I said, it is registering with asterisk. Problem now in the console is 484 Address Incomplete There are so many config options, I have no idea how to just map sip useragent 1 to FXS port 1 which is what I assume is causing the above error. Thanks, Steve Totaro http://www.asteriskhelpdesk.com _ From: Michael Sampson [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quintum Tenor AX 24 Port SIP FXS UnsupportedMedia Type I did some config with one of these. When I got that error it was because I had only the G729 codec selected on the quintum and did not have the g729 license for the asterisk. I switch alaw on the quintum and it worked. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 952-936-4000 Steve Totaro wrote: Has anyone worked with one of these boxes and Asterisk? I have the Tenor AX registering 24 extensions just fine with asterisk but when I try to call one of the configured FXS extensions on the Tenor AX, I get Got SIP response 415 Unsupported Media Type back from xx.xx.xx.xx. I have tried various codecs and get the same. I am not having much luck on google, the Quintum manual nor voip-info. Maybe someone here has a quick answer? Thanks, Steve Totaro _ ___ --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] RE: ZAP extension, DTMF?
How is your echo can the issue? Did you disable the echo can and solve the DTMF issue? I actually think my echo can had gotten into some odd state, a restart of the tellabs board fixed the issue. ___ --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] Grandstream GXP-2000
I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones. - Is it a good choice (budget limit of 100 Euro/phone is mandatory)? - Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk? if you've never tried a phone, it's always a risk. I'd advise against buying 'any' 15 phones without first trying at least 1.. However, the GXP-2000 is an okay phone. The Thomson ST2030 however is firmer (almost same price) but doesn't have the BLF and MWI. It depends on what features you need. cheers ___ --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 unique 'click to call' project - Could usesomeadvice
You could do something like :
[router-local]
exten = _613XXX,1,Goto(trunklocal,
${EXTEN:${TRUNKMSD3}},1)
exten = _613XXX,2,Congestion
[router-ld]
exten = _1NX,1,Goto(trunkld,91${EXTEN},1)
exten = _1NX,2,Congestion
[trunklocal]
exten =
XXX,1,Dial(Zap/g1/${EXTEN}|20)
exten = XXX,2,Congestion
[router-agents]
include = router-local
include = router-ld
include = trunklocal
[agents]
exten = s,1,Dial(Local/[EMAIL PROTECTED])
In your call file specify agents
as your context to call agents through PSTN
Thanks,
W
From: Aloi,
Christopher [mailto:[EMAIL PROTECTED]
Sent: Friday, February 17, 2006
11:56 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] A
unique 'click to call' project - Could usesomeadvice
Colin,
Thanks for your assistance.
Reading over your advice I seem to still
be a bit confused.
My agents are not on the Asterisk server;
it appears in your advice that my the call will travel this path:
WWW interface -- agent enters their
DID, platform to use, and termination DID -- AST calls agent -- Agent
calls termination DID
If my agents are not on the Asterisk
server (believe me, I wish there were) :) how will this work?
I need a way to pass both the desired
termination DID and the origination DID.
Maybe I missed something
Thanks,
--
Christopher T. Aloi
USA Datanet - Technical Support Engineer
318 South Clinton Street
Syracuse, NY 13202
C: (315) 569 4033
O: (315) 579 7074
E: [EMAIL PROTECTED]
-- -- --
From: Colin
Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, February 17, 2006
10:42 AM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] A
unique 'click to call' project - Could usesome advice
You create a context in your dialplan that
accepts the DID to call as a variable using the SetVar: syntax in your .call
file. You then set up the context to call your agent, and when they pick up,
the context takes the variable you set in your .call file asthe
dialstring argument for a subsequent Dial(). Once the DID picks up, the calls
are bridged together. Whatever web scripting language you use writes the .call
file, and you use POSTed arguments or querystrings:
http://foo.com/call?context=MyContextAgent=SIP/DID=1551212
You can see this in action at www.landmarkhomes.ca - click on any of
the pretty buttons that say Call us now
However, I have noticed that * 1.2.x will
not wait for the caller to pick up before executing the rest of the directives
in the context- it keeps executing regardless of the calling party's
pickup status. Using * 1.0.x the context will wait for the caller to pick up
before placing the call to the callee (i.e. executing the rest of the
directives in the context)
.call file (shortened to relevant)
Channel: SIP/
(if you are using SIP phones)
SetVar: DID=XXX
Context: MyContext
[MyContext]
exten = s,1,Dial(ZAP/g0/${DID})
hth
-Original Message-
From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]
Sent: Friday, February 17, 2006
8:07 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] A unique
'click to call' project - Could use some advice
Hello List,
I work for an IP communication provider in
upstate NY as the engineer assisting our technical support team.
We provide a number of different Telco
systems to residential subscribers; and in an effort to more effectively
trouble shoot termination problems I came up with the idea of creating a click
to call system that will allow our agents to effortlessly place test calls.
On a daily basis we place numerous
(50-100) 'test' calls to various locations in the US; these 'test' calls are routed
using one of three different phone systems:
1) The PSTN
2) Broadband phone platform one
3) Broadband phone platform two
I have an Asterisk server configured that
can terminate out three platforms listed above.
Our support agents are behind a
Televantage ACD using D-TermSeries E NEC phones.
Each agent has a DID and are permitted to
receive inbound calls on that DID.
Here is my goal:
Create a web application that will allow
the agent to enter the following information into a form:
1) The agents DID
2) The platform the agent wishes to
terminate a test call through (either 1,2,3 above)
3) The number the agent wishes to
terminate to
My thought is this form will generate a
.call file in /var/spool/asterisk/outgoing that will then ring the agents
station, pause, and terminate to the selected DID using the selected
platform. I also thought about interacting directly with the AGI.
I can successfully generate the .call
files, and ring a station on the Asterisk server - the problem is the agents
are not on the Asterisk server.
Is there a
RE: [Asterisk-Users] A unique 'click to call' project - Could use some advice
Same
as before but instead of SIP as the originationchannel you pass
ZAP/g0/XXX (the DID of the agent) to your .call file. In fact, this is
exactly how the www.landmarkhomes.ca
script works (it calls the guy who entered his phone number in the website, when
he picks up, it calls the salesperson's cell number and the two are bridged
together)
The
drawback is, of course, that it uses 2 ZAP channels to bridge the call together,
but this isn't a problem I guess for you since you seem to have ZAP channels
coming out of your yinyang.
I have
an implementation in Active Server Pages (we are a MS shop) that I can send you
- it's suprisingly simple- but it could be easily modified for PHP or what
have you.
-Original Message-From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]Sent: Friday, February 17, 2006 9:56
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users] A unique 'click to call'
project - Could usesome advice
Colin,
Thanks for your assistance.
Reading over your advice I seem to still be a bit
confused.
My agents are not on the Asterisk server; it appears
in your advice that my the call will travel this path:
WWW interface -- agent enters their DID, platform
to use, and termination DID -- AST calls agent -- Agent calls
termination DID
If my agents are not on the Asterisk server (believe
me, I wish there were) :) how will this work?
I need a way to pass both the desired termination DID
and the origination DID.
Maybe I missed something
Thanks,
--
Christopher T. Aloi USA Datanet - Technical Support
Engineer 318 South Clinton Street Syracuse, NY 13202
C: (315) 569
4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- --
--
From: Colin Anderson
[mailto:[EMAIL PROTECTED] Sent: Friday, February
17, 2006 10:42 AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] A unique 'click to call'
project - Could usesome advice
You
create a context in your dialplan that accepts the DID to call as a variable
using the SetVar: syntax in your .call file. You then set up the context to
call your agent, and when they pick up, the context takes the variable you set
in your .call file asthe dialstring argument for a subsequent Dial().
Once the DID picks up, the calls are bridged together. Whatever web scripting
language you use writes the .call file, and you use POSTed arguments or
querystrings:
http://foo.com/call?context=MyContextAgent=SIP/DID=1551212
You
can see this in action at www.landmarkhomes.ca - click on any of
the pretty buttons that say "Call us now"
However, I have noticed that * 1.2.x will not wait for the caller to
pick up before executing the rest of the directives in the context- it
keeps executing regardless of the calling party's pickup status. Using * 1.0.x
the context will wait for the caller to pick up before placing the call to the
callee (i.e. executing the rest of the directives in the
context)
.call file (shortened to relevant)
Channel: SIP/ (if you are using SIP
phones)
SetVar: DID=XXX
Context: MyContext
[MyContext]
exten = s,1,Dial(ZAP/g0/${DID})
hth
-Original Message-From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]Sent: Friday, February 17, 2006 8:07
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] A unique 'click to call'
project - Could use some advice
Hello List,
I work for an IP communication provider in upstate
NY as the engineer assisting our technical support team.
We provide a number of different Telco systems to
residential subscribers; and in an effort to more effectively trouble shoot
termination problems I came up with the idea of creating a click to call
system that will allow our agents to effortlessly place test
calls.
On a daily basis we place numerous (50-100) 'test'
calls to various locations in the US; these 'test' calls are routed using
one of three different phone systems:
1) The PSTN
2) Broadband phone platform one
3) Broadband phone platform two
I have an Asterisk server configured that can
terminate out three platforms listed above.
Our support agents are behind a Televantage ACD
using D-TermSeries E NEC phones.
Each agent has a DID and are permitted to receive
inbound calls on that DID.
Here is my goal:
Create a web application that will allow the agent
to enter the following information into a form:
1) The agents DID
2) The platform the agent wishes to terminate a
test call through (either 1,2,3 above)
3) The number the agent wishes to terminate to
My thought is this form will generate a
Re: [Asterisk-Users] simple iaxmoden configuration
Christian Lox wrote: Hi everyone, I am trying to get iaxmodem up and running. This is a very basic setup, which at this moment should only answer incoming faxes. extensions.conf: [from-pstn] exten = fax,1,Dial(IAX2/200) When trying fo fax, all I get is: Extension '265399' in context 'default' from '0123456789' does not exist. Rejecting call on channel 0/1, span 1 When changing the extensions.conf to: [default] exten = 265399,1,Dial(IAX2/200) it is working perfectly. Where are the mistakes? The mistake is you have no s extension in [from-pstn] You might try something like this: exten = s,1,Wait(1) ;sometimes you need to wait to get callerid exten = s,2,Answer() exten = fax,1,Dial(IAX2/200) You can't tell if it's a fax until it is answered. Read up on the use of the 's' extension. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
Re: [Asterisk-Users] [OT] List messages and end user outages
My guess would be that the mqueu was just too busy. On 2/17/06, Robert Webb [EMAIL PROTECTED] wrote: Sorry, this is off topic to asterisk itself, but is about the list server. I had a power failure lastnight at home, where my email server resides, and my network was down for about 20 minutes, that was after 45 minutes of uptime on UPS. Since power was restored, around 9:45 PM EST on 2/16, I have not received a single post from the users, biz, or dev lists. Normally when this has happened in the past, it has taken 24 hours for the list server to start sending to my email server again. My question is why so long? I am on other lists and it might take an hour or so for the messages to start showing up, but why 24 hours for a 20 minute loss of contact with my email server? Robert P.S. - If there is somewhere else this question should be directed, that would be constructive, please feel free to let me know. ___ --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] How do I install speex for asterisk?
Elaborating a little more I checked for files suggested by Matthew Roth: If the build goes as planned, the /codecs directory will contain three speex-related files: - codec_speex.c - codec_speex.o - codec_speex.so Then ran the show modules command and now codec_speex shows as loaded by asterisk! But still cannot make or receive calls using speex. I am investigating with my VOIP provider.. Thanks to all of you. -Original Message- From: Jesus E Zepeda [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 09:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? Mark: I did so, but that did not make asterisk to integrate speex. Do I have to tweak something in speex after installation? This is some of asterisk output when I try to use speex: -- Accepting AUTHENTICATED call from 192.168.2.32: requested format = speex, requested prefs = (), actual format = speex, host prefs = (speex|ilbc|gsm), priority = mine -- Executing Macro(IAX2/ext2-2, outbound|14802012944) in new stack -- Call accepted by 66.234.228.160 (format speex) -- Format for call is speex -- IAX2/66.234.228.160:4569-5 is circuit-busy -- Hungup 'IAX2/66.234.228.160:4569-5' Feb 17 09:20:42 WARNING[1811]: chan_iax2.c:1717 attempt_transmit: Max retries exceeded to host 66.234.228.166 on IAX2/66.234.228.166:4569-9 (type = 6, subclass= 1, ts=8, seqno=0) -- Hungup 'IAX2/66.234.228.166:4569-9' == No one is available to answer at this time (1:0/0/0) Feb 17 09:20:52 WARNING[2508]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'internal' -- Hungup 'IAX2/ext2-2' -- Registered IAX2 'ext1' (AUTHENTICATED) at 192.168.2.31:4569 Feb 17 09:30:42 NOTICE[1811]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'ext1' to 60 seconds (requested 300) -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 17:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How do I install speex for asterisk? If you did a make install with speex then everythings where it should be. Just do a make; make clean with asterisk and all will be fine. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --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:
Re: [Asterisk-Users] [OT] List messages and end user outages
On Feb 17, 2006, at 6:36 AM, Robert Webb wrote: Sorry, this is off topic to asterisk itself, but is about the list server. I had a power failure lastnight at home, where my email server resides, and my network was down for about 20 minutes, that was after 45 minutes of uptime on UPS. Since power was restored, around 9:45 PM EST on 2/16, I have not received a single post from the users, biz, or dev lists. Normally when this has happened in the past, it has taken 24 hours for the list server to start sending to my email server again. My question is why so long? I am on other lists and it might take an hour or so for the messages to start showing up, but why 24 hours for a 20 minute loss of contact with my email server? Are you using dynamic DNS? Perhaps it's a DNS update issue? ___ --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.729 woes
I have some Digium licensed Digium codecs, but when making a call and transcoding the call is only heard in one direction? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
[Asterisk-Users] indications issues in Singapore?
Hi all, haven't seen many posts about asterisk in Singapore... Getting a server going there and was wondering if TDM400Ps will be fine in FCC mode, and if there are indications / cadence values that I should be putting on there as in other international locations. Seen an unsettling post on voip-info about Singapore issues with Call Polarity/Hangup Detection -- crossing my fingers I don't run into that problem :-) Any tips appreciated, -- Chris Earle System Solutions Specialist -- 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
RE: [Asterisk-Users] Cisco 7960 won't register
Thank you, I added both to SIPDefault.cnf and I am seeing traffic now. Its strange that it would default to not registering, and wouldnt try to register even if I went into the phone and did a register 1 1 command. Im getting a 401 Unauthorized back from Asterisk now. With the following sip.conf entry and the previously posted phone config files, shouldnt I be okay? Is there anything different that Cisco does that I need to account for in Asterisk? [username] type=friend username=username secret=password qualify=yes allow=all nat=yes host=dynamic canreinvite=no dtmfmode=rfc2833 context=contextname From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: 16 February 2006 18:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7960 won't register Add the following to your Config FIles. Either one is fine. # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 360 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Newton Sent: Thursday, February 16, 2006 7:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 won't register Hello all, Ive got a Cisco 7960 running version 7.4 firmware (heard there were problems with 7.5) and I cant get it to register with Asterisk. Ive stripped down my configs on the phone to a bare minimum, and posted them below. Basically, the Cisco phone sends absolutely no packets to the proxy when it gets booted. If I make an outgoing call I see traffic getting to Asterisk, but thats the only time I do; it doesnt even *try* to register (confirmed with sip debug on the Asterisk server, and debug sip-messages on the phone itself.) Im hoping Im not the only one to have ever had this problem, and would love it if someone could help. As promised, here are my conf files: SIPDefault.cnf: image_version:P0S3-07-4-00 proxy1_address:asterisk..com telnet_level:2 SIPmacaddress.cnf line1_name:username line1_authname:username line1_password:password user_info:none ___ --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 unique 'click to call' project - Could usesome advice
Thanks Colin!
Makes sense; I will work on this later
today.
If you can, sending the example would be
great.
Thanks,
--
Christopher T.
Aloi USA
Datanet - Technical Support Engineer 318 South Clinton
Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074
E: [EMAIL PROTECTED] -- --
--
From: Colin Anderson
[mailto:[EMAIL PROTECTED] Sent: Friday, February 17,
2006 12:36 PMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] A unique 'click to call'
project - Could usesome advice
Same
as before but instead of SIP as the originationchannel you pass
ZAP/g0/XXX (the DID of the agent) to your .call file. In fact, this is
exactly how the www.landmarkhomes.ca
script works (it calls the guy who entered his phone number in the website, when
he picks up, it calls the salesperson's cell number and the two are bridged
together)
The
drawback is, of course, that it uses 2 ZAP channels to bridge the call together,
but this isn't a problem I guess for you since you seem to have ZAP channels
coming out of your yinyang.
I have
an implementation in Active Server Pages (we are a MS shop) that I can send you
- it's suprisingly simple- but it could be easily modified for PHP or what
have you.
-Original Message-From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]Sent: Friday, February 17, 2006 9:56
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users] A unique 'click to call'
project - Could usesome advice
Colin,
Thanks for your assistance.
Reading over your advice I seem to still be a bit
confused.
My agents are not on the Asterisk server; it appears
in your advice that my the call will travel this path:
WWW interface -- agent enters their DID, platform
to use, and termination DID -- AST calls agent -- Agent calls
termination DID
If my agents are not on the Asterisk server (believe
me, I wish there were) :) how will this work?
I need a way to pass both the desired termination DID
and the origination DID.
Maybe I missed something
Thanks,
--
Christopher T. Aloi USA Datanet - Technical Support
Engineer 318 South Clinton Street Syracuse, NY 13202
C: (315) 569
4033 O: (315) 579 7074 E: [EMAIL PROTECTED] -- --
--
From: Colin Anderson
[mailto:[EMAIL PROTECTED] Sent: Friday, February
17, 2006 10:42 AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] A unique 'click to call'
project - Could usesome advice
You
create a context in your dialplan that accepts the DID to call as a variable
using the SetVar: syntax in your .call file. You then set up the context to
call your agent, and when they pick up, the context takes the variable you set
in your .call file asthe dialstring argument for a subsequent Dial().
Once the DID picks up, the calls are bridged together. Whatever web scripting
language you use writes the .call file, and you use POSTed arguments or
querystrings:
http://foo.com/call?context=MyContextAgent=SIP/DID=1551212
You
can see this in action at www.landmarkhomes.ca - click on any of
the pretty buttons that say "Call us now"
However, I have noticed that * 1.2.x will not wait for the caller to
pick up before executing the rest of the directives in the context- it
keeps executing regardless of the calling party's pickup status. Using * 1.0.x
the context will wait for the caller to pick up before placing the call to the
callee (i.e. executing the rest of the directives in the
context)
.call file (shortened to relevant)
Channel: SIP/ (if you are using SIP
phones)
SetVar: DID=XXX
Context: MyContext
[MyContext]
exten = s,1,Dial(ZAP/g0/${DID})
hth
-Original Message-From: Aloi, Christopher
[mailto:[EMAIL PROTECTED]Sent: Friday, February 17, 2006 8:07
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] A unique 'click to call'
project - Could use some advice
Hello List,
I work for an IP communication provider in upstate
NY as the engineer assisting our technical support team.
We provide a number of different Telco systems to
residential subscribers; and in an effort to more effectively trouble shoot
termination problems I came up with the idea of creating a click to call
system that will allow our agents to effortlessly place test
calls.
On a daily basis we place numerous (50-100) 'test'
calls to various locations in the US; these 'test' calls are routed using
one of three different phone systems:
1) The PSTN
2) Broadband phone platform one
3) Broadband phone platform two
I have an Asterisk server configured that can
terminate out three platforms listed above.
Re: [Asterisk-Users] Festival and Asterisk - different voices?
Michael Collins wrote: Just curious to know if anyone uses Festival with * and whether or not you’ve got a different voice than the default. I’m looking at doing a commercial application but my boss doesn’t want to shell out the $ before we do some real world testing of * and Festival. Specifically, I’m looking for a female voice, preferably US English. You can change the voice by editing the asterisk function. I think you want 'voice_cmu_us_slt_arctic_hts': ;;; Command for Asterisk begin (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. ; different voices, uncomment the one you want: ;(voice_cmu_us_awb_arctic_hts) ;(voice_cmu_us_bdl_arctic_hts) ;(voice_cmu_us_jmk_arctic_hts) (voice_cmu_us_slt_arctic_hts) ;uk voices ;(voice_kal_diphone) ;(voice_ked_diphone) (utt.send.wave.client (utt.wave.resample (utt.wave.rescale (utt.synth (eval (list 'Utterance 'Text string))) 5) 8000))) ;;; Command for Asterisk end ___ --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] ARI 0.06
On 2/17/06, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi ! I always use your ARI through AAH, and indeed nice job ! A few comment : - I have seen that we could use ARI only for the Call Monitor by setting a value. would it be possible to do the same for only Voicemail ... indeed, we are using Asterisk only for Voicemail, and this would be so good only to present this tab to people ... ( And in Settings page also, hiding the Call Monitor Settings part here too ) - Same for help ( to show it or not ) I have installed it on our AAH 1.3 version and here are the error messages I get : Call Monitor Page (Only the first message on each page shows the Play link): Warning: is_dir(): Stat failed for /var/lib/asterisk/bin/archive_recordings/ (errno=20 - Not a directory) in /var/www/html/recordings/includes/bootstrap.inc on line 113 Settings Page (Didn't try to apply new settings): Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 434 Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 473 Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 577 I hope you won't take these comments as critics, you are really doing a GREAT job ! Asterisk was really lacking this application part ! Thanks again, And all the best ! Jean-Marc On 2/17/06, Dan Littlejohn [EMAIL PROTECTED] wrote: ARI (Asterisk Recording Interface) has reached another milestone. The project is starting to become a full featured user portal and handle all the common errors that people seem to have. This release supports: call monitor page – new features include column sorting and filter small duration calls in addition to the ability to listen to call monitor recordings voicemail page – allows voicemail message listening and management handset feature code help page - I can never remember them all user settings web interface - that allows setting call fowarding, voicemail email and pager, voicemail password, and call monitor recording There are also alot of i18n translations now, although with all the rework of the code many are now somewhat broken and need to be updated. If you speak one of the following, email and I will send you the page to translate or updating to the appropriate ari.po page and returning it to me would be very helpful. German Greek Spanish French Hebrew Hungarian Italian Portuguese Swedish If you would like to translate ARI into another language, I would be happy to support it. Loaded into AMP CVS and also here: www.littlejohnconsulting.com?q=ari If you have a chance, take a look. Comments and suggestions are welcome. Dan 512.791.0137 www.littlejohnconsulting.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 Jean-Marc: Thanks for the feedback. I have addressed these issues they are available on my website and have been checked into AMP cvs. I have added a setting to the /recording/includes/main.conf file. $ARI_DISABLED_MODULES = ; allows for individual modules to be disabled (they are true modules though, and you can just delete them from the /recordings/modules directory) the is_dir error is a PHP bug. http://groups.google.com/group/mailing.www.php-dev/browse_frm/thread/1b5b94e775b70cdb/877e4406600a8121?lnk=stq=Warning%3A+is_dir()%3A+Stat+failed+for+errno%3D20+-+Not+a+directoryrnum=1hl=en#877e4406600a8121 But, I think I was able to suppress the error. The settings page errors have been corrected. Thanks; Dan 512.791.0137 www.littlejohnconsulting.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
Re: [Asterisk-Users] how to add stun functionality in asterisk
Yes Sir! This is what I use: http://www.vovida.org/applications/downloads/stun/ Works like a charm! Been running it in production for about a year. On 2/17/06, Deepak Dhiman [EMAIL PROTECTED] wrote: Hi friends ! I want to add stun functionality in asterisk. can anybody give me some hint that how can i start that. thanks in advance Deepak Dhiman ___ --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
