Re: [asterisk-users] Grandstream GXW-4004
hin lee wrote: I am consider replacing my TDM card for a FXS gateway. Anyone has any issues with the Grandstream GXW-4004 on Asterisk? I would like some feedback before I spend the $$ this device. It is difficult to be objective, I did not have any problem with asterisk compatibility itsellf, and it just works. Do not expect too much from this gateway, tough it is probably the cheapest four port gw around. It has some built-in PBX functionality you will have to bypass. It seems to be quite sensitive to ESD (or I had very bad luck during a storm). As I remember, some parameters for the PSTN side cannot be configured per-port, so keep that in mind that if you have lines from different telcos. A TDM card sounds more like a safer investment for me. -- Ivn Stepaniuk Alba Fotnica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Asterisk and SPA-3000
Hello everybody, I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used as PSTN gateway to asterisk in a small office. Everything works just fine, except that sometimes, and it seems that only for long incoming calls, the IVR menu appears on the middle of the call(like a three way call, call goes on with prompts playing over the parties). Dialing an extension at the prompt at that time actually works but disconnects the original extension (and transfers the PSTN leg to the new extension as normally). At the CLI there is nothing but a new incoming call from the SPA, exactly as the original call. It seems to happen with both asterisk 1.2 and 1.4, I am quite lost, Does anyone know what could be causing this problem? -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Asterisk and SPA-3000
Andrew Hakman wrote:
The SPA-3000 is notorious for falsely detecting DTMF tones in regular
voice, and when it thinks it hears DTMF...
Thanks Andrew. This is true, we do have false DTMF detections/playback
(Lowering RX gain really helps on this). However this does not seem to
be related. The only way to get to the IVR is through the 's' extension
on the context in which the SPA is registered. The SPA has a built-in
dialplan that is set to something like (S0:s). Asterisk context
shows is the following:
[sip_pstn_linksys]
exten = s,1,NoOp(Caller-ID-number: ${CALLERID(number)})
exten = s,n,NoOp(Caller-ID-name: ${CALLERID(number)})
exten = s,n,Set(CALLERID(name)=Externa SPA PSTN)
exten = s,n,GotoIf($[${CALLERID(num)} = pstn1]?private,1)
exten = s,n,Goto(menu,s,1)
exten = private,1,Set(CALLERID(number)=Privado)
exten = private,n,Goto(menu,s,1)
--
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com
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Re: [asterisk-users] Silent Dialing
'core show application dial' should give you an idea of what to play around with... In a similar scenario, once I used the 'm' option, with a special moh class. The moh class had some soft ticking sound because the remote system was not correctly indicating ringing, and sometimes delayed the audio ringing tone too much. This sorts of comforts users with a something-is-going on feedback sound, without having a double tone sequence. I don't know if it's the right way, it worked for me. I actually prefer to have two ringback tones. Darryl Dunkin wrote: Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extens...@remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the users. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gradstream Budge Tone-201
bilal ghayyad wrote: Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? We use a couple of BT200 (BT201 with additional ethernet port) for two years now with asterisk, no problems so far, probably you have a bad one. Just what comes to my mind: Cons: 7seg numbers only display! The display is difficult to see at common desk angle Speakerphone could be better Default ringtones are horrible, (you can change them with a little bit of work tough) There isnt any kind of built-in dialplan, you must use an overlapped dialplan (484 response) Pros: Cheap! Audio is OK, no noises or hiss. Looks like a phone, good buttons. Some phones are just too fancy. Headset plug. IMHO, good price/quality relation, dont expect too much from one of the cheapest SIP phones in the market. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Software Data Modem
Cherif wrote: I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? What is 'software data modem' for you? Iaxmodem would do the modem job, then on the serial device it creates you can hook up whatever you need (hylafax, pppd, mgetty, your own software...) -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working
Joseph wrote: I always had a problem with SIP and DTMF, I'm using old sipura adapters and have one digium iaxy FXS unit which works almost perfectly, never had any problem with DTMF on this unit. However, all phones connected to Sipura don't work very well especially when I setup speed dialing numbers to do banking, entering account number etc. (sipura units are set to dtmf=auto) DTMF Detection is done at your ATA, asterisk has nothing to do with that. Try the different settings on your sipura unit to improve DTMF detection, ie: relaxed dtmf, rx/tx gain, impedance. I don't remember exaclty but you don't need auto if you are certain that your SIP endpoint is either info or rfc2833. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customising Firmware
Dan Journo wrote: Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. You can leave the units with the factory firmware and hire the Mafia to keep your customers from changing provider. Not sure about costs tough. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The City of Amsterdam has been deploying asterisk throughout the city!
Ron Arts wrote: If you're interested, here is the press release: http://www.neonova.nl/nl/content/press/?tid=129735 This list is for non-commercial discussion. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX to SIP
George Farris wrote: I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs very well. On that machine I have a SIP phone. I have configured a netgear wgt634u with asterisk and a SIP phone and linked the two systems together via IAX. Audio from Ubuntu to netgear is not bad, audio from netgear to ubuntu is unintelligible. Any clues as to whether this will work? Configuration suggestions? Is a 200MHz arm processor just too small? what codec are you using? -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN to SIP line ratio
Shahnawaz Mir wrote: I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. I don't think there is an answer, it really depends on what your 200 users are going to do. It is not the same if half of your 200 users are sales reps, or if you are setting up 200 extensions for a school where the calls are going to be mostly internal... -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS to SIP gateway
jonas kellens wrote: Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? I hope to see more replies because I was in your situation some years ago. I'm far from an expert, but in my experience, at _that_ price range you don't have a lot of products to choose from, the Cisco SPA3102 is similar to what you are describing (Plus it's also a PSTN GW). Of this kind, I used GXW-4004 and Linksys PAP2, SPA3000 (the Cisco 3102 predecesor), also some larger AddPac ATAs (www.addpac.com) with excellent results, all of them have their pros and cons. The Digium TDM410 cards and they have a very good price/quality relation, plus they are intended for asterisk, plus you are supporting asterisk development. For an ATA (FXS only) list you can check http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
listm...@websage.ca wrote: On Sat, 10 Oct 2009 18:02:04 -0700 listm...@websage.ca wrote: On Sun, 11 Oct 2009 02:11:47 +0200 Ivan Stepaniuk i...@albafotonica.com wrote: listm...@websage.ca wrote: On the LAN side I can see the INVITE and OKAY messages which end with a CANCEL, apparently initiated by the Asterisk gateway. On the WAN side I can see that my Asterisk gateway is repeatedly sending OKAY messages in response to the INVITE from my ITSP. I assume the trouble is that these messages are either not getting back to my provider or something is blocking the confirmation from them. This more or less confirms what was seen in the sip debug trace as well. Post that SIP message from the CLI (sip debug), try adding externip=XXX.XXX.XXX.XXX (your external/public IP address) to your sip.conf global section, asterisk may be including it's private address in the OKAY sent to your provider. Here's the last message in sip debug before it gives up: ... Retransmitting #6 (no NAT) to 66.51.127.173:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bKee03.6a63373.0;received=66.51.127.173 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpae7KcSeB2Bte Record-Route: sip:66.51.127.173;lr;ftag=9Z5N4eayXp3Qm From: 2508864577 sip:2508864...@66.51.127.163;tag=9Z5N4eayXp3Qm To: sip:12504129...@66.51.127.173;tag=as32af6364 Call-ID: b734bd26-2fae-122d-91a5-653b331e358a CSeq: 121440337 INVITE User-Agent: Asterisk PBX 1.6.0.15 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: sip:12504129...@96.50.76.138 Content-Type: application/sdp Content-Length: 262 v=0 o=root 992672626 992672626 IN IP4 96.50.76.138 s=Asterisk PBX 1.6.0.15 c=IN IP4 96.50.76.138 t=0 0 m=audio 15550 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt). Scheduling destruction of SIP dialog '4e8ef1b977bf0e062212334634080...@192.168.11.1' in 6400 ms (Method: INVITE) ... 66.51.127.173 is my provider's SIP server 66.51.127.163 is my provider's RTP server I even check DNS to make sure both forward and reverse records jive. Externip was a good suggestion, and worth a try, though because I'm registering with my provider and using dynamic=yes, wouldn't they just reply to that anyway, especially given that the registration works fine? Anyway, after adding externip=my-external-ip to [general] and doing a sip reload in the console the problem remains... [Bumping this in the hope that someone might have some new insight or suggestions since I posted this on a holiday weekend (in my part of the world anyway)...] I was staring at the SIP transcript and I don't see anything wrong, I'm out of suggestions, except that you could analyze and compare the packets when your phone is connected directly (if it's physically possible). I hope someone throws some light over this. -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - busy tone detection
Asterisk wrote: Hi guys, My Asterisk box is connected to a VoIP provider using SIP protocol. Unforunately, when I try to call a number that is busy, the provider plays busy tones for 15 seconds and than hungups the call with dialstatus=CHANUNAVAIL, instead of dialstatus=BUSY. Is there any way to detect busy tone pattern on a SIP call (are there any SIP equvalents of busydetect and busycount parameters)? Thanks, Alex AFAIK this is not possible, yet. See this post, about the same topic: http://archives.free.net.ph/message/20090703.122514.65031d43.en.html -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD ASR
B.Masoud @ SH wrote: Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source destination? You asked this same question two weeks ago with the same subject. You got at least 5 answers. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Dan Journo wrote: Thanks for that. I really appreciate it! Dan As pointed by the follow-ups, note that the recordings are not taken from the monitor but from an upload folder inside, the dialplan takes care to move there the files for the ended files only issuing a 'System' command. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Steve Edwards wrote: On Tue, 13 Oct 2009, Dan Journo wrote: To avoid the problem of deleting/copying calls that are still being recorded, I could record the call into a temp directory. Then using the dial plan, I could copy the temp recording into the ftp root directory once the call has ended. True, but if you need to execute a process at the end of the call, why not make it an AGI and hide all the ugly details and keep your dialplan nice and clean and shiny and maintainable? Your recordings will be instantly available and the correct operation of your system does not depend on an externally scheduled external process involving clear-text passwords and obscure packages. Your successor will thank you :) This is true, doing everything from inside an AGI script would be nicer, the ugly part comes if you are tied to an old and ugly FTP server, specially if its from a hosting provider that limits the connection count to 2, or so. AGI+sshfs/scp/nfs/whateverfs... would be much more cleanshiny (tm). -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] Asterisk to Asterisk access voicemail - not working
Joseph wrote: On 10/12/09 13:29, Ivan Stepaniuk wrote: Check the relaxed dtmf setting on yhe linksys, and also check the Impedance, Rx and Tx gain, As well as the DTMF duration, All this knobs can mess up your DTMF tones if there is something wrong. Just Solved! The problem was caused by echo, changing impedance to 900 and decreasing SPA to PSTN and PSTN to SPA gain to -3 solved the problem. Glad to know it's working, impedance mismatch at the hybrid can mess everything up. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working
Joseph wrote: I just double checked the setting of the remote asterisk and it has the same setting as mine. Sip.conf has in Global: dtmfmode = rfc2833 individual extension has no dtmf setting at all, so global setting take precedence. All units Linksys, Sipura have DTMF Tx Method: Auto Linksys has an additional setting: DTMF Tx Mode: Strict My asterisk is using old Sipura units and dtmf tones to access voicemail are recognized. The remote asterisk is using newer Linksys units and dtmf to voicemail does not work, the phone hangs up. The strange part is: PSTN -- Asterisk (voicemail access) works OK on both sytemes. Asterisk (w/Linksys) -- Asterisk (w/Sipura) to Voicemail works OK Asterisk (w/Linksys) -- Asterisk (w/Linksys) to Voicemail DOES NOT work Asterisk (w/Sipura) -- Asterisk (w/Linksys) to Voicemail DOES NOT work You are using the ATAs to access from one Asterisk to the other one? Wouldn't make sense to connect those two asterisk through SIP or IAX via Internet instead of calling via PSTN anyway? Anyway, In this case it seems that this is not asterisk related, try the several Sipura/Linksys settings related to DTMF, also if the asterisk boxes are the only thing your ATA connects to, there is no point on using the auto mode. Check the relaxed dtmf setting on yhe linksys, and also check the Impedance, Rx and Tx gain, As well as the DTMF duration, All this knobs can mess up your DTMF tones if there is something wrong. Just -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Dan Journo wrote: I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? As said by others, there is no such built-in capability I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. It is really important for you that the recordings are available asap on your FTP destination? I had a similar task and I found problematic to upload the files from inside the dialplan, Is not that it can't be done, but if the FTP is slow and your number of connections limited, you may run into a problem with simultaneous calls ending and asterisk trying to upload 20 files at the same time. In my case the files could be uploaded every hour, so I made a simple bash script that uploads the new files to the FTP using 'curlftpfs', a nice command that mounts the remote FTP on a local mount point using FUSE, then the script just moves the files from the local folder to the FTP, and voila. Asterisk just takes care of moving recordings that ended to the desired path. I can post the bash script if you are interested. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Dan Journo wrote: Thank you for replying. I hadn't thought about the problem of simultaneous calls. It would be a problem if a number of calls ended at the same time. If you can post it, the script would really be helpful as I'm only a beginner with Linux The script is very simple and far from complete, it just moves the content into the mounted FTP directory. It has some verbose output as it is run from inside another script that redirects the output to a log file. The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. You have to set up a cron so the script is run every hour, normally putting the script or a link to it inside '/etc/cron.hourly' is enough. There is also a /etc/crontab file you can use to setup something more complicated if needed (ie: runing it every 2 hours, running at tea time...). read 'man cron', and 'man crontab'. You also need to install the magic part, 'curlftps'. in Debian that's the name of the package too. I use version curlftpfs 0.9.1 libcurl/7.18.2 fuse/2.5 Be careful that in *nix, file.WAV and file.wav are different files. Here is the script: #!/bin/sh MOUNT_POINT=/mnt/remote_ftp FTP_HOST=www.ftphost.com/htdocs/recordings FTP_USER=ftpusername:difficultpassword RECORDINGS=/var/spool/asterisk/monitor/upload echo Starting upload `date` echo Connecting to $FTP_HOST... curlftpfs -o user=$FTP_USER $FTP_HOST $MOUNT_POINT echo Transfering `du -hc $RECORDINGS/*.wav | grep total` ... mv -vf $RECORDINGS/*.wav $MOUNT_POINT echo Disconnecting. umount $MOUNT_POINT exit 0 -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lawnmower man attack sip tag=Zerogij34 some one else notice this in 20th september or recently?
Marco Mouta wrote: Sad to say, but I believe this is only the small beginning…. Just a guess, and off-topic, but probably someone got very angry at citibank. At least in Spain, they (or a marketing contractor) seem to have called every single mobile phone in this country, they called me five times without even knowing I was already a customer. I bet they have the same marketing policies everywhere. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
John Novack wrote:
B.Masoud @ SH wrote:
I use elastix,
I have this for dialout:
exten = s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
where should I add the w ??
right before the dialed number
If I understand your code it should be:
exten = s,8,Dial(www${OUT_${ARG1}}/${ARG2:${length}})
John I think you are wrong, I don't know elastix but the OUT_${ARG1} var
seems to contain the channel technology, the 'w' should be inserted
after the slash.
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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Re: [asterisk-users] No sound on voicemail from analog line
Landy Landy wrote: Do you mean that incoming calls on your PSTN line works as they should, but not when they reach the voicemail? or that incomming calls on PSTN are always mute? Incoming calls on PSTN line work as they should but, when someone leaves a voicemail message the messege is mute. When I try to retrieve the messeges I get the prompt that says how many messeages are there Post the relevant part of your extensions.conf, * version, CLI output when the caller leaves a message and when you retrieve the message. -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
I have done the changes
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I am getting this:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Is it now on work? Or I have to restart?
It is not working. Issue an 'extensions reload' command at the asterisk
CLI and try again. If it still does not work, then you have edited the
wrong Dial. You should have tried that before asking in the list again.
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote: Sorry for keep asking, but I did extensions reload, and restarted asterisk, What should the message looks like? I still get the same: -- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592, DAHDI/r0/0559857826|300|) in new stack -- Called r0/0559857826 You should edit the 'Dial' command inside the macro-dialout-trunk context, the line probably starts with exten = s,19,Dial(... as stated in the CLI snippet you've posted, (s...@macro-dialout-trunk:19). -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working
Joseph wrote: I've tried setting my asterisk dtmf to rfc2833, inband it is not working. The other Asterisk Linksys is set dtmf = auto If understand correctly, you have two asterisk servers and when you dial from one the other, DTMF is not recognized. I also asume you are using SIP to connect them as you mentioned dtmfmode. In any case, this should be set to the same value on both sides, both rfc2833, or both info. You don't wand inband and auto is just rfc2833 with automatic inband fallback. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
listm...@websage.ca wrote: After running for months without issue I've got a situation where incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected. The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing calls can be made successfully and no error messages or warnings are reported by Asterisk. However, incoming calls appear to enter my dialplan as desired and go so far as to start ringing my SIP phone (Grandstream GXP-2000) but drop after two rings. The caller gets a busy tone and that's it. If I answer the call before the two rings I just get a moment of dead air and it drops in the same way. In the asterisk console (and log file) I see these messages at the fail point: [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt) Okay, so I verified that my firewall is properly accepting traffic on the range of SIP and RTP ports as specified by my ITSP. After sending them a sip debug trace my provider said this: It appears that your machine is not receiving replies when it tries to acknowledge the incoming call back to our server. This could be a firewall issue or potentially something else that changed without your knowledge. Furthermore, they suggested I might try registering and connecting directly to their Asterisk using only the Grandstream phone. I tried this and...surprise! Both inbound and outbound calls work fine but leave me without voicemail or any other services my PBX would be providing. Right, so now I'm thinking there must be something wrong with my Asterisk configuration yet I've made no config changes that would account for the sudden (and consistent) incoming call failures. Here's the relevant portions of my sip.conf if it helps (with credentials and ips replaced by Xs): [general] alwaysauthreject=yes dtmfmode=auto disallow=all allow=ulaw register = :x...@xxx.xxx.xxx.xxx:5060 register = :x...@xxx.xxx.xxx.xxx:5060 [101] type=friend context=websage host=dynamic deny=0.0.0.0/0 permit=XXX.XXX.XXX.XXX/24 qualify=yes secret= mailbox=...@default accountcode=101 Does your asterisk server have two network interfaces, one with a private IP address and another one with the public one? Did you try adding canreinvite=no to your 101 friend sip entry? What does the SIP debug say? -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
listm...@websage.ca wrote: On the LAN side I can see the INVITE and OKAY messages which end with a CANCEL, apparently initiated by the Asterisk gateway. On the WAN side I can see that my Asterisk gateway is repeatedly sending OKAY messages in response to the INVITE from my ITSP. I assume the trouble is that these messages are either not getting back to my provider or something is blocking the confirmation from them. This more or less confirms what was seen in the sip debug trace as well. Post that SIP message from the CLI (sip debug), try adding externip=XXX.XXX.XXX.XXX (your external/public IP address) to your sip.conf global section, asterisk may be including it's private address in the OKAY sent to your provider. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing sip calls work; incoming calls fail
Ivan Stepaniuk wrote: listm...@websage.ca wrote: On the LAN side I can see the INVITE and OKAY messages which end with a CANCEL, apparently initiated by the Asterisk gateway. On the WAN side I can see that my Asterisk gateway is repeatedly sending OKAY messages in response to the INVITE from my ITSP. I assume the trouble is that these messages are either not getting back to my provider or something is blocking the confirmation from them. This more or less confirms what was seen in the sip debug trace as well. Post that SIP message from the CLI (sip debug), try adding externip=XXX.XXX.XXX.XXX (your external/public IP address) to your sip.conf global section, asterisk may be including it's private address in the OKAY sent to your provider. Sorry, general section. not global. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No sound on voicemail from analog line
Landy Landy wrote: Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound Do you mean that incoming calls on your PSTN line works as they should, but not when they reach the voicemail? or that incomming calls on PSTN are always mute? -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limiting number of channels to be accessed
B.Masoud @ SH wrote: I want to grand the first asterisk box to use all the 24 channels on the main, but I want the 2^nd asterisk to use only 8 port, how can limit the second box from receiving more than 8 simultaneous calls?? (even if the main have available ports) This can be done using the GROUP functions under asterisk. Check this, look at example #2: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VPS Server
David @ULC wrote: Looking for Genuine VPS Server for 250 ports on Rent. Ask on the biz list instead. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem sending a DTMF remotely. Please need help!!
Pablo Bernasconi wrote: My Asterisk version is 1.6.0.15, but I`ve tried it in 1.6.0.6 and 1.6.1.6 version and the same happens. I dont know what I am missing... Please help me. Pablo, I did not answer in the first place because I am not completely sure, but just guessing, PlayDTMF just generates DTMF sounds, inband, and not info/rfc2833 messages, that's probably why you hear the tones but nothing happens. Just my two cents, perhaps it throws some light. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF problems during a message play
Barton Fisher wrote: I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the message to stop before the press is hear. The problem could be at your provider's system, perhaps you should try to submit a support ticket. I know Vitelity had different gateways with different DTMF configurations. Also... it would be too much bad luck, but I had a bad chinese phone phone with a bad -row- in the DTMF keyboard, it worked with the telco as the detection was more relaxed, but not with our old PBX. dtmfmode=rfc2833 disallow=all allow=ulaw Over ulaw you could try to use inband, or auto... your millage may vary. -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with inbound calls - asterisk 1.6.1.6
Carlo Dimaggio wrote: I have a new installation with asterisk 1.6.1.6 but I'm unable to receive calls from a SIP trunk: [Oct 2 14:30:09] NOTICE[21554]: chan_sip.c:18523 handle_request_invite: Call from 'user001' to extension 'user001' rejected because extension not found. You are getting calls for an 'user01' extension instead of the 's' extension. I don't know about your trunk but having in your context exten = user001,1,... would receive that call. -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having trouble with IF and blanks
Richard Kenner wrote: How do I properly quote things when I want to use the IF function on something returning a string with blanks (e.g, CALLERID(name))? Use double quotes around your variable See: http://www.voip-info.org/wiki/view/Asterisk+Expressions -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan problem
Anahi Ludueña wrote: Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten = 2001,1,Answer exten = 2001,n,Dial(local/3005) exten = 2001,n,Hangup exten = 3005,1,Set(__RINGTIMER=10) exten = 3005,n,Macro(exten-vm,novm,3005) exten = 3005,n,Hangup Why would you do that anyway? use 'Goto(3005,1)' instead of 'Dial' Saludos -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Concept
B.Masoud @ SH wrote: China too wide, but regardless! How is asterisk take care such situation? -regardless- I think I pointed it out already, please do some research, simply googling for asterisk + reinvite leads you here: http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Integration with RDP Property Management Software?
What you are looking for is probably a HOBIC (Hotel Billing Information Center) interface for asterisk, there was a similar thread in this list five years ago, I came across this some time ago and I did not find a working implementation. The links are dead, but you may found it of interest: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg39090.html PS: Please use your mail client New/Compose button when creating a new thread, do no reply an existing different one. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Concept
B.Masoud @ SH wrote: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN-UK + UK-China (around 1000ms !) or only from Japan to China??? In the case of the SIP protocol, the audio (RTP) traffic can be re-routed on the fly from A(jp) to C(ch), reducing the audio latency, (and sometimes increasing your headaches). This is calling re-INVITE, and can be turned on on asterisk. For other protocols there are similar features. I think your latency figures are a little bit exaggerated if you speak about the network latency. I am in Spain and my latency to China at my home ADSL is arround 80ms for mainland. 250ms to Tokio tough. Regards -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish a Meetme
Anahi Ludueña wrote: Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, From the asterisk cli 'core show application MeetMe': User can exit the conference by hangup, or if the 'p' option is specified, by pressing '#'. Then the dialplan resumes, but why would you need to kick that user from the MeetMe? AFAIK there is no -easy- way to automatically kick out the last user from the conference when it is the only one left. What are you using MeetMe for? Saludos -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kill sip user
Bayardo Sanchez wrote: I have a user but I need to give that user only kill and disable all connection cut calls what is the command in the CLI Please rephrase your question. I've just read your message 5 times and I still don't understand what do you want to do. Regards. PS: A 15+ line signature for a 2 line message is likely to upset many people on any mailing list. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
Jeff LaCoursiere wrote: It would be really cool if iaxmodem would actually answer an incoming modem call and pass traffic to something like pppd. For those times when the pstn link is up, but something is wrong with the 'net connection... I have a working setup of what you describe. I can dial the good ol' way into my * box if the IP link is down, just set an iaxmodem device in your inittab as a serial console. pppd is not needed if you just need to access the linux shell. Putting a password on the extension would be a good idea though. -- Ivan Stepaniuk Alba Fotonica S.L. http://www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which is best provider for G.729
silent sayz wrote: But i am only wondering which is best :( Howler gives trial G.729 licenses for free on their website. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk Sequence
B.Masoud @ SH wrote: If the call was busy on the first trunk it will go to the I dont understand what you mean. second (i.e. the called party hung-up without answering the call) How can the called party hungup without answering? PS: Please don't use the Reply button when starting a new thread! -- Iván Stepaniuk Alba Fotónica S.L. http://www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 for Asterisk
silent sayz wrote: why we need it? IMHO, Actually we don't. Why people use G.729 codec with asterisk? Because it has a very good bandwidth/quality relation. Or because you need to inter-operate with another system based on this codec. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESET CDR
B.Masoud @ SH wrote: Yes that is the problem. So what do you do when the line doesn't support polarity?? What is the best solution in this case? What kind of gateway do you use to connect to the PSTN? -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASR ACD
B.Masoud @ SH wrote: ASR: Average Success Rate ACD: Average Call duration Simply having your CDR on an SQL database would do the trick. For example, you can create a query that averages the value of your duration column. Anyway, there are many applications that can be used to analyze your CDR and even make charts on that. Google for: asterisk cdr analyzer. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call getting stucked !!
David @ULC wrote: I also have an access to VOIPSwitch ver 2 where I can see live calls. Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch) Perhaps you should try to describe your setup a little bit. How do you know the problem is on the Asterisk side? What do you mean with calls getting stuck anyway? Do you have PSTN gateways? What does the Asterisk CLI say? Regards -- Ivan Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call getting stucked !!
David @ULC wrote: I don't know where is the problem. May be with VOIPSwitch OR may be with Asterisk.. Call getting stuck : My agent hang up the call but in Active calls , I see call connected and getting charged I use VOIP and NOT PSTN With the information you gave, the problem could be actually anywhere. Your phones/soft-phones, your Asterisk setup, your VOIPSwitch, your VOIP provider... Didnt check the Asterisk CLI. Can I get any history of what asterisk REALLY had ? Depending on your configuration (/etc/asterisk/logger.conf), you may have a /var/log/asterisk/messages file. Probably, by default it does not contain the information you need to track down the problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESET CDR
Todd Routhier wrote: billsecs is a field in the CDR, it's already there.. Just don't bill based on the duration field, bill based on the billsecs field and you should have what you want. He says that the line does not support polarity reversal. It really depends on the type of PSTN interface he is using, but this probably means that the duration and the billsecs fields are going to be the same, as the channel gets answered and the ring-back tone is also counted. I would rather say that if you do not have a proper way to detect the call progress, the system is not reliable for billing your users. The main problem with the 30 seconds solution is that you will have a lot of calls made without charge (ie: 3 seconds ringing, 20 seconds call). And you will also charge people for ringing a phone during more that 30 seconds (ie: calling my grandma) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call getting stucked !!
David @ULC wrote: I see this : /etc/asterisk/logger.conf [logfiles] console = notice,warning,error messages = notice,warning,error,debug,verbose Then you should have a /var/log/asterisk/messages with detailed information on what happens when your agent hangs up. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] streaming meetme conference
Boehm, Matthew wrote: 1. icecast only supports OGG/Vorbis and neither iTunes nor WMP can play that format natively. AFAIK, icecast also supports mp3 streaming too, you could create an m3u playlist with the stream path, that would be enough to trigger your users' WMP. You could also embed the WMP object (I hate that so much!). My two (sightly off-topic) cents. -- Ivan Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo and Playtones not working on SIP after upgrade
Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten = 1003,1,Wait(1) exten = 1003,n,Playtones(!1050/1000) exten = 1003,n,Wait(1) exten = 1003,n,StopPlaytones exten = 1003,n,Echo exten = 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP channels. Over IAX2, both Echo and Playtones works fine on this same extension and system! I googled and tried several things, but nothing seems to work. Basically the log shows it is working, there are no errors or warnings, but there is no sound at all. No beeps, no Echo. Calls, voicemail, moh, and everything else we are using works just fine. We are using Asterisk 1.4.21.2~dfsg-3 (on debian stable), SIP channels with both grandstream and soft phones. Everything on the same network segment. Codec does not seem to affect this behavior (tried them all) Any clues? Thanks! -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SOLVED Re: Echo and Playtones not working on SIP after upgrade
Tilghman Lesher wrote: On Monday 07 September 2009 09:36:53 Ivan Stepaniuk wrote: I had the following echo-test extension on my Asterisk 1.2 setup. exten = 1003,1,Wait(1) exten = 1003,n,Playtones(!1050/1000) exten = 1003,n,Wait(1) exten = 1003,n,StopPlaytones exten = 1003,n,Echo exten = 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP channels. Try adding an Answer() in there, before the first Playtones. That made the trick, thank you very much. I wonder why does it work on IAX2 channels but not on SIP channels without the Answer command. Anyway, I guess that answering the channel first is the right thing to do. -- Iván Stepaniuk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telephony that just works
On Mon, 2005-10-10 at 13:28 +0200, lenz wrote: I am looking for a way to have multiple remote Windows users download a package and get connected to *. My idea would be that they run a simple app, it connects without any setting to an * box (maybe via IAX) and then people press a button to talk. It would be okay if they had to enter a username and password, but not more than that. i've tried IaxComm http://iaxclient.sourceforge.net/iaxcomm/ it works, it's iax, and it's open source so you can re-package - re-compile it with you own default settings (or even hide those settings you don't want final users to see) -- Ivan Stepaniuk [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
