[asterisk-users] More dialplan visualization (neat graphs!)
Howdy all, The Asterisk-Java project has included some rudimentary parsing related to dialplans and extensions.conf. I've done a blog post at http://asterisk-java.org/ related to it, and giving a demo of some dialplan visualizations. It could eventually get fleshed out into an open-source visual diaplan designer or visualizer. The web start demo requires Java 6. I'd love your feedback. Thanks, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 ___ -- 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] Queue Stats
excellent contribution to the asterisk community andy congratulations Nicolas rickygm ... 2008/5/16 Nicolás Gudiño <[EMAIL PROTECTED]>: > Hello, > > I have finally released the queue stats package to the public.. please go to: > > http://www.asternic.org/stats > > To get it or see the online demo. > > -- > Nicolás Gudiño > Buenos Aires - Argentina > ___ -- 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] update DB on ringing/ catch ringing event
Update on this one.
I finally went back to AMI only for implementing this particular feature, but
ofcourse I had to make an addition of a couple of lines for my particular
requirement.
On "Dial", the 'dial' event is sent over AMI which I capture. Unfortunately the
event didn't have any field identifying the account/or other user settable data
for that particular call. So, I added lines in app_dial.c to send even the CDR
userfield in the event.
So, before doing the 'Dial' I set CDR userfield with my own data, which is
captured by the AMI user and populates/updates the correct row in my DB with
the dialed channel, etc.
>From this point on, I can hangup the required channel, even before it has been
>answered/ even before it has started ringing.
static void senddialevent(struct ast_channel *src, struct ast_channel *dst)
{
manager_event(EVENT_FLAG_CALL, "Dial",
"Source: %s\r\n"
"Destination: %s\r\n"
"CallerID: %s\r\n"
"CallerIDName: %s\r\n"
"SrcUniqueID: %s\r\n"
"DestUniqueID: %s\r\n"
"CDRUserfield: %s\r\n",
src->name, dst->name, src->cid.cid_num ? src->cid.cid_num : "",
src->cid.cid_name ? src->cid.cid_name : "", src->uniqueid,
dst->uniqueid,
(dst->cdr)?(dst->cdr->userfield):"");
}
I am writing this mail from home, so don't really have the exact field names.
cheers
- Ben.
--- On Thu, 5/8/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> From: Tzafrir Cohen <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] update DB on ringing/ catch ringing event
> To: asterisk-users@lists.digium.com
> Date: Thursday, May 8, 2008, 12:00 AM
> On Thu, May 08, 2008 at 12:19:52AM +0300, Atis Lezdins
> wrote:
> > On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
> > <[EMAIL PROTECTED]> wrote:
> > > Benjamin Jacob schrieb:
> > >
> > >
> > > > Anyway in Asterisk to update a DB/ do some
> action on
> > > > events like ringing.
> > > > The issue is I need to be able to
> hangup/cancel a
> > > > call, if it's ringing(decided by the
> admin). This is
> > > > independant of the timeout that we can
> specify in the
> > > > Dial command.
> > > >
> > > > If I could somehow update a DB with the
> channel name
> > > > on ringing, it would solve my problem.
> > > >
> > > > I assume NVlinedetect is one way to do it,
> but that
> > > > isn't visible anymore, more so for
> Asterisk 1.4 and
> > > > above.
> > > >
> > > > Any bright ideas on this one?
> > >
> > > I think there is no other solution but to listen
> to events on
> > > the Asterisk manager interface.
> > >
> >
> > For now, not really.
> >
> > You could try Realtime Channels patch I just mentioned
> here:
> >
> http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html
> >
> > This should give you up-to-date list of channels in
> database, so you can use
> >
> > SELECT * FROM channels WHERE state="Ring";
> >
> > to get currently ringing channels.
> >
> > If You find this patch useful, please add a comment to
> issue
> > http://bugs.digium.com/view.php?id=12556
> > that you would like to see Realtime status implemented
> in future
> > versions of Asterisk.
>
> So you constantly poll the status of all channels? Waiting
> on manager
> interface event sounds more effective to me.
>
> But what exact "ringing" is it? Isn't the
> call by then already in the
> dialplan (and could be hung up before answered?)
>
> --
>Tzafrir Cohen
> icq#16849755 jabber:[EMAIL PROTECTED]
> +972-50-7952406 mailto:[EMAIL PROTECTED]
> http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] playing .gsm sounds through a web browser
On 16 May 2008, at 00:26, Julian Lyndon-Smith wrote: > I have a lot of recordings from asterisk in a .gsm format. I would > like > to play these files from a web browser (IE, firefox and opera) > > What do I need to do in order to achieve this goal ? > Sorry to catch up late on this, but I have a tiny Java Applet that does this. demo: http://www.westhawk.co.uk/software/playGSM/PlayGSM.html Source code included: http://www.westhawk.co.uk/software/playGSM/PlayGSM.jar Tim. ___ -- 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] CFP For HITBSecConf2008 - Malaysia Now Open
Hello from Malaysia! The Call for Papers (CFP) for the 6th Hack In The Box Security Conference in Malaysia (27th - 30th October 2008) is now open. We've got some really cool stuff lined up this year including an open-hack competition for charity, a third track in the conference (hitb-labs), 4 keynote speakers + 30 international experts, the usual team based capture the flag competition, a new wireless (bluetooth, rfid, 802.11) village and lock picking village! Summaries not exceeding 1250 words should be submitted (in plain text format) to cfp -at- hackinthebox.org for review and possible inclusion in the programme. Submissions are due no later than 30th of June 2008 TOPICS Topics of interest include, but are not limited to the following: # 3G/4G Cellular Networks # Apple / OS X security vulnerabilities # SS7/Backbone telephony networks # Analysis of network and security vulnerabilities # Firewall technologies # Intrusion detection # Data Recovery, Forensics and Incident Response # HSDPA and CDMA Security # Identification and Entity Authentication # Network Protocol and Analysis # Smart Card and Physical Security # Virus and Worms # WLAN, RFID and Bluetooth Security # Analysis of malicious code # Applications of cryptographic techniques # Analysis of attacks against networks and machines # File system security PLEASE NOTE: We do not accept product or vendor related pitches. If your talk involves an advertisement for a new product or service your company is offering, please do not submit. Your submission should include: # Name, title, address, email and phone/contact number # Short biography, qualification, occupation, achievement and affiliations (limit 250 words). # Summary or abstract for your presentation (limit 1250 words) # Technical requirements (video, internet, wireless, audio, etc.) Each non-resident speaker will receive accommodation for 2 nights/3 days. For each non-resident speaker, HITB will cover travel expenses up to USD 1,000.00. HITBSecConf2008 - Malaysia - Sponsorship Options For an opportunity to position your company as a major supporter of this event, we have several sponsorship packages which offers an extensive variety of direct and exclusive mechanisms for pre-event exposure and direct business generation during the event. If you are interested in further details regarding sponsorship of HITBSecConf2008 - Malaysia, please contact us. === On a related note, the keynote presentation videos from HITBSecConf2008 - Dubai is also now available for download from here: Day 1 Keynote: http://materials.hitbsecconf.org/hitbsecconf2008dubai/videos/Keynote-1.mov Day 2 Keynote: http://materials.hitbsecconf.org/hitbsecconf2008dubai/videos/Keynote-2.mov See you guys in October! The HITB Team. ___ -- 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 concurrent calls count
On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote: > On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote: > > It seems any constructive criticism offered, you take as an attack > > against Digium. That is not a good attitude. > > I dunno, Steve; I wouldn't call "Digium needs to 'man-up'" constructive > criticism, myself. I'd call it an ad-hominem. > > Tilghman *does* seem to be a bit of a cheerleader, but there's nothing > wrong with that... unless you're an *employee*, and you're going out of > your way to hide it. I'm been a member of this community far longer than I've worked for Digium, and even then, I form my own opinions and I call them as I see them. If I can't say something because of insider knowledge, I know well enough to keep my mouth shut, but this is not one of those times. And if there _is_ something wrong with the way Digium is doing something, I also am more than happy to put up a big fuss until it's fixed. I'm probably a bit of a loose cannon, but they knew that when they hired me. ;-) -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
On Fri, May 16, 2008 at 10:54:17AM -0400, Steve Totaro wrote: > I am very aware of this but do you think that the SoHo box (and it's > artificial cap) is maxing out no matter what the users are doing? If > so, then why can you just upgrade concurrent use via a CC and the > website? So, Steve are you saying that they're quoting conservatively, and you're *unhappy* with them for that? I think it's great, myself. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 concurrent calls count
On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote: > It seems any constructive criticism offered, you take as an attack > against Digium. That is not a good attitude. I dunno, Steve; I wouldn't call "Digium needs to 'man-up'" constructive criticism, myself. I'd call it an ad-hominem. Tilghman *does* seem to be a bit of a cheerleader, but there's nothing wrong with that... unless you're an *employee*, and you're going out of your way to hide it. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk peer definition with multiple host ip addresses
Hi guys, How would I configure a peer in sip.conf that has multiple host ip addresses? Can I just put multiple host=ip_address lines in the config file or will I have to create multiple peers each one with a different host ip address setting? A provider I am trying to hook up with gave me multiple ip addresses to accept calls from for incoming calls and multiple addresses that I can send calls too and I'm wondering what the best way to handle this is on my side? I will be working with Broadvox and they will be pushing calls to my server from a few different sources it looks like. Thanks, Tom ___ -- 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] Queue Stats
Hello, I have finally released the queue stats package to the public.. please go to: http://www.asternic.org/stats To get it or see the online demo. -- Nicolás Gudiño Buenos Aires - Argentina ___ -- 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 while running Flash Operator Panel
On Thu, May 15, 2008 at 7:57 AM, Sukhbir Singh <[EMAIL PROTECTED]> wrote: > Hi All, > > Whenever i try to start FOP using script > ./op_panel_redhat.sh start given in directory > /usr/local/op_panel-snapshot/init > > I got the following error: > > Starting Flash Operator Panel: execvp: No such file or directory > >[FAILED] > > Please let me know the reason for this. > Thanks in Advance > > With Regards, > newbie > You will have to edit op_panel_redhat.sh and change it so it reads: DAEMON=/usr/local/op_panel-snapshot/op_server.pl OPTIONS="-d" Then you will want to move the file to /etc/rc.d/init.d And add it to startup with: chkconfig --add op_panel_redhat.sh -- Nicolás Gudiño Buenos Aires - Argentina ___ -- 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] Newbie Asterisk: Install Asterisk as non-root
On Fri, May 16, 2008 at 3:04 AM, Lee, John (Sydney) <[EMAIL PROTECTED]> wrote: > First of all, thanks Philipp, Alan, Tzafrir and James for your valuable > comments. I have listed below the exact list of commands to run for > reinstalling asterisk 1.4.* as non-root on a Redhat / Fedora distro. > Hope others can benefit. > > I have the following comments/questions though: > 1) #What is safe_asterisk used for actually? I did not touch it in > my modification because I don't know when is it triggered? The safe_asterisk script monitors the actual asterisk process, and if it dies for some reason, it restarts it and optionally notifies you. It's just a precaution. MySQL is often run under a script called mysqld_safe for the same reason. > 2) #I do not actually know whether we really need to modify > /etc/asterisk/asterisk.conf? Is this file read by asterisk at all? > Seems like an important file name - asterisk.conf? It is read by asterisk, but whether you need to change any of the defaults really depends on your environment. Most of the options in it have equivalent command-line options, so you might want to use asterisk.conf instead of modifying the startup script (which could be overwritten the next time you upgrade). > 4) There is an additional chmod to run for letting voicemail.conf to be > written by group asterisk. What I found was that /etc/asterisk also needs to be writable by the asterisk user, because asterisk will unlink and recreate the file, so it needs to be able to write to the directory, not just the file. You can protect yourself a little bit by setting the sticky bit on /etc/asterisk, so even if asterisk goes nuts, it can't whack files it doesn't actually have write permissions on. chmod g+w /etc/asterisk/voicemail.conf chmod g+w,+t /etc/asterisk -James ___ -- 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] Fetching Binary data from SQL Server
Tilghmanm, Thanks a lot, I have changed the value in FREETDS and it worked. Regards, Sanjay Rajdev - Original Message - From: "Tilghman Lesher" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, May 17, 2008 4:09:43 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Fetching Binary data from SQL Server On Friday 16 May 2008 16:10:22 Sanjay Rajdev wrote: > I am trying to write a customized app using C that would fetch voice file > from SQL Server 2000 using ODBC and FREETDS. > > Currently I am only able to fetch first 63 KB chunk from the DB, and not > able to fetch the rest of the file, below is the code that i am using to do > so, Actually, if you Google, you'll find that in freetds.conf, the default 'text size' parameter is set to exactly 64512, which is the limit that FreeTDS itself is placing on the data. You might try increasing that (to a maximum of 2GB) and see if that works better for you. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Fetching Binary data from SQL Server
On Friday 16 May 2008 16:10:22 Sanjay Rajdev wrote: > I am trying to write a customized app using C that would fetch voice file > from SQL Server 2000 using ODBC and FREETDS. > > Currently I am only able to fetch first 63 KB chunk from the DB, and not > able to fetch the rest of the file, below is the code that i am using to do > so, Actually, if you Google, you'll find that in freetds.conf, the default 'text size' parameter is set to exactly 64512, which is the limit that FreeTDS itself is placing on the data. You might try increasing that (to a maximum of 2GB) and see if that works better for you. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fetching Binary data from SQL Server
On Friday 16 May 2008 16:10:22 Sanjay Rajdev wrote:
> I am trying to write a customized app using C that would fetch voice file
> from SQL Server 2000 using ODBC and FREETDS.
>
> Currently I am only able to fetch first 63 KB chunk from the DB, and not
> able to fetch the rest of the file, below is the code that i am using to do
> so,
>
> //Trying to fetch data in chunks
> for (offset = 0; offset < colsize; offset += CHUNKSIZE) {
What is your value of CHUNKSIZE?
--
Tilghman
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Re: [asterisk-users] trixbox, sangoma a200, dell poweredge 2550 issue
> Is there any reason why I should be experiencing such bad line quality > on inbound calls from PSTN? Call quality is perfect when plugging in a > regular analogue phone. So outgoing PTSN calls are fine but incoming PTSN calls have poor quality. Do both parties hear the crackling, etc? Can you reproduce this, ie if you use your cell to call in, does the problem occur every time? What happens if you call your PTSN number from your handsets? Does the call go out onto the PTSN and come back in with poor quality? Do you have other phone lines you can try the A200 with? Have you asked Sangoma support? ___ -- 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] Not hearing first prompts
Alan Lord wrote: >> Unfortunately not. It is the same if we use our Siemens DECT/SIP >> handsets or the Ekiga softphone... >> >> I recall having this problem once before and that it "went away" when I >> changed from Ekiga to Twinkle. When I get chance, I will re-install >> Twinkle and see if that exhibits the same problem. >> I've always had this kind of problems with ekiga, the first 2/3 seconds of sound are of bad quality (cutted, metallic, ..) and always missing one or two seconds, and this last for at least 3 years. this remind me that i still hasn't verified if this as been reported as a bug to ekiga team it is the reason why i'm using twinkle as a phone client, and twinkle isn't a good client (weird translation, usability inexistant) but at least audio is good and it can ring on a sound card and phone using another one. Anyway instead of doing a Wait(), i used a Anwser() + Playback(silence/1) to get around this kind of thing. Since it's most probably problem while decoding first bunch of audio sample, using Wait() won't help. -- Benoit Plessis +33 6 77 42 78 32 <[EMAIL PROTECTED]> +33 4 67 28 06 96 ___ -- 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] Fetching Binary data from SQL Server
I am trying to write a customized app using C that would fetch voice file from
SQL Server 2000 using ODBC and FREETDS.
Currently I am only able to fetch first 63 KB chunk from the DB, and not able
to fetch the rest of the file, below is the code that i am using to do so,
fd = open(fullpath, O_RDWR | O_CREAT | O_TRUNC, 0770);
if (fd < 0) {
ast_log(LOG_WARNING, "Failed to write '%s': %s\n", fullpath, strerror(errno));
res = -1;
goto free_res;
}
res = SQLGetData(stmt, 1, SQL_BINARY, empty, 0, &colsize);
fdlen = colsize;
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "COLSIZE = %d", colsize); //PRINTING COLSIZE ON
CLI
if (fd > -1) {
char tmp[1]="";
lseek(fd, fdlen - 1, SEEK_SET);
if (write(fd, tmp, 1) != 1) {
close(fd);
res = -1;
goto free_res;
}
}
if (fd > -1){
//Trying to fetch data in chunks
for (offset = 0; offset < colsize; offset += CHUNKSIZE) {
if ((fdm = mmap(NULL, CHUNKSIZE, PROT_READ | PROT_WRITE, MAP_SHARED, fd,
offset)) == MAP_FAILED) {
ast_log(LOG_WARNING, "Could not mmap the output file: %s (%d)\n",
strerror(errno), errno);
goto free_res;
} else {
res = SQLGetData(stmt, x + 1, SQL_BINARY, fdm, CHUNKSIZE, NULL);
munmap(fdm, CHUNKSIZE);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
unlink(fullpath);
goto free_res;
}
}
}
}
close(fd);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
The value of colsize printed on CLI is 64512, Is there some limitation
somewhere in FREETDS or ODBC.
Can anyone please help me to get this fixed?
Regards,
Sanjay Rajdev
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Re: [asterisk-users] Alternate names in Directory (dial-by-name)
On Fri, 2008-05-16 at 14:42 -0600, Steve Johnson wrote: > What creative methods are used to support dial-by-name functionality > for people who go by more than one name? > > e.g.: Rebecca/Becky, Margaret/Peggy, William/Bill, Liz/Elizabeth, etc. I've always done the following (and thought it was documented somewhere, but was recently informed that it's probably not the right way to do it): In voicemail.conf, when you define your mailboxes, do something like: 1234 => ,Rebecca Jones,[EMAIL PROTECTED] 1234 => d,Becky Jones 1234 => d,Becka Jones 1234 => d,BB Jones Again, it's probably not the *correct* way to do it, but it seems to work for me. I'm sure someone smarter than me can come up with a more correct way. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Alternate names in Directory (dial-by-name)
Hi everyone, What creative methods are used to support dial-by-name functionality for people who go by more than one name? e.g.: Rebecca/Becky, Margaret/Peggy, William/Bill, Liz/Elizabeth, etc. We'd like to use the "f" first name option of the Directory function, as the particular phone system has multiple members of the same families (same last names). In the first example, the caller should be able to key either R-E-B or B-E-C and get to the same person/mailbox. Any suggestions? Thanks, S. ___ -- 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] Polycom XML Files / asterisk
So for HTTP provisioning in a hosted environment, how would you make it secure using Option 66 in a customers router. Would you have to pass a variable with a password in the Option 66 string? eg http://http.provider.com?customer=999?password=password and would the polycoms automatically be able to upload log files etc using a method such as this? Does anyone have any ideas on running this securely? Robert On Thu, May 15, 2008 at 5:13 PM, Mark Hamilton <[EMAIL PROTECTED]> wrote: > Since, we're on the the topic of phones, and TFTPing.. if someone on this > thread has some knowledge of putting configs on Cisco IP Phone 7960, can > they please contact me off list? > > I've done the configs via tftp, etc but anything into the speaker/handset > relating to voice doesn't work. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Robert > McNaught > Sent: May 15, 2008 6:41 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Polycom XML Files / asterisk > > Limiting to HTTP would be OK if every customer had a static IP - if > you have small offices, then they maybe on DSL without static IP, > which makes that difficult - you could of course force your users to > have static IPs. > > Robert > > On Thu, May 15, 2008 at 1:45 PM, Atis Lezdins <[EMAIL PROTECTED]> wrote: >> On Thu, May 15, 2008 at 10:08 PM, Robert McNaught >> <[EMAIL PROTECTED]> wrote: >>> The way I understood it is that TFTP does not allow you to set a >>> username and password in a URL >>> like tftp://username:[EMAIL PROTECTED] is not possible >>> when setting option 66 >>> >>> Is it not possible to require a username and password with HTTP? I >>> assumed that you could just like if you were protecting the web root >>> directory on a webserver to require authentication credentials, >>> although have never tried this. >> >> You can always limit access to HTTP for certain IP range. Isn't that >> enough? Then add auth in your request string - for example: >> http://provisioning.mysite.com/secure/234sdfsdf3247sd/- unless you >> enable directory listing, it should be at same security level as http >> with authentication or ftp (any of those can be sniffed) >> >> Another thing I like in HTTP - you can redirect config read to execute >> any script, write simple PHP that will generate resulting config, with >> lookup of correct extension by MAC. Much like DHCP. >> >> Regards, >> Atis >> >>> >>> Robert >>> >>> >>> >>> On Thu, May 15, 2008 at 10:43 AM, Anthony Francis <[EMAIL PROTECTED]> > wrote: I am confused how TFTP is less secure than HTTP. TFTP does not allow any browsing, ever. Neither technologies will allow the device to authenticate before downloading a configuration file, and both are easily secured by only permitting connections from specific hosts. Robert McNaught wrote: > Yes, perhaps a script would always be better than hand-touching these > files, and getting an XML editor only really makes it easier on the > eyes. > > On the same subject, I have noticed that Snom and Linksys phones do > not support FTP provisioning - only TFTP and HTTP. With TFTP being an > insecure option for a hosted architecture, is everyone moving to > provision Polycoms with HTTP, so that both can be auto-provisioned via > Option 66. > > One thing I found is that, with option 66 in a LAN router, you cannot > specify more than one protocol. > > Has anyone had any problems provisioning Polycoms with HTTP? > > > On Thu, May 15, 2008 at 1:35 AM, Philipp Kempgen > <[EMAIL PROTECTED]> wrote: > >> Robert McNaught schrieb: >> >> >>> Does anyone know how to apply a style sheet to the polycom automatic >>> provisioning XML files? >>> >> Why should applying a stylesheet be different than for any other >> XML files? >> >> >>> Even better, does anyone know of a web-based XML editor where you can >>> just edit the files from a browser directly ie entering in phone >>> number, display name, proxy address etc. From what I gather, most >>> people are just using Notepad to change the files then upload them, > or >>> vi from the command line, which is fiddly and time-consuming. >>> >> Just use your preferred editor. Nobody forces Notepad or vi upon you. >> >> Even better: Generate the config files with Perl/PHP/> language>. >> >> >> Grüße, >> Philipp Kempgen >> -- >> Asterisk-Tag.org 2008, 26.-27. Mai -> http://www.asterisk-tag.org >> amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de >> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or upd
Re: [asterisk-users] Problem with cisco 7970G [EMAIL PROTECTED]
Jorge Munoz wrote: > > Hi everyone > > This is the first time I post something here so I’m sorry about my > English, I don’t know how to write properly. > > Well, I’ve been working with Cisco 7960 telephones and my boss bought > new ones , 7970G with SIP70.8-2-2SR3S firmware version, those work > perfectly, but one of them has the SIP70.8.3.5S version, and this one > doesn’t connect to the server , I wanted to install the > SIP70.8.2.2SR3S version, but I couldn’t, is there anyone who knows how > to do it? > > Many thanks. > > > > When I updated to SIP70.8.3.5S on my 7970 I had to change 1 to in the XML file to make the phone register. I believe it is a bug in the new firmware. ___ -- 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] Digium TDM4xx CID problem
Hi list, Has anyone here used one of these cards and got it to recognize incoming CIDs in Denmark, Sweden, or the Netherlands? I'm still looking for a way to attach an analog line to my Asterisk system in the Netherlands that recognizes incoming CIDs. I've now purchased a Digium Wildcard TDM401BF: a basic card with a single FXO module and no echo cancellation. Yes, it's more expensive than a Linksys SPA3102, but if anything was supposed to work it was this, but so far it ain't. Most recently, I was pointed to this bug report: http://bugs.digium.com/print_bug_page.php?bug_id=9 Looks very relevant! However, even though it seems to apply to Asterisk v1.4.19 (the version I'm currently using) this issue was closed almost four years ago and it looks like the diff files have long since been added to the main development tree. Anyway, except for not detecting incoming CIDs, the card works fine. My zapata.conf looks like this: [trunkgroups] [channels] language=en rxwink=300 cidsignalling=dtmf relaxdtmf=yes cidstart=polarity usecallerid=yes callerid=asreceived callwaitingcallerid=yes callwaiting=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=14 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming group=1 signalling=fxs_ks callerid=asreceived context=from-pstn channel => 1 Any ideas? Thanks, Jaap ___ -- 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-biz] Anonymous statistics collection tool forAsterisk servers?
On Fri, 16 May 2008, Dean Collins wrote: > It would be voluntary to download the module in the first place so not > necessary to be able to 'turn off'. Even "scrubbed" anonymous data has been able to be used in ways that really don't make it anonymous. When Netflix offered $1m to anyone who could improve their movie recommendations, they released a large amount of what they believed was scrubbed, anonymous data. Turns out, it wasn't so anonymous. http://www.wired.com/politics/security/commentary/securitymatters/2007/12/securitymatters_1213 I decidedly would not participate. Beckman --- Peter Beckman Internet Guy [EMAIL PROTECTED] http://www.angryox.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] Not hearing first prompts
Sherwood McGowan wrote: >> > Hrm...I have encountered this before and sometimes doing an explicit > Answer() then a Wait(2), then calling the service can help. > > Hope this is helpful > > Sherwood McGowan > Bingo! Thanks a bunch. That sorted it. Al -- The way out is open! http://www.theopensourcerer.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 concurrent calls count
Alexander Olekhnovich wrote: > Hi Asterisk Users, > > I'm interested in how many concurrent calls Asterisk can process > without troubles. I mean 1 Asterisk server (software) like either > proxy or media server (any numbers will be appropriate). > > 1. Is there any limitations by the software? What is this number? > 2. What is the maximum count of concurrent calls you've ever seen/tested? > > -- > Thanks in advance > Alexander Olekhnovich > > > ___ > -- 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 Rather than jump into the heavy list of replies, in which there's some heated discussion, I thought I'd offer a quick $0.02: Asterisk's concurrent call capabilities is limited (as far as I know) only by the hardware you're using and the implementation. By this I mean that the amount of transcoding, meetme conferences, SIP/IAX/Zap channels, recording, CDR backend, etc...all take their toll on your hardware's capabilities. I'll give you two examples: 1. On a Dual 1.5Ghz XEON, 2GB RAM server running CentOS 4.5(unsure on this anymore) with only Asterisk 1.4 TRUNK in 1995 in a SIP only environment with ONLY ulaw encoding, I've seen 500+ concurrent calls with over 2K users on a single machine. All clients were set for canreinvite=no, and qualify=yes. This system did not show degradation of performance. 2. I'm currently working with a client that has a Dual 2.5 Ghz, 2GB RAM server, running Debian Etch. They are running two EM Wink T1 Trunks, and 51 Zap phones locally running through Adtran Total Access Channel Banks, 12 POTS lines running through a Rhino channel bank, and 27 SIP Phones. Concurrent calls only run at around 43 calls currently, although I've seen it as high as 53, and ALL calls are recorded other than local spying on channels and inter-extension calls. Additionally, this server has PostgreSQL and Apache running on it to allow administration to review CDRs and pull recordings, and a Zabbix monitoring agent daemon sending data to a local network Zabbix server. This server showed little or no degradation in call quality or service (even with Sox and Speexmix running in the background converting recordings via a background script) until just recently when we changed T1 providers and got EM Wink instead of the requested PRI. Before we had 99.999% of all calls complete from dial to hangup with no issues. Now we're at 98.8%, with calls being dropped in midconversation. I have not found the answer to what is causing the server to drop calls, other than after the switchover to EM_W our Zaptel accuracy started degrading. We are in the process of figuring out how we can resolve this, including possible hardware upgrades (which were already planned for handling recordings better) I hope these two examples help show you how two similar machines can vary drastically in performance with similar hardware. Differences in implementation make a BIG difference. Slainte, Sherwood McGowan ___ -- 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 concurrent calls count
On Fri, 16 May 2008, Steve Totaro wrote: > On Fri, May 16, 2008 at 9:47 AM, David Backeberg <[EMAIL PROTECTED]> wrote: >> >> Has anybody ever tried to roll their own VoIP or Zaptel load >> simulator? How did they do it? >> > SIPP can help with benchmarking SIP calls and you can loop back T1 > calls if you have two machines with T1 cards or even one machine with > multiple T1 ports. > > Then just look at top. Make a few test calls and see if they are choppy What value do you look at with top? (Especially with multiple processor/core servers.) I have an old 1.2.7 server with "custom features" hacked in that leaks memory. We know audio quality goes to hell when Asterisk hoards more than 100mb. How do you quantify "choppy?" Anybody volunteer to write "app-MOS?" Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] trixbox, sangoma a200, dell poweredge 2550 issue
Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the Linksys SPA942 VoIP Handsets, they are connected also over VPN to the remote Trixbox. When placing calls from either Office 1 or Office 2 calls are fine, there is echo to start with (Echo training) but then it clears. Volume is ok, quality is ok. However when an incoming call is made to the office the quality is pretty bad, crackles, low volume. There has also been mention (In office 1) of some cross talk. The Trixbox is plugged into the master switch in the server room, where the various production servers are running, and all workstations are also connected directly to the switch. Is there any reason why I should be experiencing such bad line quality on inbound calls from PSTN? Call quality is perfect when plugging in a regular analogue phone. I am in the process of having all the VoIP handsets in Office 1 connected directly to its own switch, and then connected to the Trixbox directly to avoid server/workstation bandwidth issues, but I don't think would solve the quality issues. Is the PowerEdge powerful enough to handle the VoIP traffic. I am really at an end with this, and really need to find a solution. Any help, advice or pointers would be greatly welcomed. Thanks, PaulG. ___ -- 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-biz] Anonymous statistics collection tool forAsterisk servers?
On Fri, 16 May 2008, Dean Collins wrote: > It would be voluntary to download the module in the first place so not > necessary to be able to 'turn off'. I think adoption would be higher if it could create a web page showing stats that the customer would be interested in -- like a graph of concurrent calls for each minute in a day. Sort of a "carrot" to encourage sharing. Counting installations that don't want to report stats would also be interesting. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 concurrent calls count
David Backeberg wrote: >>> Has anybody ever tried to roll their own VoIP or Zaptel load >>> simulator? How did they do it? >>> >> SIPP can help with benchmarking SIP calls and you can loop back T1 >> calls if you have two machines with T1 cards or even one machine with >> multiple T1 ports. >> > > SIPp looks like it's exactly the right tool for voip load generation / > simulation. Sweet! > > ___ > -- 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 > I've used SIPp before for benchmarking, it works quite well ___ -- 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 concurrent calls count
On Friday 16 May 2008 11:00:09 Steve Totaro wrote: > On Fri, May 16, 2008 at 11:44 AM, Tilghman Lesher wrote: > > On Friday 16 May 2008 09:11:11 Steve Totaro wrote: > >> On Fri, May 16, 2008 at 9:56 AM, Tilghman Lesher wrote: > >> > Digium is under no obligation to give you numbers for your own > >> > hardware. That's up to you (and you get to control your own set of > >> > variables). > >> > >> While under no obligation, it certainly would help sales. > > > > Whose sales? If you're talking about the appliances, then yes, I'm sure > > the publication of those numbers help with sales. If you mean your own > > sales, well, you're right, Digium's numbers probably don't help your > > sales. You could certainly put together a lab and do your own testing. > > Why don't you do that? > > Sales in general. You don't need to benchmark everything, just a few > basic benchmarks, maybe gear it to your hardware and SIP as a gateway, > then build from there. Most companies do this. Precisely. The numbers Digium gives are geared to their own machines. > >> This is in the style of legacy proprietary systems and anther reason > >> why the sale cycle goes a little tougher than a custom job. Asterisk > >> with FreePBX (and maybe Druid) eliminate these artificial constraints > >> on usage. > > > > Yes, but the point of those constraints is to permit support a manageable > > job. Yes, you could probably add 2 or 3 or 10 or 15 to the number of > > calls that a particular machine could handle, but from a support > > perspective, it doesn't matter how many the machine could theoretically > > handle, it matters how many it could handle in the particular > > installation in a supportable configuration (those are all those pesky > > variables we've been talking about). > > Maybe that is what the official corporate answer is or, you were > brainwashed to believe, but I tend to think it is to sell SMB and > Enterprise software and support. It is all about money. I didn't > fall off the turnip truck yesterday. Now who's on the attack here? Instead of taking issue with the logic, you're personally attacking me, and I do take offense. The logic is sound, and it is precisely the reason why we say "X machine supports Y users". It makes it easier for the support department, that they don't have to deal with edge cases of "Well, if you're doing the maximum transcoding AND recording AND conferences AND a few other things, then maybe it won't support Y users." No, we want the numbers solid; we never want it to be said that we sold what we could not support. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI debugging ...
Have a problem with an ISDN30 line in the UK. It mostly works though, but I get these messages on the output: -- SIP/211-081e5fa0 answered Zap/5-1 Write to 35 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) and this is followed immediately by: May 16 11:59:27 WARNING[17242]: chan_zap.c:3878 zt_handle_event: Detected alarm on channel 5: Red Alarm == Spawn extension (macro-dialInternal, s, 41) exited non-zero on 'Zap/5-1' in macro 'dialInternal' == Spawn extension (macro-dialInternal, s, 41) exited non-zero on 'Zap/5-1' plus a whole host of other messages I see when I reset it - eg. May 16 11:59:27 WARNING[1227]: chan_zap.c:6532 handle_init_event: Detected alarm on channel 6: Red Alarm May 16 11:59:27 WARNING[1227]: chan_zap.c:1584 zt_disable_ec: Unable to disable echo cancellation on channel 6 All the way to channel 14, then: May 16 11:59:27 NOTICE[1226]: chan_zap.c:8365 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2 May 16 11:59:27 WARNING[1226]: chan_zap.c:2441 pri_find_dchan: No D-channels available! Using Primary channel 20 as D-channel anyway! and then it resets the whole thing, and is back again a few seconds later, but meanwhile all calls have been dropped. Now the ISDN30 line has been running for some time (years) with a Panasonix PBX, so I'm fairly sure it's OK. This is my little experimental box I mentioned recently - a TDM400 card and a TE120P card. They're on separate interrupts according to /proc/interrupts and lspci -v. The TE120P card is sharing an interrupt with the USB hardware, but there is nothing plugged into it (however the next time I can get console access, I'll disable USB in the BOIS) The only thing I can see which might not be right is /proc/zaptel/2: Span 2: WCT1/0 "Wildcard TE120P Card 0" HDB3/CCS/CRC4 IRQ misses: 690 So 690 misses since power up - 24 hours now, just over 1000 calls since 9am today. I can't find anything specific that triggers it - eg. it doesn't seem to have anything to do with calls via the TDM400 card (2 fax machines and 2 analogue trunks to a premicell GSM box) So where to start trying to track this down... Any hints? Config files are relatively sane, I hope: /etc/zaptel.conf: fxoks=1 fxoks=2 fxsks=3 fxsks=4 span=2,1,0,ccs,hdb3,crc4 bchan=5-19 dchan=20 bchan=21-35 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf: ; Channels 5-19, 21-35: ; ISDN30 channels context=incoming resetinterval=never group=2 usecallerid=yes faxdetect=none signalling=pri_cpe switchtype=euroisdn internationalprefix=00 nationalprefix=0 overlapdial=yes pridialplan=unknown immediate=no rxgain=0.0 txgain=0.0 channel => 5-14 We only have 10 channels "lit". asterisk-1.2.26.1 zaptel-1.2.23 libpri-1.2.7 Also kernel 2.6.18 if that's any issues. Cheers, Gordon ___ -- 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] Connecting a PSTN gateway to Asterisk using PRI
On point to point data, you will just be sending the calls over SIP. While theoretically, you could use a Digium or Sangoma card to terminate your data T1, I would suggest a Cisco box. Using G729 or GSM (or even Speex if you are cool ;-) you can push many more calls through the circuit than a point to point tie or voice circuit. Thanks, Steve Totaro On Fri, May 16, 2008 at 11:44 AM, Peter Eisch <[EMAIL PROTECTED]> wrote: > > Using a T1/E1 ISDN interface it's somewhat trivial. In zapata's conf: > > group=0,11 > context=from-pbx-custom > switchtype = national > signalling = pri_net > pridialplan=national > prilocaldialplan=national > channel => 1-23 > group= > context=default > > Note the pri_net for signalling. I have several PRI spans running this way > to PBXs. Then configure your dialplan to match the remote site's extensions > to use the right trunk interface. > > > On 5/16/08 9:48 AM, "Al Baker" <[EMAIL PROTECTED]> wrote: > >> This is 'basically' a tie-line between the boxes. >> Yes - it is done all the time between PBX's. You are basically nailing >> up a circut between the boxes. >> It could be a simple as a simple POTS leased line or a multi-t1 bundle >> between them. >> How it is physically done with DIGIUM's boards under * ? >> >> Someone else will have to answer that >> >> >> Pascal Maugeri wrote: >>> Hi >>> >>> I have a system (S) that has a PSTN gateway to accept incoming calls >>> and setup outgoing calls from/to Telco network. In the other hand I >>> have a distant Asterisk box (A) that I would like to connect to (S) >>> using the PRI interface. >>> >>> I understand that the proper way is to order to my Telco two PRI lines >>> one for (S) and another for (A), and configure (S) and (A) to call >>> each other numbers when they have to interconnect. >>> >>> Now, might it be possible to connect directly (A) and (S) using their >>> PSTN interfaces without having to go through to my Telco ?! Does it >>> make sense ? Is it technically feasible ? I guess that the Telco >>> network is providing routing, number assignation, etc. and it sounds >>> pointless to do this. Nevertheless could you confirm it is >>> possible/impossible and why ? Is there a better way to do that ? >>> >>> Thanks in advance, >>> Pascal >>> >>> >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- 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 concurrent calls count
Inline On Fri, May 16, 2008 at 11:44 AM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Friday 16 May 2008 09:11:11 Steve Totaro wrote: >> On Fri, May 16, 2008 at 9:56 AM, Tilghman Lesher wrote: >> > On Friday 16 May 2008 06:59:15 Al Baker wrote: >> >> this is one very weak area for *. There is NO ANSWER. >> >> Now in fairness to *, the answer DOES depend on a # of critical >> >> variables. How much CODEC to CODEC transcription is going on. >> >> How many MEET Me conferences are going on. >> >> >> >> On the other hand, DIGIUM COULD, since they have a lab take 4-5 >> >> 'standard' workloads >> >> on two of the most common hardware boxes, say Dell & HP, and run x # of >> >> transcriptions and >> >> show the #'s. >> >> Then x # of meet-me conferences. >> >> >> >> Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks >> >> >> >> Rockwell and NORTEL can tell you this for every piece of hardware they >> >> sell. >> >> >> >> It is a an area DIGIUM need to "man-up" in. >> > >> > I'm not sure what your problem is with Digium. They sell several >> > machines for which they publish very specific numbers as to how many >> > users those machines will support (the Switchvox appliances). Note that >> > these machines are configurable only from the web interface, and they do >> > not allow you to install additional software. In other words, when they >> > give you a specific machine, with a ton of those variables controlled, >> > they can give you a number. >> > >> > Digium is under no obligation to give you numbers for your own hardware. >> > That's up to you (and you get to control your own set of variables). >> >> It seems any constructive criticism offered, you take as an attack >> against Digium. That is not a good attitude. > > I don't see how you figured out what I was thinking. Al said Digium doesn't > publish any numbers, and I responded, saying that he was incorrect; Digium > does indeed publish numbers (they're just not for his hardware). "I'm not sure what your problem is with Digium." Proof, period. > >> While under no obligation, it certainly would help sales. > > Whose sales? If you're talking about the appliances, then yes, I'm sure the > publication of those numbers help with sales. If you mean your own sales, > well, you're right, Digium's numbers probably don't help your sales. You > could certainly put together a lab and do your own testing. Why don't you do > that? Sales in general. You don't need to benchmark everything, just a few basic benchmarks, maybe gear it to your hardware and SIP as a gateway, then build from there. Most companies do this. I have my own lab and bechmarks but they are for Sangoma hardware and very specific servers and all geared to callcenter apps. > >> I take "Appliance Numbers" with a grain of salt. The sales model of >> SwitchVox (and most others) is based on number of ports (SoHO, SMB, >> Enterprise) not maximum number of ports that the appliance could >> actually handle if not artificially constrained. > > Consider the maximum number of ports that Switchvox will enable on a single > machine and consider that the maximum number that they're willing to support > comfortably without running into some hard limit. You never want to run into > a hard limit in the field anyway. High powered ervers are cheap and so are appliances once you settle on an enclosure and guts and start cranking out boxes. Hard limit common. > >> This is in the style of legacy proprietary systems and anther reason >> why the sale cycle goes a little tougher than a custom job. Asterisk >> with FreePBX (and maybe Druid) eliminate these artificial constraints >> on usage. > > Yes, but the point of those constraints is to permit support a manageable > job. Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls > that a particular machine could handle, but from a support perspective, it > doesn't matter how many the machine could theoretically handle, it matters > how many it could handle in the particular installation in a supportable > configuration (those are all those pesky variables we've been talking about). Maybe that is what the official corporate answer is or, you were brainwashed to believe, but I tend to think it is to sell SMB and Enterprise software and support. It is all about money. I didn't fall off the turnip truck yesterday. > >> I have load averages and CPU usage stats in my mind for all the >> various usages and hardware through experience in my mind. Of course >> they are only valuable to the exact setup I was doing. > > Precisely. > > -- > Tilghman > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- as
[asterisk-users] Meetme join conference notification.
Anyway to disable the join conference notification on the party that is joining, but not the parties already in the conference? ___ -- 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] playing .gsm sounds through a web browser
I usually use lame to do the decoding and pipe it back into sox. l. On Fri, 16 May 2008 12:58:41 +0200, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote: > Does anyone know where I can a copy of sox for windows with mp3 built in > ? > > Julian > > David Backeberg wrote: >> No, no, no. >> >> Don't try to play them directly as gsm files. Convert them to wav on >> the fly, when demanded by the user from the webpage. Have a php, or >> perl, or whatever script call sox, and push the wav to the user. sox >> runs so fast that you can do the conversion on-demand. You can decide >> what to do with the wav file afterward. Delete it immediately, cache >> it, whatever. Don't force the users to know anything they don't >> already know. There are a lot of media players that can handle a wav >> file. >> >> On Thu, May 15, 2008 at 7:26 PM, Julian Lyndon-Smith >> <[EMAIL PROTECTED]> wrote: >>> I have a lot of recordings from asterisk in a .gsm format. I would like >>> to play these files from a web browser (IE, firefox and opera) >>> >>> What do I need to do in order to achieve this goal ? >>> >>> Thanks >>> >>> Julian >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Loway Research - Home of QueueMetrics http://queuemetrics.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] Connecting a PSTN gateway to Asterisk using PRI
Using a T1/E1 ISDN interface it's somewhat trivial. In zapata's conf: group=0,11 context=from-pbx-custom switchtype = national signalling = pri_net pridialplan=national prilocaldialplan=national channel => 1-23 group= context=default Note the pri_net for signalling. I have several PRI spans running this way to PBXs. Then configure your dialplan to match the remote site's extensions to use the right trunk interface. On 5/16/08 9:48 AM, "Al Baker" <[EMAIL PROTECTED]> wrote: > This is 'basically' a tie-line between the boxes. > Yes - it is done all the time between PBX's. You are basically nailing > up a circut between the boxes. > It could be a simple as a simple POTS leased line or a multi-t1 bundle > between them. > How it is physically done with DIGIUM's boards under * ? > > Someone else will have to answer that > > > Pascal Maugeri wrote: >> Hi >> >> I have a system (S) that has a PSTN gateway to accept incoming calls >> and setup outgoing calls from/to Telco network. In the other hand I >> have a distant Asterisk box (A) that I would like to connect to (S) >> using the PRI interface. >> >> I understand that the proper way is to order to my Telco two PRI lines >> one for (S) and another for (A), and configure (S) and (A) to call >> each other numbers when they have to interconnect. >> >> Now, might it be possible to connect directly (A) and (S) using their >> PSTN interfaces without having to go through to my Telco ?! Does it >> make sense ? Is it technically feasible ? I guess that the Telco >> network is providing routing, number assignation, etc. and it sounds >> pointless to do this. Nevertheless could you confirm it is >> possible/impossible and why ? Is there a better way to do that ? >> >> Thanks in advance, >> Pascal >> >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
On Friday 16 May 2008 09:11:11 Steve Totaro wrote: > On Fri, May 16, 2008 at 9:56 AM, Tilghman Lesher wrote: > > On Friday 16 May 2008 06:59:15 Al Baker wrote: > >> this is one very weak area for *. There is NO ANSWER. > >> Now in fairness to *, the answer DOES depend on a # of critical > >> variables. How much CODEC to CODEC transcription is going on. > >> How many MEET Me conferences are going on. > >> > >> On the other hand, DIGIUM COULD, since they have a lab take 4-5 > >> 'standard' workloads > >> on two of the most common hardware boxes, say Dell & HP, and run x # of > >> transcriptions and > >> show the #'s. > >> Then x # of meet-me conferences. > >> > >> Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks > >> > >> Rockwell and NORTEL can tell you this for every piece of hardware they > >> sell. > >> > >> It is a an area DIGIUM need to "man-up" in. > > > > I'm not sure what your problem is with Digium. They sell several > > machines for which they publish very specific numbers as to how many > > users those machines will support (the Switchvox appliances). Note that > > these machines are configurable only from the web interface, and they do > > not allow you to install additional software. In other words, when they > > give you a specific machine, with a ton of those variables controlled, > > they can give you a number. > > > > Digium is under no obligation to give you numbers for your own hardware. > > That's up to you (and you get to control your own set of variables). > > It seems any constructive criticism offered, you take as an attack > against Digium. That is not a good attitude. I don't see how you figured out what I was thinking. Al said Digium doesn't publish any numbers, and I responded, saying that he was incorrect; Digium does indeed publish numbers (they're just not for his hardware). > While under no obligation, it certainly would help sales. Whose sales? If you're talking about the appliances, then yes, I'm sure the publication of those numbers help with sales. If you mean your own sales, well, you're right, Digium's numbers probably don't help your sales. You could certainly put together a lab and do your own testing. Why don't you do that? > I take "Appliance Numbers" with a grain of salt. The sales model of > SwitchVox (and most others) is based on number of ports (SoHO, SMB, > Enterprise) not maximum number of ports that the appliance could > actually handle if not artificially constrained. Consider the maximum number of ports that Switchvox will enable on a single machine and consider that the maximum number that they're willing to support comfortably without running into some hard limit. You never want to run into a hard limit in the field anyway. > This is in the style of legacy proprietary systems and anther reason > why the sale cycle goes a little tougher than a custom job. Asterisk > with FreePBX (and maybe Druid) eliminate these artificial constraints > on usage. Yes, but the point of those constraints is to permit support a manageable job. Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls that a particular machine could handle, but from a support perspective, it doesn't matter how many the machine could theoretically handle, it matters how many it could handle in the particular installation in a supportable configuration (those are all those pesky variables we've been talking about). > I have load averages and CPU usage stats in my mind for all the > various usages and hardware through experience in my mind. Of course > they are only valuable to the exact setup I was doing. Precisely. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems passing variables from a macro
I pass a value from a macro by storing the value needed to the $
{MACRO_RESULT} variable. This is returned and because of this
available after finishing the macro. I'm not sure that it works in the
way you are looking for but it works for me.
Erik de Wild
Tripple-o
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Re: [asterisk-users] Connecting a PSTN gateway to Asterisk using PRI
There is nothing special (beyond the regular configs) that need to be done with Digium (Sangoma) or any compatible board. Since you are using point to point data lines, I would suggest using SIP and whatever codec fits your needs and bandwidth. The rest is done it the dialplan. Thanks, Steve Totaro On Fri, May 16, 2008 at 10:48 AM, Al Baker <[EMAIL PROTECTED]> wrote: > This is 'basically' a tie-line between the boxes. > Yes - it is done all the time between PBX's. You are basically nailing > up a circut between the boxes. > It could be a simple as a simple POTS leased line or a multi-t1 bundle > between them. > How it is physically done with DIGIUM's boards under * ? > > Someone else will have to answer that > > > Pascal Maugeri wrote: >> Hi >> >> I have a system (S) that has a PSTN gateway to accept incoming calls >> and setup outgoing calls from/to Telco network. In the other hand I >> have a distant Asterisk box (A) that I would like to connect to (S) >> using the PRI interface. >> >> I understand that the proper way is to order to my Telco two PRI lines >> one for (S) and another for (A), and configure (S) and (A) to call >> each other numbers when they have to interconnect. >> >> Now, might it be possible to connect directly (A) and (S) using their >> PSTN interfaces without having to go through to my Telco ?! Does it >> make sense ? Is it technically feasible ? I guess that the Telco >> network is providing routing, number assignation, etc. and it sounds >> pointless to do this. Nevertheless could you confirm it is >> possible/impossible and why ? Is there a better way to do that ? >> >> Thanks in advance, >> Pascal >> >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?
Eric Wieling wrote: > Make SURE you are not using callprogress=yes or busydetect=yes (they > default to no). These options are commonly known in the Asterisk world > as "randomlydisconnectmycalls=yes". > > Sherwood McGowan wrote: > >> Steve Totaro wrote: >> >>> On Thu, May 15, 2008 at 12:59 PM, Don Pobanz >>> <[EMAIL PROTECTED]> wrote: >>> >>> On Thursday, May 15, 2008 11:11 AM - Sherwood McGowan said ... > we've been temporarily stuck with a pair of EM Wink T's. Ever since > then, we've been dropping 1-2% of all calls (in or out) and even more > strange, when a call gets dropped, a phantom call was being > generated on > the incoming side, but only by Asterisk, the T providers (Qwest) say > they have no records of those calls. > > ... I don't know whether this could be related or not but are you set to loop timing on your incoming phone company T1 port? I have seen timing issues create some strange issues. By the way, we are using incoming EM wink trunks delivered over a T1 and are not having any issues. We are using Asterisk 1.4.18 with Zaptel 1.4.10. Don Pobanz -- MailDefender Message Security: Click below to verify authenticity http://www.exchangedefender.com/verify.asp?id=m4FH3AwE015747&[EMAIL PROTECTED] >>> How did your dialplan change. When do the hangups occur? Is there a >>> pattern? >>> >>> I have done a few E&M wink setups with no issues other than getting >>> the configs and dialplan right. >>> >>> Personally, I would ream your provider rep and demand that your >>> situation be escalated to the top. If they don't agree, go over their >>> head, I always get an escalation form with contacts for each >>> escalation level. Obviously, when thing are straightened out, make it >>> a point be very thankful to everyone and CC them with what a great job >>> they did. >>> >>> Thanks, >>> Steve Totaro >>> >>> > > >> Steve, >> Thanks for the input. The dialplan did not change when this symptom >> started happening. There is no pattern other than we drop 1 or 2 calls >> at a time, I've reviewed the dialplan executions (verbose level 3 output >> to log file) and the call is executing as normal but then the far end >> (trunk side) just disconnects, and we then go to timeout (which it should). >> >> I've tried to be cool with our reps so far, but it IS getting a little >> ridiculous, sounds like I may have to go that route. >> >> Sherwood McGowan >> > > > Thanks for the tip. I had those enabled, but not for the EM_W trunks, only for my POTS end. After speaking with Digium I have disabled them completely, and I'm waiting to see the results from today's set of calls. Thanks to everyone so far for the suggestions, it's good to know that even after all these years (and an absence from the mailing lists) I still find the community just as helpful and kind as before :) Sherwood McGowan ___ -- 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] Not hearing first prompts
Alan Lord wrote: > Stefan Guenther wrote: > >> Alan Lord wrote: >> >> >When I connect to various asterisk services such as VoicemailMain(), >> >MeetMe() as examples, I do not get to hear the first greeting messages. >> > >> >I've tried adding a Wait(1) before or after the application but this >> >seems to have no effect. >> > >> >Is there another setting/parameter I can play with to delay the start >> >of the playback of these messages? >> > >> do you use SNOM phones? >> >> We had the same problem with a number of SNOM phones. >> >> > > Unfortunately not. It is the same if we use our Siemens DECT/SIP > handsets or the Ekiga softphone... > > I recall having this problem once before and that it "went away" when I > changed from Ekiga to Twinkle. When I get chance, I will re-install > Twinkle and see if that exhibits the same problem. > > Thanks anyway. > > Al > Hrm...I have encountered this before and sometimes doing an explicit Answer() then a Wait(2), then calling the service can help. Hope this is helpful Sherwood McGowan ___ -- 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 concurrent calls count
On Fri, May 16, 2008 at 10:42 AM, John Signorello <[EMAIL PROTECTED]> wrote: > > > Steve Totaro wrote: > > I'm not sure what your problem is with Digium. They sell several machines > for which they publish very specific numbers as to how many users those > machines will support (the Switchvox appliances). Note that these machines > are configurable only from the web interface, and they do not allow you to > install additional software. In other words, when they give you a specific > machine, with a ton of those variables controlled, they can give you a > number. > > Digium is under no obligation to give you numbers for your own hardware. > That's up to you (and you get to control your own set of variables). > > -- > Tilghman > > > > To prove that the the numbers are artificial for SwitchVox and profit > driven: > > Boot your SwitchVox in single user mode. Create an account with root > privileges. Login via SSH and you can install any software you want > and have access to top and whatever else you care to use. Switchvox > is running Fedora Core 6 and Asterisk 1.2 just an FYI > > Thanks, > Steve Totaro > > ___ > -- 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 > > > Concurrent users is only valid if you can specify what all of the concurrent > users are doing. > > If you have 13 users engaged in phone conversations, you have a different > load and system > dynamic versus 4 users in conversations and 9 users checking voice mail and > perhaps recording > new greetings. > > The artificiality arises when you use your "concurrent user" figures to > portray your product as better > than the other guys without qualifying what the "concurrent users" are > doing. > I am very aware of this but do you think that the SoHo box (and it's artificial cap) is maxing out no matter what the users are doing? If so, then why can you just upgrade concurrent use via a CC and the website? I don't have a product per se. I am generally against "Appliances" because they are moving back to the proprietary system pricing scheme of per port and cut my bottom line. They also generally have a lot of legalese attached. My "product" is built to meet the current and future needs of the customer. This is all part of the sales and quoting cycle. Thanks, Steve Totaro ___ -- 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 concurrent calls count
You can NOT use bogomips as any kind of measurement for system performance. First of all, Bogomips is a linux-specific thing and not available on other OS that Asterisk runs on. The second, and far more important point. "Bogo" is taken from the word "Bogus". Bogomips are not a measurement of system performance, it is simply a number used for calibrating parts of the kernel for your CPU. The problem with coming up with these numbers of concurrent calls is that Asterisk is not a "complete package". Meaning, it's the software portion only, most other systems when you get them are going to be the software & the hardware in one package, the 2 go hand in hand and are specifically designed for each other. Asterisk does not fall into that category unless you invest in one of the many asterisk appliances out there. Digium has no control over what hardware you are going to run Asterisk on, so they can't provide you with these numbers. Heres a few questions at the top of my head that I think would influence the answer: are you recording calls? are you transcoding calls? are you using T1s or SIP/IAX trunks? Did you buy the 7.2krpm, 10krpm, or 15krpm hard-drives? Do your harddrives have 8mb, 16mb, or 32mb cache? Did you buy the better SAS controller? Did you buy 667mhz or 800mhz ram? Are you using EXT3, ReisferFS, XFS, JFS, ZFS, UFS? Are you using AGIs? Are you using MeetMEs? How many? Whats the average length of the conferences? Are devices using re-invites to take Asterisk out of the call loop? The list goes on and on... and every single one of those answers is going to influence that number for "How many calls can my system handle?" -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Friday, May 16, 2008 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk concurrent calls count I wonder if there's a proportion where somebody could take some standard kernel output, say bogomips, and guesstimate some proportionality from that. As in: bogomips says this, expect ballpark 120 SIP over codec calls. It certainly seems like there could be some kind of asterisk benchmarking utility kindof like Sandra for Windows. I know there are a gazillion variables in asterisk, and that's why asterisk is so powerful. But some benchmarking utility would at least allow some (even if phony baloney) relative comparisons between similar hardware. Has anybody ever tried to roll their own VoIP or Zaptel load simulator? How did they do it? On Fri, May 16, 2008 at 7:59 AM, Al Baker <[EMAIL PROTECTED]> wrote: > this is one very weak area for *. There is NO ANSWER. > Now in fairness to *, the answer DOES depend on a # of critical variables. > How much CODEC to CODEC transcription is going on. > How many MEET Me conferences are going on. > > On the other hand, DIGIUM COULD, since they have a lab take 4-5 > 'standard' workloads > on two of the most common hardware boxes, say Dell & HP, and run x # of > transcriptions and > show the #'s. > Then x # of meet-me conferences. > > Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks > > Rockwell and NORTEL can tell you this for every piece of hardware they sell. > > It is a an area DIGIUM need to "man-up" in. > > Alexey Shimeshov wrote: >> Hello, Alexander. >> >> AO> Hi Asterisk Users, >> >> AO> I'm interested in how many concurrent calls Asterisk can process without >> AO> troubles. I mean 1 Asterisk server (software) like either proxy or media >> AO> server (any numbers will be appropriate). >> >> AO> 1. Is there any limitations by the software? What is this number? >> AO> 2. What is the maximum count of concurrent calls you've ever seen/tested? >> >> Look at this example >> >> http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm >> >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a PSTN gateway to Asterisk using PRI
This is 'basically' a tie-line between the boxes. Yes - it is done all the time between PBX's. You are basically nailing up a circut between the boxes. It could be a simple as a simple POTS leased line or a multi-t1 bundle between them. How it is physically done with DIGIUM's boards under * ? Someone else will have to answer that Pascal Maugeri wrote: > Hi > > I have a system (S) that has a PSTN gateway to accept incoming calls > and setup outgoing calls from/to Telco network. In the other hand I > have a distant Asterisk box (A) that I would like to connect to (S) > using the PRI interface. > > I understand that the proper way is to order to my Telco two PRI lines > one for (S) and another for (A), and configure (S) and (A) to call > each other numbers when they have to interconnect. > > Now, might it be possible to connect directly (A) and (S) using their > PSTN interfaces without having to go through to my Telco ?! Does it > make sense ? Is it technically feasible ? I guess that the Telco > network is providing routing, number assignation, etc. and it sounds > pointless to do this. Nevertheless could you confirm it is > possible/impossible and why ? Is there a better way to do that ? > > Thanks in advance, > Pascal > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a PSTN gateway to Asterisk using PRI
On Fri, May 16, 2008 at 10:28 AM, Pascal Maugeri <[EMAIL PROTECTED]> wrote: > Hi > > I have a system (S) that has a PSTN gateway to accept incoming calls and > setup outgoing calls from/to Telco network. In the other hand I have a > distant Asterisk box (A) that I would like to connect to (S) using the PRI > interface. > > I understand that the proper way is to order to my Telco two PRI lines one > for (S) and another for (A), and configure (S) and (A) to call each other > numbers when they have to interconnect. > > Now, might it be possible to connect directly (A) and (S) using their PSTN > interfaces without having to go through to my Telco ?! Does it make sense ? > Is it technically feasible ? I guess that the Telco network is providing > routing, number assignation, etc. and it sounds pointless to do this. > Nevertheless could you confirm it is possible/impossible and why ? Is there > a better way to do that ? > > Thanks in advance, > Pascal > You could price out a point to point T1 between A and S and then send the traffic over via VoIP. You could use LCR hop on and hop off. Provide your own routing and number assignment (beyond what is required by the telco. I guess it comes down to the cost of the point to point. The calls between sites would be free as they are VoIP. You could also eliminate getting the second voice T1 at the remote location if you don't really need all those channels or you could get the second T1 in the first site and use Asterisk to handle routing for local or calls to go over the point to point connection. Some people will say you can just go over the public internet, but I don't recommend it. It "may" work OK but what if it doesn't? What are your calls worth? Thanks, Steve Totaro ___ -- 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 concurrent calls count
Steve Totaro wrote: I'm not sure what your problem is with Digium. They sell several machines for which they publish very specific numbers as to how many users those machines will support (the Switchvox appliances). Note that these machines are configurable only from the web interface, and they do not allow you to install additional software. In other words, when they give you a specific machine, with a ton of those variables controlled, they can give you a number. Digium is under no obligation to give you numbers for your own hardware. That's up to you (and you get to control your own set of variables). -- Tilghman To prove that the the numbers are artificial for SwitchVox and profit driven: Boot your SwitchVox in single user mode. Create an account with root privileges. Login via SSH and you can install any software you want and have access to top and whatever else you care to use. Switchvox is running Fedora Core 6 and Asterisk 1.2 just an FYI Thanks, Steve Totaro ___ -- 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 Concurrent users is only valid if you can specify what all of the concurrent users are doing. If you have 13 users engaged in phone conversations, you have a different load and system dynamic versus 4 users in conversations and 9 users checking voice mail and perhaps recording new greetings. The artificiality arises when you use your "concurrent user" figures to portray your product as better than the other guys without qualifying what the "concurrent users" are doing. -- John Signorello Managing Partner ISPBX LLC Bus: 866 GO ISPBX ext 2000 Dir: 973-841-2061 Cell: 973-534-0888 http://ispbx.com http://cogoblue.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] Connecting a PSTN gateway to Asterisk using PRI
Hi I have a system (S) that has a PSTN gateway to accept incoming calls and setup outgoing calls from/to Telco network. In the other hand I have a distant Asterisk box (A) that I would like to connect to (S) using the PRI interface. I understand that the proper way is to order to my Telco two PRI lines one for (S) and another for (A), and configure (S) and (A) to call each other numbers when they have to interconnect. Now, might it be possible to connect directly (A) and (S) using their PSTN interfaces without having to go through to my Telco ?! Does it make sense ? Is it technically feasible ? I guess that the Telco network is providing routing, number assignation, etc. and it sounds pointless to do this. Nevertheless could you confirm it is possible/impossible and why ? Is there a better way to do that ? Thanks in advance, Pascal ___ -- 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 concurrent calls count
> > I'm not sure what your problem is with Digium. They sell several machines > for which they publish very specific numbers as to how many users those > machines will support (the Switchvox appliances). Note that these machines > are configurable only from the web interface, and they do not allow you to > install additional software. In other words, when they give you a specific > machine, with a ton of those variables controlled, they can give you a number. > > Digium is under no obligation to give you numbers for your own hardware. > That's up to you (and you get to control your own set of variables). > > -- > Tilghman > To prove that the the numbers are artificial for SwitchVox and profit driven: Boot your SwitchVox in single user mode. Create an account with root privileges. Login via SSH and you can install any software you want and have access to top and whatever else you care to use. Switchvox is running Fedora Core 6 and Asterisk 1.2 just an FYI Thanks, Steve Totaro ___ -- 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] playing .gsm sounds through a web browser
On Fri, May 16, 2008 at 6:58 AM, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote: > Does anyone know where I can a copy of sox for windows with mp3 built in ? It might be in the cygwin project, but I've never tried. If it's not in the project, I'm not surprised. You could download knoppix and encode them on your machine, then boot back into Windows. ;) Or encode them on that asterisk box and then copy them back. ___ -- 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 concurrent calls count
On Fri, May 16, 2008 at 9:56 AM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Friday 16 May 2008 06:59:15 Al Baker wrote: >> this is one very weak area for *. There is NO ANSWER. >> Now in fairness to *, the answer DOES depend on a # of critical variables. >> How much CODEC to CODEC transcription is going on. >> How many MEET Me conferences are going on. >> >> On the other hand, DIGIUM COULD, since they have a lab take 4-5 >> 'standard' workloads >> on two of the most common hardware boxes, say Dell & HP, and run x # of >> transcriptions and >> show the #'s. >> Then x # of meet-me conferences. >> >> Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks >> >> Rockwell and NORTEL can tell you this for every piece of hardware they >> sell. >> >> It is a an area DIGIUM need to "man-up" in. > > I'm not sure what your problem is with Digium. They sell several machines > for which they publish very specific numbers as to how many users those > machines will support (the Switchvox appliances). Note that these machines > are configurable only from the web interface, and they do not allow you to > install additional software. In other words, when they give you a specific > machine, with a ton of those variables controlled, they can give you a number. > > Digium is under no obligation to give you numbers for your own hardware. > That's up to you (and you get to control your own set of variables). > > -- > Tilghman > Tilghman, It seems any constructive criticism offered, you take as an attack against Digium. That is not a good attitude. While under no obligation, it certainly would help sales. I take "Appliance Numbers" with a grain of salt. The sales model of SwitchVox (and most others) is based on number of ports (SoHO, SMB, Enterprise) not maximum number of ports that the appliance could actually handle if not artificially constrained. This is in the style of legacy proprietary systems and anther reason why the sale cycle goes a little tougher than a custom job. Asterisk with FreePBX (and maybe Druid) eliminate these artificial constraints on usage. I have load averages and CPU usage stats in my mind for all the various usages and hardware through experience in my mind. Of course they are only valuable to the exact setup I was doing. Thanks, Steve Totaro ___ -- 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 concurrent calls count
>> Has anybody ever tried to roll their own VoIP or Zaptel load >> simulator? How did they do it? > > SIPP can help with benchmarking SIP calls and you can loop back T1 > calls if you have two machines with T1 cards or even one machine with > multiple T1 ports. SIPp looks like it's exactly the right tool for voip load generation / simulation. Sweet! ___ -- 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 concurrent calls count
On Fri, May 16, 2008 at 9:47 AM, David Backeberg <[EMAIL PROTECTED]> wrote: > > Has anybody ever tried to roll their own VoIP or Zaptel load > simulator? How did they do it? > SIPP can help with benchmarking SIP calls and you can loop back T1 calls if you have two machines with T1 cards or even one machine with multiple T1 ports. Then just look at top. Make a few test calls and see if they are choppy Thanks, Steve Totaro ___ -- 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 concurrent calls count
> I'm interested in how many concurrent calls Asterisk can process without > troubles. I mean 1 Asterisk server (software) like either proxy or media > server (any numbers will be appropriate). Since one standard answer to this question is: "it depends on how you're using it", The ideal situation is that people could rattle off statistics of their eventual load, and be able to size their hardware purchase accordingly. The reality is that while that's hard, we can do the next best thing, which is once you have the hardware running asterisk, get historical data about your real-world asterisk load. We're running the open SNMP daemon, and we've configured the open software project Cacti to do SNMP polling against our cpu load. We now have a few months of data on how two systems running Zaptel cards, with no VoIP are holding up under load. Our business is amazingly seasonal, not quite as bad as H & R Block, but similar scenario where we're very busy parts of the year, and the rest of the year, not so much. Our results: at our US Eastern time zone business, load peaks a little after 2pm EST/EDT, most business days, and dramatically tails off most days. Once we have more months of data we'll also be able to more accurately profile the seasonality of our business, as well as make some predictions about next peaks from previous peaks, given the growth rate of our business. Hope this helps people! ___ -- 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 concurrent calls count
On Friday 16 May 2008 06:59:15 Al Baker wrote: > this is one very weak area for *. There is NO ANSWER. > Now in fairness to *, the answer DOES depend on a # of critical variables. > How much CODEC to CODEC transcription is going on. > How many MEET Me conferences are going on. > > On the other hand, DIGIUM COULD, since they have a lab take 4-5 > 'standard' workloads > on two of the most common hardware boxes, say Dell & HP, and run x # of > transcriptions and > show the #'s. > Then x # of meet-me conferences. > > Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks > > Rockwell and NORTEL can tell you this for every piece of hardware they > sell. > > It is a an area DIGIUM need to "man-up" in. I'm not sure what your problem is with Digium. They sell several machines for which they publish very specific numbers as to how many users those machines will support (the Switchvox appliances). Note that these machines are configurable only from the web interface, and they do not allow you to install additional software. In other words, when they give you a specific machine, with a ton of those variables controlled, they can give you a number. Digium is under no obligation to give you numbers for your own hardware. That's up to you (and you get to control your own set of variables). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
I wonder if there's a proportion where somebody could take some standard kernel output, say bogomips, and guesstimate some proportionality from that. As in: bogomips says this, expect ballpark 120 SIP over codec calls. It certainly seems like there could be some kind of asterisk benchmarking utility kindof like Sandra for Windows. I know there are a gazillion variables in asterisk, and that's why asterisk is so powerful. But some benchmarking utility would at least allow some (even if phony baloney) relative comparisons between similar hardware. Has anybody ever tried to roll their own VoIP or Zaptel load simulator? How did they do it? On Fri, May 16, 2008 at 7:59 AM, Al Baker <[EMAIL PROTECTED]> wrote: > this is one very weak area for *. There is NO ANSWER. > Now in fairness to *, the answer DOES depend on a # of critical variables. > How much CODEC to CODEC transcription is going on. > How many MEET Me conferences are going on. > > On the other hand, DIGIUM COULD, since they have a lab take 4-5 > 'standard' workloads > on two of the most common hardware boxes, say Dell & HP, and run x # of > transcriptions and > show the #'s. > Then x # of meet-me conferences. > > Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks > > Rockwell and NORTEL can tell you this for every piece of hardware they sell. > > It is a an area DIGIUM need to "man-up" in. > > Alexey Shimeshov wrote: >> Hello, Alexander. >> >> AO> Hi Asterisk Users, >> >> AO> I'm interested in how many concurrent calls Asterisk can process without >> AO> troubles. I mean 1 Asterisk server (software) like either proxy or media >> AO> server (any numbers will be appropriate). >> >> AO> 1. Is there any limitations by the software? What is this number? >> AO> 2. What is the maximum count of concurrent calls you've ever seen/tested? >> >> Look at this example >> >> http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm >> >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with cisco 7970G [EMAIL PROTECTED]
Hi everyone This is the first time I post something here so I'm sorry about my English, I don't know how to write properly. Well, I've been working with Cisco 7960 telephones and my boss bought new ones , 7970G with SIP70.8-2-2SR3S firmware version, those work perfectly, but one of them has the SIP70.8.3.5S version, and this one doesn't connect to the server , I wanted to install the SIP70.8.2.2SR3S version, but I couldn't, is there anyone who knows how to do it? Many thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
http://voipusersconference.org On Fri, May 16, 2008 at 1:59 PM, Al Baker <[EMAIL PROTECTED]> wrote: > this is one very weak area for *. There is NO ANSWER. Hi, There have been a couple of threads on this subject this week, so I'd remind everyone that next Friday's VoIP Users Conference is about *large scale* asterisk: After many requests, we finally have someone to talk on large scale implementation of VoIP systems with asterisk. Using a farm of Asterisk and Digium cards, tens Of Thousands of simultaneous calls can be made and Mike Trest has offered to take it all apart for us to look inside. More about Mike Trest: http://www.mike.trest.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] *72 Telco Call Forwarding
WaitForRing() solved the issue completely, since I never get the 2nd ring it waits and then the channel hangs up when ATT completes the call forward. Thanks for the tip. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson Sent: Thursday, May 15, 2008 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *72 Telco Call Forwarding Is there any reason you don't want to use Wait()? However, I would use WaitForRing() myself - its also a great solution on dirty analog lines where you receive "phantom calls". That being said, I don't know how to do it without using some form of Wait.. as far as I know zapata.conf doesn't provide a method of telling Asterisk to wait for a specific period of time or rings -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Thursday, May 15, 2008 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] *72 Telco Call Forwarding Is there a way to force asterisk to ignore the first ring of a call without using Wait() ? When I active *72 call forward on my analog lines from the telco, they always send a single ring and then do the forwarding. Asterisk picks up essentially a dead line and rings the phones which gets really annoying. Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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-biz] Anonymous statistics collection tool forAsterisk servers?
It would be voluntary to download the module in the first place so not necessary to be able to 'turn off'. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-biz- > [EMAIL PROTECTED] On Behalf Of Nitzan Kon > Sent: Friday, 16 May 2008 8:13 AM > To: Commercial and Business-Oriented Asterisk Discussion > Subject: Re: [asterisk-biz] Anonymous statistics collection tool forAsterisk servers? > > If you want an honest opinion... ;) > > Anonymous statistics take bandwidth and resources, not to mention > may be a security risk (you never know how people may use sniffed > data!). If you add a statistics module to *, please make sure there > is an easy way to TURN IT OFF... :P > > What would be the benefit of having it on, anyway? > > Thanks! > > -- Nitzan > > --- On Fri, 5/16/08, Dean Collins <[EMAIL PROTECTED]> wrote: > > > From: Dean Collins <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-biz] Anonymous statistics collection tool for Asterisk > servers? > > To: asterisk-users@lists.digium.com, "Commercial and Business-Oriented > Asterisk Discussion" <[EMAIL PROTECTED]> > > Date: Friday, May 16, 2008, 8:01 AM > > Ok guys I'm only going to leave this survey up for a few > > more days; > > > > > > > > Click this link and answer what you think - it's only 4 > > questions and > > will take you less than 30 seconds to answer. > > > > http://www.questionpro.com/akira/TakeSurvey?id=954613 > > > > > > > > Lets support the Asterisk community and get an honest > > opinion on what > > people think about anonymous statistics. > > > > > > > > Cheers, > > Dean > > > > > > > > > > > > > > > > > > > > From: Dean Collins > > Sent: Saturday, 3 May 2008 12:19 PM > > To: Asterisk Users List (asterisk-users@lists.digium.com) > > Subject: Anonymous statistics collection tool for Asterisk > > servers? > > > > I think it would be great for someone to write a small > > 'anonymous collection module' that an Asterisk > > sys-admin could download > > and install on their asterisk server which uploaded the > > stats to a > > community website where the data was anonymous but still > > valuable for > > the community. > > > > > > > > Even if it just collected number of new installations > > globally > > this would be a huge help to people selling asterisk to > > their customers > > who continually ask "I've heard about this > > Asterisk open source stuff > > but how many are there installed globally anyway?" > > > > > > > > So this survey is to gauge the community's reaction to > > the > > development of an analytics tool like this. Please answer > > with your > > honest thoughts so we can gauge demand within the community > > for this > > tool. > > > > > > > > If people respond to this survey and show interest in > > sharing > > their information with the community then something > > we'll put together a > > working group to establish design parameters etc but for > > the moment just > > go here and vote if you think this is a good idea or not > > > > > > > > http://www.questionpro.com/akira/TakeSurvey?id=954613 > > > > > > > > Remember this Asterisk community analytics tool will only > > collect anonymous information. > > > > > > > > Potential analytics may include - eg. How many ip handsets > > does > > your Asterisk server have connected, but it will never > > collect their IP > > addresses or personal information/passwords etc. You will > > be able to see > > what information is being collected in a text file before > > it uploads and > > will be able to control all stats it collects. > > > > > > > > > > > > > > > > Cheers, > > > > Dean___ > > --Bandwidth and Colocation Provided by > > http://www.api-digital.com-- > > > > asterisk-biz mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-biz > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz ___ -- 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] Anonymous statistics collection tool for Asterisk servers?
Ok guys I'm only going to leave this survey up for a few more days; Click this link and answer what you think - it's only 4 questions and will take you less than 30 seconds to answer. http://www.questionpro.com/akira/TakeSurvey?id=954613 Lets support the Asterisk community and get an honest opinion on what people think about anonymous statistics. Cheers, Dean From: Dean Collins Sent: Saturday, 3 May 2008 12:19 PM To: Asterisk Users List (asterisk-users@lists.digium.com) Subject: Anonymous statistics collection tool for Asterisk servers? I think it would be great for someone to write a small 'anonymous collection module' that an Asterisk sys-admin could download and install on their asterisk server which uploaded the stats to a community website where the data was anonymous but still valuable for the community. Even if it just collected number of new installations globally this would be a huge help to people selling asterisk to their customers who continually ask "I've heard about this Asterisk open source stuff but how many are there installed globally anyway?" So this survey is to gauge the community's reaction to the development of an analytics tool like this. Please answer with your honest thoughts so we can gauge demand within the community for this tool. If people respond to this survey and show interest in sharing their information with the community then something we'll put together a working group to establish design parameters etc but for the moment just go here and vote if you think this is a good idea or not http://www.questionpro.com/akira/TakeSurvey?id=954613 Remember this Asterisk community analytics tool will only collect anonymous information. Potential analytics may include - eg. How many ip handsets does your Asterisk server have connected, but it will never collect their IP addresses or personal information/passwords etc. You will be able to see what information is being collected in a text file before it uploads and will be able to control all stats it collects. Cheers, Dean ___ -- 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 concurrent calls count
this is one very weak area for *. There is NO ANSWER. Now in fairness to *, the answer DOES depend on a # of critical variables. How much CODEC to CODEC transcription is going on. How many MEET Me conferences are going on. On the other hand, DIGIUM COULD, since they have a lab take 4-5 'standard' workloads on two of the most common hardware boxes, say Dell & HP, and run x # of transcriptions and show the #'s. Then x # of meet-me conferences. Face it the DB Industry did this 15-2- YEARS ago with TP benckmarks Rockwell and NORTEL can tell you this for every piece of hardware they sell. It is a an area DIGIUM need to "man-up" in. Alexey Shimeshov wrote: > Hello, Alexander. > > AO> Hi Asterisk Users, > > AO> I'm interested in how many concurrent calls Asterisk can process without > AO> troubles. I mean 1 Asterisk server (software) like either proxy or media > AO> server (any numbers will be appropriate). > > AO> 1. Is there any limitations by the software? What is this number? > AO> 2. What is the maximum count of concurrent calls you've ever seen/tested? > > Look at this example > > http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm > > ___ -- 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] PBX deployment big problems: Voip traffic analysis
look - you MUST have a minimum of the following - a clear 24x7 graph of all you network segments show packet loss, packet delay for several weeks prior to 1st turn up. -Unless you have a 100% totally dedicated IP network for you voice, you must have Qos on every piece of hardware in the network, and you must test it to makes sure it works. As you turn up the service, in controlled stages you MUST measure network and systems. and fix any errors or bad trends in the data. Or you can say the hell with, just turn it up, and see what happens :) Bhrugu Mehta wrote: > hi, > Yes, there are many problem to implement and setup asterisk in a callcenter. > but , all these problem can be remove if you set up your hardware and > your LAN network > verywell. > Generaly, your server Configuration should be greater and your LAN also. > You have to use Proper Codecs for voice. Generaly , g729 is greater. > > regards, > Bhrugu Mehta > > > On 5/16/08, gincantalupo <[EMAIL PROTECTED]> wrote: > >> Hi, >> hope not to be OT :) >> after more than 3 years of PBX installations we can adfirm Asterisk is >> stable enough to be considered a good product but still we encounter a >> lot of problems when deploying a new PBX. It seems that the biggest >> problems are all networking related: one way voice (also inside a LAN), >> calls drops, etc... >> How do you face this kind of problems? Which diagnose tools/methods do >> you use? >> >> Thank you. >> >> Giorgio Incantalupo >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] playing .gsm sounds through a web browser
Does anyone know where I can a copy of sox for windows with mp3 built in ? Julian David Backeberg wrote: > No, no, no. > > Don't try to play them directly as gsm files. Convert them to wav on > the fly, when demanded by the user from the webpage. Have a php, or > perl, or whatever script call sox, and push the wav to the user. sox > runs so fast that you can do the conversion on-demand. You can decide > what to do with the wav file afterward. Delete it immediately, cache > it, whatever. Don't force the users to know anything they don't > already know. There are a lot of media players that can handle a wav > file. > > On Thu, May 15, 2008 at 7:26 PM, Julian Lyndon-Smith <[EMAIL PROTECTED]> > wrote: >> I have a lot of recordings from asterisk in a .gsm format. I would like >> to play these files from a web browser (IE, firefox and opera) >> >> What do I need to do in order to achieve this goal ? >> >> Thanks >> >> Julian >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX deployment big problems: Voip traffic analysis
On Fri, 2008-05-16 at 11:37 +0200, gincantalupo wrote: > How do you face this kind of problems? Which diagnose tools/methods do > you use? If it were me, I'd start by using tcpdump, Wireshark, and or ngrep to look at the network traffic and follow the signaling. Yes, it takes a little while to wrap your head around SIP and/or IAX2 conversations on the wire and what they're supposed to look like, but any time you invest will more than pay for itself later. Also, it's always a good idea to have a firm understand of your network. Are you measuring throughput on your switches? VoIP traffic tends to highlight network problems much quicker than typical network traffic, due to its high sensitivity to latency and jitter. If your network is poor, then you're really just implementing VoBN (Voice over Bad Network). -- Jared Smith Training Manager Digium, Inc. ___ -- 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] PBX deployment big problems: Voip traffic analysis
hi, Yes, there are many problem to implement and setup asterisk in a callcenter. but , all these problem can be remove if you set up your hardware and your LAN network verywell. Generaly, your server Configuration should be greater and your LAN also. You have to use Proper Codecs for voice. Generaly , g729 is greater. regards, Bhrugu Mehta On 5/16/08, gincantalupo <[EMAIL PROTECTED]> wrote: > Hi, > hope not to be OT :) > after more than 3 years of PBX installations we can adfirm Asterisk is > stable enough to be considered a good product but still we encounter a > lot of problems when deploying a new PBX. It seems that the biggest > problems are all networking related: one way voice (also inside a LAN), > calls drops, etc... > How do you face this kind of problems? Which diagnose tools/methods do > you use? > > Thank you. > > Giorgio Incantalupo > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] playing .gsm sounds through a web browser
I don't want to add to the list of supported players - my suggestion is to do the recordings as WAV, so you get the same quality and compression results but it's playable natively on any Windows box. Another choice to get this issue solved, if you have a system that is not 24/7 in use, why not running a nightly job to do recoding to voice-rate MP3? playable anywhere, good quality, small footprint. I have a makefile somewhere to do that, even starting from .gsm I think it's this one: http://astrecipes.net/index.php?n=294 l. On Fri, 16 May 2008 01:26:15 +0200, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote: > I have a lot of recordings from asterisk in a .gsm format. I would like > to play these files from a web browser (IE, firefox and opera) > > What do I need to do in order to achieve this goal ? > > Thanks > > Julian > -- Loway Research - Home of QueueMetrics http://queuemetrics.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] PBX deployment big problems: Voip traffic analysis
Hi, hope not to be OT :) after more than 3 years of PBX installations we can adfirm Asterisk is stable enough to be considered a good product but still we encounter a lot of problems when deploying a new PBX. It seems that the biggest problems are all networking related: one way voice (also inside a LAN), calls drops, etc... How do you face this kind of problems? Which diagnose tools/methods do you use? Thank you. Giorgio Incantalupo ___ -- 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] queue autopause
You could use a "real" agent with autologoff, for instance. Of course there may be drawbacks in using the agent channel, though it's usually a reasonable choice for most setups. Thanks l. On Fri, 16 May 2008 05:39:52 +0200, Rilawich Ango <[EMAIL PROTECTED]> wrote: > Hi all, > > There is a setting called autopause in queue.conf to pause a queue > member if they fail to answer a call. > The autopause setting will pause the agent even when they are on the > line. I want to know if it is possible to pause the queue member only > when they don't answer after timeout? > ango > -- Loway Research - Home of QueueMetrics http://queuemetrics.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-biz] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008
Hi Steve, we tested Druid for a while - our setup was pretty much call-center-ish, so we did not do much testing of hardware and advanced PBX functionalities, while we used it to build queues and trunks and extensions. Generally speaking, it looked quite polished and very easy to use (never needed to look up the docs); the dialplan code it produced was a bit leaner than the one FreePBX uses. From our point of view, we found it quite easy to build a reasonable call-center set up with it, so we produced a small guide on setting up QueueMetrics with it - you may want to have a look at it to see how Druid looks like: http://queuemetrics.com/download/QM_Druid_100.pdf Well worth trying out - though I personally have not yet found a reasonable replacement for vi, my favourite Asterisk configuration editor. Just my euro 0.02, l. On Wed, 14 May 2008 21:38:51 +0200, Steve Totaro <[EMAIL PROTECTED]> wrote: > This is *exactly* where I am. It installed fine on an HP DL380 and > Digium TDM400P I had laying around and looked good, but I am > interested in some real world testimonials. > > Thanks, > Steve Totaro -- Loway Research - Home of QueueMetrics http://queuemetrics.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] Not hearing first prompts
Stefan Guenther wrote: > Alan Lord wrote: > > >When I connect to various asterisk services such as VoicemailMain(), > >MeetMe() as examples, I do not get to hear the first greeting messages. > > > >I've tried adding a Wait(1) before or after the application but this > >seems to have no effect. > > > >Is there another setting/parameter I can play with to delay the start > >of the playback of these messages? > > > do you use SNOM phones? > > We had the same problem with a number of SNOM phones. > Unfortunately not. It is the same if we use our Siemens DECT/SIP handsets or the Ekiga softphone... I recall having this problem once before and that it "went away" when I changed from Ekiga to Twinkle. When I get chance, I will re-install Twinkle and see if that exhibits the same problem. Thanks anyway. Al -- The way out is open! http://www.theopensourcerer.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] Not hearing first prompts
Alan Lord wrote: >When I connect to various asterisk services such as VoicemailMain(), >MeetMe() as examples, I do not get to hear the first greeting messages. > >I've tried adding a Wait(1) before or after the application but this >seems to have no effect. > >Is there another setting/parameter I can play with to delay the start >of the playback of these messages? > do you use SNOM phones? We had the same problem with a number of SNOM phones. After we did an update to the current firmware version, we heard the beginning of the messages, too. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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] Problems passing variables from a macro
The macros is executed in the OUTBOUND call leg, which is different from the INBOUND that executes the rest of the dialplan. So variables you set in that call leg, stays in that channel (They're called "channel variables" since they have the channel as the scope). /O ___ -- 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] Newbie Asterisk: Install Asterisk as non-root
First of all, thanks Philipp, Alan, Tzafrir and James for your valuable comments. I have listed below the exact list of commands to run for reinstalling asterisk 1.4.* as non-root on a Redhat / Fedora distro. Hope others can benefit. I have the following comments/questions though: 1) #What is safe_asterisk used for actually? I did not touch it in my modification because I don't know when is it triggered? 2) #I do not actually know whether we really need to modify /etc/asterisk/asterisk.conf? Is this file read by asterisk at all? Seems like an important file name - asterisk.conf? 3) It is safer to define a user called asterisk in group asterisk unless you want to make more changes to 2 files i.e. zaptel.rules and /etc/init.d/asterisk 4) There is an additional chmod to run for letting voicemail.conf to be written by group asterisk. # /etc/init.d/asterisk stop Shutting down asterisk:[ OK ] # /usr/sbin/groupadd asterisk # /usr/sbin/useradd -d /var/lib/asterisk -g asterisk asterisk useradd: warning: the home directory already exists. Not copying any file from skel directory into it. # cp Makefile Makefile.org *** *** Change the following line from: *** *** ASTVARRUNDIR=$(localstatedir)/run *** *** to *** *** ASTVARRUNDIR=$(localstatedir)/run/asterisk *** # vi Makefile [...] ifeq ($(OSARCH),SunOS) ASTETCDIR=/var/etc/asterisk ASTLIBDIR=/opt/asterisk/lib ASTVARLIBDIR=/var/opt/asterisk ASTSPOOLDIR=/var/spool/asterisk ASTLOGDIR=/var/log/asterisk ASTHEADERDIR=/opt/asterisk/include ASTBINDIR=/opt/asterisk/bin ASTSBINDIR=/opt/asterisk/sbin ASTVARRUNDIR=/var/run/asterisk ASTMANDIR=/opt/asterisk/man else ASTETCDIR=$(sysconfdir)/asterisk ASTLIBDIR=$(libdir)/asterisk ASTHEADERDIR=$(includedir)/asterisk ASTBINDIR=$(bindir) ASTSBINDIR=$(sbindir) ASTSPOOLDIR=$(localstatedir)/spool/asterisk ASTLOGDIR=$(localstatedir)/log/asterisk ASTVARRUNDIR=$(localstatedir)/run/asterisk ASTMANDIR=$(mandir) [...] cd /usr/src/asterisk-1.4 make clean ./configure make make install *** *** Don't panic! *** /var/run/asterisk should just be an empty directory but should just exist. *** # chown --recursive asterisk:asterisk /var/lib/asterisk # chown --recursive asterisk:asterisk /var/log/asterisk # chown --recursive asterisk:asterisk /var/run/asterisk # chown --recursive asterisk:asterisk /var/spool/asterisk # chown --recursive asterisk:asterisk /usr/lib/asterisk # chown --recursive asterisk:asterisk /dev/zap # chmod --recursive u=rwX,g=rX,o= /var/lib/asterisk # chmod --recursive u=rwX,g=rX,o= /var/log/asterisk # chmod --recursive u=rwX,g=rX,o= /var/run/asterisk # chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk # chmod --recursive u=rwX,g=rX,o= /usr/lib/asterisk # chmod --recursive u=rwX,g=rX,o= /dev/zap # chown --recursive root:asterisk /etc/asterisk # chmod --recursive u=rwX,g=rX,o= /etc/asterisk # cp /etc/asterisk/asterisk.conf /etc/asterisk/asterisk.conf.org # vi /etc/asterisk/asterisk.conf *** *** Change the following line from: *** *** astrundir => /var/run *** *** to *** *** astrundir => /var/run/asterisk *** # cp /etc/init.d/asterisk /etc/init.d/asterisk.org # vi /etc/init.d/asterisk *** *** Uncomment the following line from: *** *** #AST_USER="asterisk" *** #AST_GROUP="asterisk" *** *** to *** *** AST_USER="asterisk" *** AST_GROUP="asterisk" *** *** *** Asterisk needs to write to voicemail.conf for password change. *** # chmod g+w /etc/asterisk/voicemail.conf *** *** Restart Asterisk by either of below: *** # /etc/init.d/asterisk restart # asterisk -U asterisk -G asterisk ___ -- 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] Citel Gateways
Terrible experience. VERY loud blast of static at the beginning of calls and pops and click throughout. The static was not limited to one box, all four had the same issue. The pops and clicks were determined to be only on extensions connected to one box that was RMAed and the replacement still blasted you with static. The amount of time I put into trying to get the Definity units working as expected far outweighed the cost of the Polycoms, not to mention taking me away from other work. This was replacing a Definity G3 about a year and a half ago. I finally returned the Citels and did what should have been done to begin with, replace the phones with new Polycoms and Ebay the old phones. Of course that was my recommendation from the start but when you have a semi tech savvy CEO, they like to google and second guess to save a buck, they buy all the "We are the leaders of) marketing crap. Thanks, Steve Totaro On Thu, May 15, 2008 at 9:40 PM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote: > On Thu, May 15, 2008 at 01:59:39PM -0500, Jonathan C. Bailey wrote: >> W're looking at using some Citel gateways to serve one of our sites >> (40 extensions, Toshiba phones). I've found that people seem to like >> the product from demos, but I was wondering how many have some of the >> gateways in production and if they seem to do the job for the long >> run. > > Check the list archives: I've seen polarized opinions both ways. > > Cheers, > -- jra > -- > Jay R. Ashworth Baylink [EMAIL > PROTECTED] > Designer The Things I Think RFC 2100 > Ashworth & Associates http://baylink.pitas.com '87 e24 > St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 > > Those who cast the vote decide nothing. > Those who count the vote decide everything. > -- (Joseph Stalin) > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk concurrent calls count
Hello, Alexander. AO> Hi Asterisk Users, AO> I'm interested in how many concurrent calls Asterisk can process without AO> troubles. I mean 1 Asterisk server (software) like either proxy or media AO> server (any numbers will be appropriate). AO> 1. Is there any limitations by the software? What is this number? AO> 2. What is the maximum count of concurrent calls you've ever seen/tested? Look at this example http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm -- Alexey mailto:[EMAIL PROTECTED] ___ -- 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] Problems passing variables from a macro
I haven't used ael but in extension.conf whenever we set a channel variable
we use a SET command just like you used it to set the variable MACRO_RESULT.
try using the set command, if still it does not work then try to initialize
the wrongpin=0 before the dial command, or outside the macro.
On Fri, May 16, 2008 at 12:36 PM, Tobias Ahlander <[EMAIL PROTECTED]>
wrote:
> Good day,
>
> I'm using a dial string as follows:
> Dial(SIP/${phonenumber},30,grM(screen^${pin})L(${30}[:6]));
> When I set a variable in the macro screen, it doesn't get passed back to
> the extension from where the dial was called. I can always put the result in
> the MySQL database, but that feels a bit overkill... the macro looks as
> follows:
>
> macro screen (arg1) {
>
> Wait(0.2);
> Read(acceptcall|sounds/pin|7);
> if(${acceptcall} = ${arg1}) {
> NoOp(connect them);
> wrongpin=0;
> } else {
> Set(MACRO_RESULT=CONTINUE);
> wrongpin=1;
> }
> NoOp(MACRO_RESULT = ${MACRO_RESULT});
>
> }
>
> This is the output from the CLI, and I can see that the wrongpin is set to
> 1, but when I do a NoOp right after leaving the macro, it says its empty...
>
> -- Executing [EMAIL PROTECTED]:36] Dial("SIP/1003-b7619b78",
> "SIP/1203|30|grM(screen^1234)L(30[:6])") in new stack
> -- Limit Data for this call:
>> timelimit = 30
>> play_warning = 6
>> play_to_caller = yes
>> play_to_callee = yes
>> warning_freq = 0
>> start_sound= (null)
>> warning_sound = beep
>> end_sound = beep
> -- Called 1203
> -- SIP/1203-08d62408 is ringing
> -- SIP/1203-08d62408 answered SIP/1003-b7619b78
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/1203-08d62408", "arg1=1234")
> in new stack
> -- Executing [EMAIL PROTECTED]:2] Wait("SIP/1203-08d62408", "0.2") in
> new stack
> -- Executing [EMAIL PROTECTED]:3] Read("SIP/1203-08d62408",
> "acceptcall|sounds/pin|7") in new stack
> -- Accepting a maximum of 7 digits.
> -- Playing 'sounds/pin' (language 'en')
> -- User entered '1'
> -- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/1203-08d62408", "0?5:8")
> in new stack
> -- Goto (macro-screen,s,8)
> -- Executing [EMAIL PROTECTED]:8] Set("SIP/1203-08d62408",
> "MACRO_RESULT=CONTINUE") in new stack
> -- Executing [EMAIL PROTECTED]:9] Set("SIP/1203-08d62408", "wrongpin=1")
> in new stack
> -- Executing [EMAIL PROTECTED]:10] NoOp("SIP/1203-08d62408", "Finish
> if-screen-32753") in new stack
> -- Executing [EMAIL PROTECTED]:11] NoOp("SIP/1203-08d62408",
> "MACRO_RESULT = CONTINUE") in new stack
> -- Executing [EMAIL PROTECTED]:37] NoOp("SIP/1003-b7619b78",
> "DIALSTATUS:ANSWER") in new stack
> -- Executing [EMAIL PROTECTED]:38] NoOp("SIP/1003-b7619b78", "wrongpin=")
> in
> new stack
>
> Is there a good way to pass this variable back to the context "connect"?
>
> Thanks,
> Best regards,
> Tobias
>
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--
Best Regards
Rizwan Hisham
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[asterisk-users] Asterisk concurrent calls count
Hi Asterisk Users, I'm interested in how many concurrent calls Asterisk can process without troubles. I mean 1 Asterisk server (software) like either proxy or media server (any numbers will be appropriate). 1. Is there any limitations by the software? What is this number? 2. What is the maximum count of concurrent calls you've ever seen/tested? -- Thanks in advance Alexander Olekhnovich ___ -- 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] Problems passing variables from a macro
Good day,
I'm using a dial string as follows:
Dial(SIP/${phonenumber},30,grM(screen^${pin})L(${30}[:6]));
When I set a variable in the macro screen, it doesn't get passed back to the
extension from where the dial was called. I can always put the result in the
MySQL database, but that feels a bit overkill... the macro looks as follows:
macro screen (arg1) {
Wait(0.2);
Read(acceptcall|sounds/pin|7);
if(${acceptcall} = ${arg1}) {
NoOp(connect them);
wrongpin=0;
} else {
Set(MACRO_RESULT=CONTINUE);
wrongpin=1;
}
NoOp(MACRO_RESULT = ${MACRO_RESULT});
}
This is the output from the CLI, and I can see that the wrongpin is set to
1, but when I do a NoOp right after leaving the macro, it says its empty...
-- Executing [EMAIL PROTECTED]:36] Dial("SIP/1003-b7619b78",
"SIP/1203|30|grM(screen^1234)L(30[:6])") in new stack
-- Limit Data for this call:
> timelimit = 30
> play_warning = 6
> play_to_caller = yes
> play_to_callee = yes
> warning_freq = 0
> start_sound= (null)
> warning_sound = beep
> end_sound = beep
-- Called 1203
-- SIP/1203-08d62408 is ringing
-- SIP/1203-08d62408 answered SIP/1003-b7619b78
-- Executing [EMAIL PROTECTED]:1] Set("SIP/1203-08d62408", "arg1=1234") in
new stack
-- Executing [EMAIL PROTECTED]:2] Wait("SIP/1203-08d62408", "0.2") in new
stack
-- Executing [EMAIL PROTECTED]:3] Read("SIP/1203-08d62408",
"acceptcall|sounds/pin|7") in new stack
-- Accepting a maximum of 7 digits.
-- Playing 'sounds/pin' (language 'en')
-- User entered '1'
-- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/1203-08d62408", "0?5:8") in
new stack
-- Goto (macro-screen,s,8)
-- Executing [EMAIL PROTECTED]:8] Set("SIP/1203-08d62408",
"MACRO_RESULT=CONTINUE") in new stack
-- Executing [EMAIL PROTECTED]:9] Set("SIP/1203-08d62408", "wrongpin=1")
in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp("SIP/1203-08d62408", "Finish
if-screen-32753") in new stack
-- Executing [EMAIL PROTECTED]:11] NoOp("SIP/1203-08d62408", "MACRO_RESULT
= CONTINUE") in new stack
-- Executing [EMAIL PROTECTED]:37] NoOp("SIP/1003-b7619b78",
"DIALSTATUS:ANSWER") in new stack
-- Executing [EMAIL PROTECTED]:38] NoOp("SIP/1003-b7619b78", "wrongpin=") in
new stack
Is there a good way to pass this variable back to the context "connect"?
Thanks,
Best regards,
Tobias
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