Re: [asterisk-users] Sending SMS from Asterisk
The same ones which, by total coincidence, you just advertised on asterisk-biz, perhaps? What are the chances of that? On 14/02/07, Sam Tam [EMAIL PROTECTED] wrote: Drop me an email I know some GSM Gateway that has a direct serial port for SMS Sam -Original Message- From: Jon Pounder [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 14, 2007 10:50 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending SMS from Asterisk Quoting Patrick [EMAIL PROTECTED]: On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote: Singer Wang wrote: by your .ca address I assume your in Canada.. both Telus and Rogers have a email-to-SMS gateway... Well, those are notoriously unreliable. I've had messages take hours to arrive when sent by the email-to-SMS gateway. I was kinda hoping for something more direct. Rogers prioritizes internal SMS messages over e-mailed ones. we do this with the vmobile.ca gateway (which is just using the actual bell cellular network), and only a handful of times in several years hasn't it been instant. I get the sms before my desktop mail reader has even picked up the same messages in most cases. What I'd like is some kind of SMSC -- or something that accomplishes the same thing. Maybe http://www.kannel.org/ provides some useful info. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and chan_misdn
Hi list, I installed a fresh Debian/Etch with Asterisk 1.4 and Zaptel 1.4 from SVN for 2 Digium B410P card. I ran configure in Asterisk dir, went in zaptel dir and: make, make install, make b410p. Everything is ok. Now I want to compile Asterisk but can't activate the chan_misdn channel which depends on -from menuselect- isdnnet(E), misdn(E), suppserv(E) When I made the make b410p, all the misdn stuff was downloaded from digium's ftp. Also, running /etc/init.d/misdn-init --scan show me the 2 cards I have, /etc/init.d/misdn-init --config prepare me the misdn.conf and after a /etc/init.d/misdn-init start I see: mISDN_dsp 191656 0 mISDN_capi 88716 0 mISDN_l2 34452 0 mISDN_l1 11036 0 mISDN_core 71360 6 mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 kernelcapi 44576 2 mISDN_capi,capi My questions: why Asterisk doesn't want to let me activate the misdn channel? Is misdn ready for 1.4? Thanks for any hint -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
Try to use SMS Server tools its an SMS Gateway software which can send and receive short messages through GSM modems and mobile phones. Its very useful for sending SMS alerts, we used it with Nagios and Asterisk as well. Best Regards, Joanna Liza Mariazeta On 2/14/07, Stephen Bosch [EMAIL PROTECTED] wrote: Singer Wang wrote: by your .ca address I assume your in Canada.. both Telus and Rogers have a email-to-SMS gateway... Well, those are notoriously unreliable. I've had messages take hours to arrive when sent by the email-to-SMS gateway. I was kinda hoping for something more direct. Rogers prioritizes internal SMS messages over e-mailed ones. What I'd like is some kind of SMSC -- or something that accomplishes the same thing. Any ideas? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
I forgot to mention that you have to get a GSM modem or a compatible phone to send the messages, with this set up you dont have to worry about negotiating with carriers. It will be just like a normal user sending SMS to one another. On 2/14/07, joannaliza mariazeta [EMAIL PROTECTED] wrote: Try to use SMS Server tools its an SMS Gateway software which can send and receive short messages through GSM modems and mobile phones. Its very useful for sending SMS alerts, we used it with Nagios and Asterisk as well. Best Regards, Joanna Liza Mariazeta On 2/14/07, Stephen Bosch [EMAIL PROTECTED] wrote: Singer Wang wrote: by your .ca address I assume your in Canada.. both Telus and Rogers have a email-to-SMS gateway... Well, those are notoriously unreliable. I've had messages take hours to arrive when sent by the email-to-SMS gateway. I was kinda hoping for something more direct. Rogers prioritizes internal SMS messages over e-mailed ones. What I'd like is some kind of SMSC -- or something that accomplishes the same thing. Any ideas? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
On 14/02/07, Stephen Bosch [EMAIL PROTECTED] wrote: Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere (they are not interested in helping an individual customer and technical staff Tiers I and II have no idea what I am talking about). Are there SMS aggregators that I could use for sending messages to this particular phone over the Internet? There are plenty - I've used 2sms.com, clickatell.com and csoft.co.uk. bayhamsystems.com have a service tailored for Asterisk users. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
On Tue, 13 Feb 2007, Vincent Delporte wrote: At 10:09 11/02/2007 -0500, Gordon Henderson wrote: Check the processor spec. carefully. [...] Also make sure you compile asterisk for an i586 OK, I'll make sure it has enough cache and I'll recompile the code myself. I'm thinking of getting an ML 8000 http://via.com.tw/en/products/mainboards/motherboards.jsp?motherboard_id=301 . At 10:09 11/02/2007 -0500, Manny A. Wise wrote: I did, and I was NOT happy with the results... Mini-itx have a serious problems with IRQ sharing... I am happily using a embeded system now, but the FXO and FXS have to be external. Those boards only come with one PCI slot. Do you mean it could share an IRQ with some embedded component like the video card? On the CN1000 boards I'm using, the PCI slot seems tobe locked to IRQ10. The on-board USB hardware also seems to be wired to IRQ 10 )-: Using the BIOS to reserve IRQ 10 caused the on-board USB hardware to move to IRQ5 on the old VIA 533MHz boards I use for RD, but not on the new CN1000 boards. You'll need to experiment with this on the EX board... So I disable the on-board USB device, and have a custom compiled kernel that doesn't include USB drivers. However, on a test board, I did leave USB enabled with a kernel that supproted USB just to test - an - well - it just works - however I only planned to use USB to perform an upgrade, so the times it would be in-use would be so minimal as to (hopefully) not have an issue. On an older 533MHz board: $ cat /proc/interrupts CPU0 0: 48124962 XT-PIC timer 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd:usb1, uhci_hcd:usb2 7: 1 XT-PIC acpi 8: 4 XT-PIC rtc 11: 75120 XT-PIC eth0 12: 48084364 XT-PIC wctdm 14: 2763 XT-PIC ide0 15: 5373 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 $ /sbin/zttest Opened pseudo zap interface, measuring accuracy... ... --- Results after 42 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.995350 BTW, in this age of big USB drive, I don't really nee a DVD/CDRW combo. Does someone know if the Via motherboards (at least the ML series) supports booting off a USB drive, so I can use this to start Linux and fetch install files from an FTP server? I've not tried it (I boot them off a flash IDE device I create on a host system), but can't you just temporarily plug in a CD drive to do the install (onto a local IDE/SATA drive) then unplug it put the lid back on? Thats how I build some of my servers... (Although the CD drive is an IDE drive these days for speed...) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with safe_asterisk
On Tue, 13 Feb 2007, Tzafrir Cohen wrote: On Tue, Feb 13, 2007 at 01:35:50PM +0100, Andrea De Vita wrote: Hi all, I have installed some Asterisk machine, all with the same problem. My typical configuration is: - Asterisk 1.2.14 (or 1.4.0beta3) - CentOS 4.4 server. The problem is this: When I start Asterisk with the default init script (/etc/init.d/asterisk start) distributed with the source, and kill (or kill -9) Asterisk-pid, then safe_asterisk doesn't correctly work (it dies and not restart Asterisk). Instead, if I start Asterisk with safe_asterisk command from shell, after kill Asterisk-pid, safe_asterisk restart Asterisk correctly. I would use the init script because I like to use Linux-HA that require this. Edit that init.d script lightly not to use safe_asterisk. safe_asterisk is not close to robust anyway, and thus will only complicate things. Seconded. Have a look at this: http://www.drogon.net/init.d.asterisk Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway promotion from £69GBP
On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote: Hello All This month we would like to offer our GSM Gateway range for less to clear up some spaces. etc Perhaps, you could explain what is NON COMMERCIAL about your post. I would not buy anything from a spammer. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small CDR Billing Program
Hi Mark, there are a lot of open source CDR billing program, try this link http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems. But based on your explanation, it looks simple enough, why not just create one that will suit your needs. Those billing programs out there has a lot of features that some of them you dont really need. What we did was just to create our own billing system, since we have a different computation. Best Regards, Joanna Liza Mariazeta www.mariazeta.com On 2/13/07, Roland Ndaka Fru [EMAIL PROTECTED] wrote: Hi Mark, Take a look at the YakaVOIP solution from http://www.yakasoftware.com. Probably suits your requirements. Greetz, Roland. -- *Von:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *Im Auftrag von *MBIT Technologies *Gesendet:* 12 February 2007 22:23 *An:* asterisk-users@lists.digium.com *Betreff:* [asterisk-users] Small CDR Billing Program Hi Guys I am just looking around for a small billing program but can't really find what I am looking for. It needs to bill straight off the CDR. It should grab all the CDR records from the asteriskcdrdb mysql database then have a rates table to that it calculate a bill from. Is there any open source packages or commercial packages that will account for billing say only 5 extensions? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Following call forwards
I have a challenge that is ending up quite interesting. I need to
identify which SIP phone touched a call last, that is, which phone did
the last transfer or dialed the original call if no transfers were
done.
It is easy in the case of a regular, non-transfered call. Just put
something in callerid= in sip.conf, and that will show up in
${CALLERID}. The same with an attended transfer, since that is just
another outgoing call which gets bridged later.
Unattended transfers are not so difficult (at least not if the
asterisk version is reasonably new), because ${BLINDTRANSFER} is set,
and I can get the phone name from that.
It is much more difficult if call forwarding is set on the phone. Then
it just replies to requests with 302 Moved Temporarily. In that case
${BLINDTRANSFER} is not set. ${RDNIS} is set, but it is not always
easy or even possible to turn the dialed number into a phone ID.
Is there a variable I can check to see which phone did the redirect?
If you are asking yourself why I care, here is the (long) background
story:
In Sweden, there are local calls, national calls, and international
calls. National calls are prefixed with 0, international ones with 00.
Those are easy to handle.
Local calls start by [1-9], and I need to massage them into national
calls. There are several locations connected by IP, and they each
need to be able to dial local numbers, but the calls all exit at the
same location and the users would get all confused if their local
calls from Malmö end up connecting to a phone in Stockholm.
It's not so difficult in that specific case. Add context=frommalmoe to
sip.conf, and do something like this:
[frommalmoe]
exten = _Z.,1,Goto(outgoing,032${EXTEN},1)
exten = _0.,1,Goto(outgoing,${EXTEN},1)
(I have no idea what the real prefix for Malmö is, this is just an
example).
When you have phones all over the country it gets complicated though.
You need a context for each area code, and that gets unwieldy. It is
much easier to have a database of phones and their locations -- but
that does not work if someone sets their phone to forward calls to a
local number. I have no way to find out which area that number belongs
to. If I knew which phone did the transfer, I would know which area to
use.
/Benny
(I'm sure someone will now give me the solution in just one line)
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Re: [asterisk-users] SMS via VoIP and web
On Tue, 13 Feb 2007, Steve Kennedy wrote: On Wed, Feb 14, 2007 at 07:17:32AM +0800, Ronald Wiplinger wrote: Where can I get a starting point for setting up sms via VoIP and via web. I want to send SMS from VoIP or web to VoIP phones and GSM phones. 1. how to set-up? 2. which smsc should I use? (what is the price?) 3. which phones can be used? Some telcos support sending SMS down phone lines, it's reasonably common in Europe and there's an ETSI spec for it. However it's probably easier to use something like Kannel which has an http interface and then either connect that to an SMSC or locally through a GSM terminal (phone). SMSC connections and pricing will vary depending on what country you're in. As a small customer (in the UK at least) it's unlikely you'd get an connection to an operator's SMSC and you'd have to go through an aggregator. Best I got (in the UK) was 3p a text when I looked nearly 2 years ago. This was to an outfit in Scotland - you connect via an internet enabled API (ie. a bit of PHP code) and send the text via their service centre (or texts - I was looking at 2-4000 a month, but they went up to half a million a month). What I do these days is use a Siemens GSM modem - with a SIM card in it on a PAYG tarrif (my needs are minimal right now) Linux supports sending SMS messages via getsms and putsms (which works in the pager section of voicemail.conf) Sending TXT messages to a VoIP phone is going to be challenging, I think, but I'd like to think it is (or might be) possible! Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can anyone help me out with Polycom 2.1 firmware please?
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Re: [asterisk-users] Dial out from AGI
Hi Roy,
If its perl script,you can try this.
use Asterisk::AGI;
our $AGI = new Asterisk::AGI;
$AGI-EXEC('Dial', 'Zap/g2/8005551212');
On 2/11/07, Roy Kidder [EMAIL PROTECTED] wrote:
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look something like this (pseudo-code
follows):
if ($a){
dial(8005551212);
}else{
dial(866555);
}
The part I can't seem to get right is the dial function. I tried to
mimic the dial plan like so
sub dial($number){
print Dial(\Zap/1-1\, \Zap/g2/$number\)\n;
}
but I get the error
handle_exec: Could not find application (Dial(Zap/1-1,Zap/g2/866555)
Anyone have any suggestions?
Thanks,
Roy
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Re: [asterisk-users] SMS via VoIP and web
On 07:17, Wed 14 Feb 07, Ronald Wiplinger wrote: Where can I get a starting point for setting up sms via VoIP and via web. I want to send SMS from VoIP or web to VoIP phones and GSM phones. 1. how to set-up? 2. which smsc should I use? (what is the price?) 3. which phones can be used? We use bayhamsystems for sms. Works great. SMS to voip phones is something we did not explore yet. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with asterisk AGI
Hi there,
to give you an idea of what Jon is saying...
in your extensions.conf you can probably try this..
exten =
1,1,BackGround(/var/lib/asterisk/sounds/TEXX-JP-WAV-8000/TEXX-JP-7-welcome)
exten = 1,2,SayDigits(${CALLERIDNUM})
exten = 1,3,AGI(checkRegist.agi,${CALLERIDNUM})
exten = 1,4,GotoIf($[${ISREGISTERED} =
0]?texx-nihonggo-temp-regt|readnum|1:texx-nihonggo-regt-menu|readnum|1)
exten = h,1,Hangup
then in you agi script...
$sql = select status from phone where phonenumber = ? and status '1';
$sth = $dbh-prepare($sql);
$sth-execute($phonenumber);
$ret = $sth-rows();
if ($ret 0) {
$AGI-set_variable('ISREGISTERED', '1');
exit;
} else {
$AGI-set_variable('ISREGISTERED','0');
}
Hope that helps..
Best Regards,
Joanna Liza Mariazeta
www.mariazeta.com
On 2/8/07, prasanth [EMAIL PROTECTED] wrote:
I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I
execute AGI in java which plays few wav files depending on external
parameters.
Can I have a dial plan inside my AGI? If not, how do I accomodate user
who needs to reach extension 2 from my agi? I have tried stream file and
get data but the two commands did not work at all.
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Re: [asterisk-users] Sending SMS from Asterisk
Hi, On Tue, 2007-02-13 at 18:04 -0700, Stephen Bosch wrote: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? you can also use a gsm card. the vgsm card allows sending sms from the AMI, along with full charset support (even cirillic!), sms reports, multipage sms and so on... you can check out http://open.voismart.it/index.php/VGSM_SMS or http://open.voismart.it/index.php/VGSM_Manager_Interface matteo. -- Matteo Brancaleoni RD Director Tel :+39.02.70633354 Voip :sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SMS via VoIP and web
I have used UTStarcom F3000 phone to send an SMS message to another phone of same model and that worked fine. I have also sent a SMS from F3000 to Snom 360 phone over sip which worked fine but snom phones doesn't have any editor to send outbound SMS. Ahsan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: 14 February 2007 09:55 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SMS via VoIP and web On 07:17, Wed 14 Feb 07, Ronald Wiplinger wrote: Where can I get a starting point for setting up sms via VoIP and via web. I want to send SMS from VoIP or web to VoIP phones and GSM phones. 1. how to set-up? 2. which smsc should I use? (what is the price?) 3. which phones can be used? We use bayhamsystems for sms. Works great. SMS to voip phones is something we did not explore yet. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] genzaptool from xorcom
hi every body; i installed zaptel 1.4,libpri 1.4, asterisk 1.4, asterisk-addons 1.4 succefuly, but i can't find the command genzaptelconf, so i tink to install handy zaptel toolset please can someone tell me whitch package goes with zaptel 1.4, i consult http://updates.xorcom.com/rapid/pool/main/z/zaptel/; but ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with T.38
Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax IP Asterisk IP Patton M-ATA Analog Fax 2 I tried Analog Fax 2 - Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have some errors : [Feb 14 11:28:55] WARNING[10547]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-mg-02-006f37f0(256) to SIP/0625037998-006de430(8) [Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 14 11:29:05] NOTICE[10547]: rtp.c:772 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xx.xx.xx.xx In my SIP.conf file: [general] context=default ; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all; First disallow all codecs allow=g729 allow=gsm allow=alaw ; Allow codecs in order of preference dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes I have the definition of the phone in DB. voip-test-01*CLI sip show peer 0625037998 voip-test-01*CLI * Name : 0625037998 Realtime peer: No Secret : Set MD5Secret: Not set Context : sipresidential Subscr.Cont. : Not set Language : fr AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 6 Dynamic : Yes Callerid : 0625037998 0625037998 MaxCallBR: 384 kbps Expire : -1 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : inband LastMsg : 0 ToHost : Addr-IP : (Unspecified) Port 0 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 0625037998 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw:20,ulaw:20) Auto-Framing: No Status : UNKNOWN Useragent: Patton Smartlink MATA 4.01.001 OE EN MA (0412)00a0ba01a154 Reg. Contact : sip:[EMAIL PROTECTED]:5060 Thanks a lot for your help, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CME integration using h323
Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323.
Cisco conf:
dial-peer voice 8 voip
destination-pattern 2...
session target ipv4:asterisk ip
codec g711alaw
no vad
h323.conf
[general]
port = 1720
bindaddr = 0.0.0.0
;tos=lowdelay
;
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=from-internal
extension.conf
[from-internal]
exten = _1XXX,1,Dial(SIP/${EXTEN}@cme ip)
exten = 2000,1,Dial(SIP/2000)
I'm able from Asterisk to call ip phone connected to cme but from cme to
asterisk the phones ring but go in hangup immediatly.
My debug:
---
localhosAnswering call ip$192.168.99.2:53716/21
localhos-- Transmitting RFC2833 on payload 101
localhos-- Received Facility message...
localhos-- Received Facility message...
localhos-- Inbound RFC2833 on payload 101
localhos-- Received RELEASE COMPLETE message...
localhos-- ClearCall: Request to clear call with token
ip$192.168.99.2:53716/21, cause 22
localhos-- Sending RELEASE COMPLETE
localhost*CLI channelsOpen = 1
channelsOpen = 0
localhos-- ClearCall: Request to clear call with token
ip$192.168.99.2:53716/21, cause 7
Scheduling destruction of call
'[EMAIL PROTECTED]' in 32000 ms
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port
to send to
set_destination: set destination to 192.168.99.122, port 5060
Reliably Transmitting (no NAT) to 192.168.99.122:5060:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.99.254:5060;branch=z9hG4bK1a8c7be3;rport
From: 1003 sip:[EMAIL PROTECTED];tag=as769a2c55
To: sip:[EMAIL PROTECTED]:5060;tag=1473512925
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
-- Call with Enrico [192.168.99.2] completed (22)
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
localhost*CLI
-- SIP read from 192.168.99.122:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.99.254:5060;branch=z9hG4bK1a8c7be3;rport
From: 1003 sip:[EMAIL PROTECTED];tag=as769a2c55
To: sip:[EMAIL PROTECTED]:5060;tag=1473512925
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
Server: X-Lite release 1105d
Content-Length: 0
--- (9 headers 0 lines) ---
Destroying call '[EMAIL PROTECTED]'
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection:
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
localhos== H.323 Connection deleted.
I don't understand why the call goes down only from cisco to
asterisk any ideas?
Thanks Enrico
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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[asterisk-users] asterisk 1.4 is stable?
since my update from 1.2.4 i am experiencing some problem with misdn ( mISDN_rdata: rport queue overflow 256/25 kernel error) and channels instability. Asteisk 1.4 is stable or should i downgrade to 1.2.x? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
On 14 Feb 2007, at 10:08, matteo brancaleoni wrote: Hi, On Tue, 2007-02-13 at 18:04 -0700, Stephen Bosch wrote: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? you can also use a gsm card. the vgsm card allows sending sms from the AMI, along with full charset support (even cirillic!), sms reports, multipage sms and so on... you can check out http://open.voismart.it/index.php/VGSM_SMS or http://open.voismart.it/index.php/VGSM_Manager_Interface matteo. We've used www.Simplewire.com , they have a x86 linux executable which we wrap in a shell script and call from the dialplan with a System() call. We've been happy with them for years. Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway promotion from£69G BP
Spamming aside, you can buy these cheaper from a ebay seller in Germany - including post - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 14, 2007 8:41 AM Subject: Re: [asterisk-users] GSM Gateway promotion from£69GBP On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote: Hello All This month we would like to offer our GSM Gateway range for less to clear up some spaces. etc Perhaps, you could explain what is NON COMMERCIAL about your post. I would not buy anything from a spammer. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway promotion from £69GBP
On 14/02/07, Dave Cotton [EMAIL PROTECTED] wrote: On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote: Hello All This month we would like to offer our GSM Gateway range for less to clear up some spaces. etc Perhaps, you could explain what is NON COMMERCIAL about your post. He does this all the time, and never bothers to respond to objections. Doesn't answer questions about how he mis-describes his products, either. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
On Wed, Feb 14, 2007 at 10:27:03AM +, younss azzayani wrote: hi every body; i installed zaptel 1.4,libpri 1.4, asterisk 1.4, asterisk-addons 1.4 succefuly, but i can't find the command genzaptelconf, so i tink to install handy zaptel toolset please can someone tell me whitch package goes with zaptel 1.4, i consult http://updates.xorcom.com/rapid/pool/main/z/zaptel/; but It is included as part of latest zaptel. wget http://svn.digium.com/svn/zaptel/branches/1.4/xpp/utils/genzaptelconf wget http://svn.digium.com/svn/zaptel/branches/1.4/xpp/utils/genzaptelconf.8 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
should i use them bouth ? or just select one of them ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
On Wed, Feb 14, 2007 at 11:24:15AM +, younss azzayani wrote: should i use them bouth ? or just select one of them man ./genzaptelconf.8 will give you the documentation of genzaptelconf . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
when i type : # ./genzaptelconf i got an error line 0: Unable to open master device '/dev/zap/ctl' Why ? :-( any idea please? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
On Wed, Feb 14, 2007 at 03:42:23AM +0100, Patrick wrote: On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote: Singer Wang wrote: by your .ca address I assume your in Canada.. both Telus and Rogers have a email-to-SMS gateway... Well, those are notoriously unreliable. I've had messages take hours to arrive when sent by the email-to-SMS gateway. I was kinda hoping for something more direct. Rogers prioritizes internal SMS messages over e-mailed ones. What I'd like is some kind of SMSC -- or something that accomplishes the same thing. Maybe http://www.kannel.org/ provides some useful info. Kannel is a pretty mature solution, it will drive a local GSM terminal or connect through to SMSC's using standard protocols (SMPP, CIMD, UCP/EMI etc) or even http/SOAP. Terminals such as Siemens TC/MC35, Wavecomm, Falcom etc seem to work well and Kannel tends to have driver modules for them, also many phones can also work. Make sure SIM buffering isn't used or you'll wear out the SIM (they have limited writes). Most operators wont allow direct connectivity unless you delivering 10's of millions of SMSs per month and you'll have to go through an aggregator. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway promotion from ?69GBP
On Wed, Feb 14, 2007 at 09:41:32AM +0100, Dave Cotton wrote: On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote: Hello All This month we would like to offer our GSM Gateway range for less to clear up some spaces. etc Perhaps, you could explain what is NON COMMERCIAL about your post. I would not buy anything from a spammer. Because Sam likes to do this about once per month. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Got SIP response 482 Loop Detected
Thanks Stephen - I will do this ... Mohamed Farid ,, Telecommunication Security Section Head ,, Mediterranean Smart Cards Company ,, 92 Tahreer Street. Dokki / Cairo / Egypt Website: www.mscc.com.eg Email : [EMAIL PROTECTED] Phone : +2 02 3331439/+2 02 3331400 Fax : +2 02 7621164 Mobile : +2 0122258350 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, February 13, 2007 2:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Got SIP response 482 Loop Detected Mohamed Farid wrote: On 2/12/07, Mohamed Farid [EMAIL PROTECTED] wrote: I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten = 558,1,Answer exten = 558,2,Playback(message.wav) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error : -- Called [EMAIL PROTECTED] -- Got SIP response 482 Loop Detected back from CallManager -- Now forwarding SIP/CallManager-097b3dc0 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/CallManager-1781) == Everyone is busy/congested at this time (1:0/0/1) How can I overcome this ... First start a fresh thread rather than replying to a different one. In other words: Don't pick a message, hit reply, and then rewrite the subject line. Instead - Click New Message, write a fresh subject line, and put the asterisk-users list address in the To: field. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This e-mail (including attachments) is classified as Mediterranean Smart Cards Company confidential and proprietary information The recipient hereby is committed to hold in strict confidence the contents of this (e-mail, document, and information) and not to disclose to any third party without the prior written consent of Mediterranean Smart Cards Company. Recipient will be held liable for any unauthorized disclosure. It is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please notify the sender by return e-mail and delete the message in its entirety, including any attachments * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP response 482 Loop Detected
I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten = 558,1,Answer exten = 558,2,Playback(message.wav) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error : -- Called [EMAIL PROTECTED] -- Got SIP response 482 Loop Detected back from CallManager -- Now forwarding SIP/CallManager-097b3dc0 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/CallManager-1781) == Everyone is busy/congested at this time (1:0/0/1) How can I overcome this ... Mohamed Farid ,,, * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This e-mail (including attachments) is classified as Mediterranean Smart Cards Company confidential and proprietary information The recipient hereby is committed to hold in strict confidence the contents of this (e-mail, document, and information) and not to disclose to any third party without the prior written consent of Mediterranean Smart Cards Company. Recipient will be held liable for any unauthorized disclosure. It is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please notify the sender by return e-mail and delete the message in its entirety, including any attachments * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
On Wed, Feb 14, 2007 at 12:02:31PM +, younss azzayani wrote: when i type : # ./genzaptelconf i got an error line 0: Unable to open master device '/dev/zap/ctl' Why ? :-( any idea please? The default mode of operation is to detect your currently loaded modules. This does not even require restarting Asterisk. genzaptelconf -d (also consider the options -s and -v) unloads all currently-loaded modules, and then probes them one by one. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] To jitter buffer or not to jitter buffer?
Greetings list, Some time ago (probably about a year ago now) we disabled IAX jitter buffering on all our boxes because it was causing issues in a mixed 1.0 and 1.2 environment. One thing I've noticed over the last few months as more and more clients have moved from the 512k/1mb/2mb ADSL connections they were using onto up to 8mb connections is that whilst overall throughput is a lot better, the connections do seem to be more variable and have a tendency to stutter somewhat even with very little load on them. As a result, I'm considering reintroducing jitter buffering on our boxes now that everything's running 1.2 thoughout. Are there any pearls of wisdom out there on 1) whether enabling the jitter buffer is a good idea, and 2) what the recommended settings would be on an ADSL connection? I know that configuration is going to be a bit of a black art, as I'd imagine the best settings will be different for different users, but a starting point that folks have found working well over low-cost ADSL connections would be much appreciated. Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To jitter buffer or not to jitter buffer?
Chris Bagnall wrote: Greetings list, Some time ago (probably about a year ago now) we disabled IAX jitter buffering on all our boxes because it was causing issues in a mixed 1.0 and 1.2 environment. One thing I've noticed over the last few months as more and more clients have moved from the 512k/1mb/2mb ADSL connections they were using onto up to 8mb connections is that whilst overall throughput is a lot better, the connections do seem to be more variable and have a tendency to stutter somewhat even with very little load on them. As a result, I'm considering reintroducing jitter buffering on our boxes now that everything's running 1.2 thoughout. Are there any pearls of wisdom out there on 1) whether enabling the jitter buffer is a good idea, and 2) what the recommended settings would be on an ADSL connection? I know that configuration is going to be a bit of a black art, as I'd imagine the best settings will be different for different users, but a starting point that folks have found working well over low-cost ADSL connections would be much appreciated. Thanks in advance. Regards, Chris Hi, not really a pearl of wisdom, but using JB on IAX with trunking seems to cause a few problems. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS via VoIP and web
Am Mittwoch, den 14.02.2007, 07:17 +0800 schrieb Ronald Wiplinger: Where can I get a starting point for setting up sms via VoIP and via web. I want to send SMS from VoIP or web to VoIP phones and GSM phones. 1. how to set-up? 2. which smsc should I use? (what is the price?) 3. which phones can be used? If you want to have SMS service for SMS/landline capable phones connected to your Asterisk, the SMS application will be your friend. As that works all-internal, it is of course free. The handling of incoming SMS from the world, as well as outbound SMS, is far more tricky, and usually requires an outside gateway. I use the landline SMS facility of several Siemens gigaset phones connected to my Asterisk server through e.g. AVM boxes, but only for internal messages. This works fine, after a bit dialplan hacking. You will need to Wait() for at least 2 seconds before the actual SMS command can take over, else you will have lots of failed calls. This at least applies to my Siemens/AVM combination. Reliable after inserting a 2 second wait. For anything else, I prefer using my mobile phone, just thinking about those 100 free texts a month that I never manage to use up :) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FRITZ!Box Fon ata
Am Dienstag, den 13.02.2007, 21:41 + schrieb Razza: Hi all, is it possible to to dumb down a FRITZ!Box Fon ata (http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html##) and have the two FXS ports AND the ISDN interface register with Asterisk. In much the same way a sipura SPA3K works? In short terms: NO. The FritzBox devices are not planned to connect their landline side to a VoIP provider (the 2x analogue will work fine). You could of course activate a immediate call-redirect of incoming calls to a VoIP number (which would then create a call on the asterisk side), and for outgoing, you could make use of the call-through feature. Both solutions are hacks only because most probably you will lose the ISDN features on the call, as well as the callers number which will not be transmitted. About the latter, this number can be retrieved if you allow for syslog messages in the Fritzbox, send them through your Linux box and filter out the incoming call information. Still a hack though. BTW: Most Fritzboxes seem to have a separate analogue and ISDN landline logic, with a Y-cable you can use both simultaneously. Google for that if it helps you any further. Outgoing line would be selected by *10# or *11x# respectively, IIRC. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33
Hi Demuel, 1st. Do you have a card that support 'zttranscode' ? if Yes, go ahead with this, if no, there's no use for you to be compiling it. 2nd. Do NOT do this: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb Do this [EMAIL PROTECTED]:/usr/src/linux/include/linux$ updatedb (as root, or you will have to permit that command 'updatedb' in the sudoers list for the user 'demuel', in ur case) then: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ slocate page-flags.h If u dont have it (slocate will certainly finds it if u do), then try to get it (of course, not just that file cause you could be missing another one in the farther process of compilation). Try to find out of what package or source where that file belongs to, and get it... J. Espinal [EMAIL PROTECTED] wrote: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ pwd /usr/src/linux/include/linux [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb make: *** No rule to make target `updatedb'. Stop. [EMAIL PROTECTED]:/usr/src/linux/include/linux$ ls -la page-flags.h /bin/ls: page-flags.h: No such file or directory [EMAIL PROTECTED]:/usr/src/linux/include/linux$ Did i missed something down here? Weird thing is, even a fresh install of slackware produced the same kind of error. Actually, it used to be working about a week before I made a source upgrade. Any thoughts? Regards, Demuel On Tue, Feb 13, 2007 at 02:23:21PM +, J. Espinal wrote: make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. J. Espinal, [EMAIL PROTECTED] wrote: Anybody, I have download asterisk 1.4 via svn. whem I compiled it, I got the following error: /lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules zttranscode.c:37:30: linux/page-flags.h: No such file or directory make[1]: *** [zttranscode.o] Error 1 make[1]: Leaving directory `/home/kingkong/code/projects/asterisk/source/zaptel-1.4' make: *** [all] Error 2 make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. Better yet: simply don't build zttranscode, unless you have a card that actually supports it... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bandwidth shapping device
I would use a Mikrotik - www.mikrotik.com Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, February 14, 2007 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MRTG with 4 graphs
How can I set-up a MRTG with 4 graphs, whereby: 1 data in 2 data out 3 ONLY voice(/video) data in 4 ONLY voice(/video) data out bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth shapping device
Quoting Ronald Wiplinger [EMAIL PROTECTED]: I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) install openbsd on some old hardware with 2 x nics in it. use a bridge configuration with no ips, and use the pf traffic shaping rules to split it up however you want. you don't have to just dedicate chunks of the bandwidth, you can setup limits, but still let them borrow from other non-full peer channels as well. One setup like this at either end will manage the traffic in both directions through the link. openbsd is a little known operating system that focuses on security above all else, and its the perfect tool for routers/firewalls/traffic shapers, etc. you can generate the pf configurations with fwbuilder from linux or windows, using a gui instead of hand editing the file, but I am not sure if the the traffic shaping features are supported there or not, never tried it. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem Transferring Direct to Voicemail
I am having an issue with 1.4 where we can't successfully transfer a
call directly to a voicemail box. We hit Transfer on the phone and
dial the mailbox number we want to send it to,
My dial plan for this is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
The voicemail system picks up and starts to play its message and at this
point. We should then hit Transfer again at this point the person
doing the transfer should drop off the call. However we just continue to
hear the voicemail message and the caller continues to sit on hold.
On the Asterisk CLI I see the following:
[Feb 9 11:52:03] WARNING[5054]: chan_sip.c:12310 handle_response:
Notify answer on an owned channel?
Can anyone tell me what this means or how to fix it?
Please help.
Thanks
_
Kevin Savoy
Business Unit Telecom Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com http://www.novo1.com/
Novo 1 is a service mark of Novo 1, Inc
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Re: [asterisk-users] genzaptool from xorcom
i get this message: * CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20 * What does it mean? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth shapping device
On 22:19, Wed 14 Feb 07, Ronald Wiplinger wrote: I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) I second Jon Pounder's advice. Get an OpenBSD device. You dont need 2 boxes, you can shape on both nics. That way one machine is enough. Here's the official FAQ about queueing in OpenBSD: http://www.openbsd.org/faq/pf/queueing.html -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please some advice on setting hook-flash timing on Linksys PAP2
Hello everybody, I have a analog Fax connected to a Linksys PAP2 Adapter but got some problems with the Hook-Flash timing. I played around with the Hook Flash Timer min and Hook Flash Timer max settings at the PAP2 regional section, but the best I can get is: If I call my fax (for example from my cellphone - just for testing), the fax responds (= goes off hook). Asterisk starts music on hold and then stops music on hold again. This behavior causes that sometimes the 2 fax machines cannot recognise each other. Can somebody give me some advice on correct hook-flash timing for an analog device in Austria (or Germany)? Thanks, Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20
hello my friends, when i make a genzaptelconf i get this message CAS signalling on span 2 conflicts with HDLC with FCS check on channel *** Any idea Please? I m installing zaptel 1.4 i checked in http://bugs.digium.com/view.php?id=7860; that it's a bug but beacause i m a newbie in asterisk i can't undrestand what exactly mean Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MRTG with 4 graphs
Quoting Ronald Wiplinger [EMAIL PROTECTED]: How can I set-up a MRTG with 4 graphs, whereby: 1 data in 2 data out 3 ONLY voice(/video) data in 4 ONLY voice(/video) data out if you find out a simple recipe for that please send to me - I have tried a few times and could never get it quite right. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling Zaptel-1.2.13 on FC3
Hi Guys, Can anyone tell me why can't i compile the new zaptel driver 1.2.13 on a FC3 with kernel 2.6.9? Regards, Lawrence -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.441 / Virus Database: 268.17.39/685 - Release Date: 13/2/2007 10:01 PM Blank Bkgrd.gif Description: GIF image ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
On Wed, Feb 14, 2007 at 02:38:55PM +, younss azzayani wrote: i get this message: * CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20 * What does it mean? What card do you have? What is th output of: cat /proc/zaptel/* What is the generated /etc/zaptel.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember
On 2/13/07, gc [EMAIL PROTECTED] wrote:
I am developing an ACD front end using Asterisk 1.2.14. I heard that
AgentCallBackLogin will be deprecated in future version of *.
Is this true? If it is, how can I use AddQueueMember to replace
AgentCallBackLogin? I mean to login an agent in multiple queues at once. I
have multiple queues and a lot of agents defined in queues.conf and
agents.conf. Each agent may login more than one queue. It seem that
AgentCallBackLogin is much easier than AddQueueMember to manage this kind
of situation.
The setup to use AddQueueMember isn't terribly difficult.
Here's a contrived example where dialing *11[23]XX adds channel SIP[23]XX to
queue sales, and *21 with the same suffix removes them. *12/*22 is for
custserv and *13/*23 is for techsupp. There's no authentication here, but
that's not the difficult part of the exercise:
exten = _*11[23]XX,1,AddQueueMember(sales,SIP/${EXTEN:3})
exten = _*11[23]XX,n,Saydigits(${EXTEN:3})
exten = _*11[23]XX,n,Hangup()
exten = _*21[23]XX,1,RemoveQueueMember(sales,SIP/${EXTEN:3})
exten = _*21[23]XX,n,Saydigits(${EXTEN:3})
exten = _*21[23]XX,n,Hangup()
exten = _*12[23]XX,1,AddQueueMember(custserv,SIP/${EXTEN:3})
exten = _*12[23]XX,n,Saydigits(${EXTEN:3})
exten = _*12[23]XX,n,Hangup()
exten = _*22[23]XX,1,RemoveQueueMember(custserv,SIP/${EXTEN:3})
exten = _*22[23]XX,n,Saydigits(${EXTEN:3})
exten = _*22[23]XX,n,Hangup()
exten = _*13[23]XX,1,AddQueueMember(techsupp,SIP/${EXTEN:3})
exten = _*13[23]XX,n,Saydigits(${EXTEN:3})
exten = _*13[23]XX,n,Hangup()
exten = _*23[23]XX,1,RemoveQueueMember(techsupp,SIP/${EXTEN:3})
exten = _*23[23]XX,n,Saydigits(${EXTEN:3})
exten = _*23[23]XX,n,Hangup()
Then, calls to Queue(queuename) will work like AgentCallbackLogin() do.
The problem I am having is that the channel that shows up in the CDR and the
queue log is the phone that took the call, not the agent on the phone. It
seems that I will have to establish a mapping between agents and channels
and remove down the mapping at agent logoff, then use the map to determine
which actual agent was on SIP/200 when the call came in in order to produce
meaningful per-agent reports.
Any suggestions on how to make that part easier are welcome.
--
j.
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Re: [asterisk-users] Bandwidth shapping device
I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MRTG with 4 graphs
Ronald Wiplinger wrote: How can I set-up a MRTG with 4 graphs, whereby: 1 data in 2 data out 3 ONLY voice(/video) data in 4 ONLY voice(/video) data out bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ronald, What do you want to gather data from? Switch, router, asterisk? Model/Manufacturer. Are there MIBs specifying the info you want to gather. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
Thank You Cohen What card do you have? * Digium TE110P TDM400P, think the problem is with TE110P (configured as span 2) because i remark that the dchannel=20 * What is th output of: cat /proc/zaptel/* * Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 FXSKS 3 WCTDM/0/2 FXSKS 4 WCTDM/0/3 FXSKS Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS 5 WCT1/0/1 Clear 6 WCT1/0/2 Clear 7 WCT1/0/3 Clear 8 WCT1/0/4 Clear 9 WCT1/0/5 Clear 10 WCT1/0/6 Clear 11 WCT1/0/7 Clear 12 WCT1/0/8 Clear 13 WCT1/0/9 Clear 14 WCT1/0/10 Clear 15 WCT1/0/11 Clear 16 WCT1/0/12 Clear 17 WCT1/0/13 Clear 18 WCT1/0/14 Clear 19 WCT1/0/15 Clear 13 WCT1/0/9 Clear 14 WCT1/0/10 Clear 15 WCT1/0/11 Clear 16 WCT1/0/12 Clear 17 WCT1/0/13 Clear 18 WCT1/0/14 Clear 19 WCT1/0/15 Clear 20 WCT1/0/16 21 WCT1/0/17 22 WCT1/0/18 23 WCT1/0/19 24 WCT1/0/20 25 WCT1/0/21 26 WCT1/0/22 27 WCT1/0/23 28 WCT1/0/24 29 WCT1/0/25 30 WCT1/0/26 31 WCT1/0/27 32 WCT1/0/28 33 WCT1/0/29 34 WCT1/0/30 35 WCT1/0/31 What is the generated /etc/zaptel.conf ? ** # Autogenerated by ./genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 # Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS span=2,1,1,ccs,hdb3 bchan=5-19,21-35 dchan=20 # Global data loadzone= us defaultzone = us * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling Zaptel-1.2.13 on FC3
Hi Guys, I'm experiencing some problem with the subject above. I'm running FC3 kernel 2.6.9. I Can't seems to compile the latest zaptel driver successfully. Below are the errors I'm facing: # make linux26 make: *** No rule to make target `linux26'. Stop. # make make -C /lib/modules/2.6.9-5.ELsmp/build SUBDIRS=/root/zaptel-1.2.13 HOTPLUG_FIRMWARE=yes modules make[1]: Entering directory `/usr/src/kernels/2.6.9-5.EL-smp-i686' CC [M] /root/zaptel-1.2.13/xpp/card_fxo.o In file included from /root/zaptel-1.2.13/xpp/card_fxo.c:27: /root/zaptel-1.2.13/xpp/xpd.h:111: error: syntax error before gfp_t /root/zaptel-1.2.13/xpp/xpd.h:111: warning: function declaration isn't a prototype In file included from /root/zaptel-1.2.13/xpp/card_fxo.c:32: /root/zaptel-1.2.13/xpp/xbus-core.h:46: error: syntax error before gfp_t /root/zaptel-1.2.13/xpp/xbus-core.h:46: warning: function declaration isn't a prototype make[3]: *** [/root/zaptel-1.2.13/xpp/card_fxo.o] Error 1 make[2]: *** [/root/zaptel-1.2.13/xpp] Error 2 make[1]: *** [_module_/root/zaptel-1.2.13] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-5.EL-smp-i686' make: *** [all] Error 2 Regards, Lawrence -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.441 / Virus Database: 268.17.39/685 - Release Date: 13/2/2007 10:01 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: asterisk 1.4 FC5 and Gtalk
On 2/11/07, Stefan van der Eijk [EMAIL PROTECTED] wrote: Applied the patch, and when I call the gmail account registered on my asterisk server. Asterisk didn't crash (like it used to do before). I've had to restart asterisk a number of times over the last few days due to it eating up all the CPU. Removing the gtalk / jabber functionality solves this issue. regards, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compiling Zaptel-1.2.13 on FC3
On Wed, Feb 14, 2007 at 11:21:07PM +0800, Lawrence Na Chong Guan wrote: Hi Guys, Can anyone tell me why can't i compile the new zaptel driver 1.2.13 on a FC3 with kernel 2.6.9? Regards, Lawrence http://lists.digium.com/pipermail/asterisk-dev/2007-February/026085.html There are workarounds there. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bandwidth shapping device
Why do that? Just traffic shape each user/group of IP addresses to the total bandwidth you want them to have and then set up a low latency queue for voip traffic, that way the voip bandwidth can be used for data when there are no calls but will give VoIP traffic priority over other traffic. Any old refurbished Cisco 2611 or 2621 will do the trick. Look up low latency queuing and traffic shaping on cisco.com If you are doing NAT on the router I recommend a general deployment (GD) 12.3 IP feature set IOS image. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Wednesday, February 14, 2007 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bandwidth shapping device I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] The High Performance Echo Canceller (HPEC)
I gotta take issue with your comments that a HWEC is just software running on a DSP. In the case of Octasic, it's an ASIC. How it does EC is VERY different because.it's done completely in hardware, not firmware loaded into memory and run on a specialized CPU! Yes, the ASIC does contain an DSP but it is customized for EC. You cannot think of it as a CPU. -Original Message- From: Nic Bellamy [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 13, 2007 5:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) shadowym wrote: Interesting, Is this just a more advanced software echo canceller or software with hardware hooks or software with hardware assisted processing? A more advanced software canceller (there's no magical thing that makes hardware echo cancellers better, it's still software, but it's running on a DSP so it has more grunt available to it). It's licensed from Adaptive Digital Technologies - G.168 compliant, and supports up to 1024 taps (128ms) of tail coverage. Comes as a binary blob, but such is life. How would it compare to a true hardware echo canceller like the one Sangoma uses. Besides the extra CPU cycles required. Quite comparable - not sure if Octasic (as used by Sangoma and the latest Digium cards) or ADT would win in a shootout, but they're both in the same quality class. The main issue is going to be CPU usage - getting this going at 1024 taps on a full T1/E1 span would likely require two fast CPUs with the interrupts distributed evenly between them... and even then, *shrug* Cheers, Nic. -Original Message- From: Nic Bellamy [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 13, 2007 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Feb 13, 2007, at 5:48 PM, shadowym wrote: Interesting, Is this just a more advanced software echo canceller or software with hardware hooks or software with hardware assisted processing? How would it compare to a true hardware echo canceller like the one Sangoma uses. Besides the extra CPU cycles required. We noticed that it has slightly better performance characteristics than the Octasic, particularly in double talk scenarios, at least from our internal lab testing. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels
You have the PRIs set up to recover clock from the Asterisk box, is that what you want? If so, you certainly do *not* want span=1,1,0 or 2,2,0 since that will make Asterisk think the 81C should be clock master. Are there any telco-timed PRIs somewhere? If so, set up the PRIs on the 81C to be CLOK INT and then use span=1,1,0,esf,b8zs and span=2,2,0,esf,b8zs on the Asterisk box. I'm going to assume that a big system like an existing 81C already has the master clock set, but of course that will be a necessity if using internal clocking. - Brad They type of card we are using on the Nortel 81C will not allow clocking. The clock must be supplied by the Asterisk. We do not have any other clocking running into the Asterisk. Marlon Blair DOH, Network System Analyst (850) 245-4400, Cell (850) 528-4244 Fax (850) 412-1148 Work Hours 7 AM to 3:30 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recomended POE Phones
new ci$co phones are compliant with 802.3af, but are incompatible with asterisk ;-) .cnf.xml config files are undocumented, remote phone management (eg. restart) is very difficult, if you are not use callmanager personaly can't recommend new ci$co phones, nor obsolete models, like 7912/40/60... Tijl Van den Broeck wrote: Whatever you take: Stay away from cisco poe phones unless you're using cisco poe switches.. and even then. Cisco doesn't always apply the POE standard, older models are totally not conform the POE standard (they switched the + and - poles at the socket). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember
So you have to hard code each queue name in the dialplan for an agent to login. What about hundreds of agents login 30-40 different queues? If this is the only way to do it, I will not use AddQueueMember at all. I do not know the reason for deprecating AgentCallBackLogin. But I do think remove it without appropriate replacement is bad idea. Gary___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to launch Sendmail warning
Hi, From a bristuffed 1.0.8 Asterisk on a Gentoo system, I've got this : WARNING[8094]: Unable to launch '/usr/sbin/sendmail -t' Where could it come from ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember
- Original Message -
From: James FitzGibbon
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, February 14, 2007 10:34 AM
Subject: Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember
On 2/13/07, gc [EMAIL PROTECTED] wrote:
I am developing an ACD front end using Asterisk 1.2.14. I heard that
AgentCallBackLogin will be deprecated in future version of *.
Is this true? If it is, how can I use AddQueueMember to replace
AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have
multiple queues and a lot of agents defined in queues.conf and agents.conf.
Each agent may login more than one queue. It seem that AgentCallBackLogin is
much easier than AddQueueMember to manage this kind of situation.
The setup to use AddQueueMember isn't terribly difficult.
Here's a contrived example where dialing *11[23]XX adds channel SIP[23]XX to
queue sales, and *21 with the same suffix removes them. *12/*22 is for
custserv and *13/*23 is for techsupp. There's no authentication here, but
that's not the difficult part of the exercise:
exten = _*11[23]XX,1,AddQueueMember(sales,SIP/${EXTEN:3})
exten = _*11[23]XX,n,Saydigits(${EXTEN:3})
exten = _*11[23]XX,n,Hangup()
exten = _*21[23]XX,1,RemoveQueueMember(sales,SIP/${EXTEN:3})
exten = _*21[23]XX,n,Saydigits(${EXTEN:3})
exten = _*21[23]XX,n,Hangup()
exten = _*12[23]XX,1,AddQueueMember(custserv,SIP/${EXTEN:3})
exten = _*12[23]XX,n,Saydigits(${EXTEN:3})
exten = _*12[23]XX,n,Hangup()
exten = _*22[23]XX,1,RemoveQueueMember(custserv,SIP/${EXTEN:3})
exten = _*22[23]XX,n,Saydigits(${EXTEN:3})
exten = _*22[23]XX,n,Hangup()
exten = _*13[23]XX,1,AddQueueMember(techsupp,SIP/${EXTEN:3})
exten = _*13[23]XX,n,Saydigits(${EXTEN:3})
exten = _*13[23]XX,n,Hangup()
exten = _*23[23]XX,1,RemoveQueueMember(techsupp,SIP/${EXTEN:3})
exten = _*23[23]XX,n,Saydigits(${EXTEN:3})
exten = _*23[23]XX,n,Hangup()
Then, calls to Queue(queuename) will work like AgentCallbackLogin() do.
The problem I am having is that the channel that shows up in the CDR and the
queue log is the phone that took the call, not the agent on the phone. It
seems that I will have to establish a mapping between agents and channels and
remove down the mapping at agent logoff, then use the map to determine which
actual agent was on SIP/200 when the call came in in order to produce
meaningful per-agent reports.
Any suggestions on how to make that part easier are welcome.
--
j.
--
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Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember
So you have to hard code the each queue name in the dialplan for an agent to login. What about hundreds of agents login 30-40 different queues? If this is the only way to do it, I will not use AddQueueMember at all. I do not know the reason for deprecating AgentCallBackLogin. But I do think remove it without appropriate replacement is bad idea. Gary___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime via ODBC breaks for Voicemail
Hi all, We have an asterisk installation here that uses realtime for voicemails through ODBC. It works very well except that every now and then (ie four or five days or so) it breaks. I have included a log from the CLI of the most recent break, it looks like this: Start of output -- Executing Dial(SIP/sip.ict.ru.ac.za-b7721690, SIP/[EMAIL PROTECTED]Zap/10|20|rtT) in new stack -- Called [EMAIL PROTECTED] -- Requested transfer capability: 0x00 - SPEECH -- Called 10 -- SIP/myserver-0a145e90 is ringing -- Zap/10-1 is proceeding passing it to SIP/myserver-b7721690 -- Channel 0/1, span 4 got hangup request -- Channel 0/1, span 4 received AOC-E charging 0 units -- Hungup 'Zap/10-1' -- Nobody picked up in 2 ms -- Executing VoiceMail(SIP/myserver-b7721690, u7506) in new stack Feb 14 16:00:26 WARNING[7565]: res_odbc.c:171 odbc_smart_execute: SQL Execute returned an error -1: HYT00: [MySQL][ODBC 3.51 Driver][mysqld-5.0.27]MySQL server has gone away (66) Feb 14 16:00:26 WARNING[7565]: res_odbc.c:171 odbc_smart_execute: SQL Execute returned an error -1: 0: [MySQL][ODBC 3.51 Driver][mysqld-5.0.27]MySQL server has gone away (66) Feb 14 16:00:26 WARNING[7565]: res_config_odbc.c:124 realtime_odbc: SQL Execute error! [SELECT * FROM users WHERE mailbox = ?] Feb 14 16:00:26 WARNING[7565]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '7506' -- Executing VoiceMail(SIP/myserver-b7721690, b7506) in new stack Feb 14 16:00:26 WARNING[7565]: res_odbc.c:171 odbc_smart_execute: SQL Execute returned an error -1: HYT00: [MySQL][ODBC 3.51 Driver][mysqld-5.0.27]MySQL server has gone away (66) Feb 14 16:00:26 WARNING[7565]: res_odbc.c:171 odbc_smart_execute: SQL Execute returned an error -1: 0: [MySQL][ODBC 3.51 Driver][mysqld-5.0.27]MySQL server has gone away (66) Feb 14 16:00:26 WARNING[7565]: res_config_odbc.c:124 realtime_odbc: SQL Execute error! [SELECT * FROM users WHERE mailbox = ?] Feb 14 16:00:26 WARNING[7565]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '7506' -- Executing Hangup(SIP/myserver-b7721690, ) in new stack End of output After I saw this in the CLI, I tried to do an ODBC show to see the status of the connection, and asterisk broke: - Start of odbc show frog*CLI odbc show frog*CLI Disconnected from Asterisk server Executing last minute cleanups -- End of odbc show Any ideas why asterisk is so volatile or why the ODBC stuff breaks? Thanks, Mos ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router for supply a connection from PABX to Asterisk
some howto configuration for asterisk controlling ci$co router (pri/qsig ports especially) using mgcp interests me too... ;-) Yehavi Bourvine +972-8-9489444 wrote I am using a Cisco to connect Asterisk via PRI to our Nortel TX-1. The Cisco is a voice bundle of 2,811 + E1 + PDLM card. Note that you need PDLMs as the same number as the PRI channels you are going to define (i.e. 32 PDLMs for each PRI). I am controlling the Cisco via SIP; it works, but a few problems: - Only basic connectivity. No additional features (like names) as the Cisco supports them only via MGCP (in MGCP is passes all the Q.sig signals to the PBX - Asterisk in this case - and it should do all the handling, but I did not find how to do it with Asterisk). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fanless solution
Hi there, I'm looking for a compact fanless solution preferrably wall mountable and not too exotic. It needs to be commercial grade. I don't really consider most of the Via ITX solutions I have seen commercial grade but perhaps someone can convince me otherwise. This solution is about the best I have found. Maybe a bit on the exotic side but I like the fact it is wall mountable AND has 2 PCI slots which I have been having trouble finding. Anyone have any experience with this company and their products? http://www.nexcom.com/product/productshow.jsp?iid=11pid=377 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
Peter Bowyer wrote: The same ones which, by total coincidence, you just advertised on asterisk-biz, perhaps? What are the chances of that? Thanks for the heads-up. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Guide to better performance using * ?
Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to launch Sendmail warning
Olivier wrote: Hi, From a bristuffed 1.0.8 Asterisk on a Gentoo system, I've got this : WARNING[8094]: Unable to launch '/usr/sbin/sendmail -t' Where could it come from ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users is sendmail binary there and it's executable? ls -l /usr/sbin/sendmail Check too that user that run asterisk has permission to it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to launch Sendmail warning
On Wed, Feb 14, 2007 at 05:42:13PM +0100, Olivier wrote: Hi, From a bristuffed 1.0.8 Asterisk on a Gentoo system, I've got this : WARNING[8094]: Unable to launch '/usr/sbin/sendmail -t' Where could it come from ? ls -l /usr/sbin/sendmail -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
Steve Kennedy wrote: On Wed, Feb 14, 2007 at 03:42:23AM +0100, Patrick wrote: On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote: Singer Wang wrote: by your .ca address I assume your in Canada.. both Telus and Rogers have a email-to-SMS gateway... Well, those are notoriously unreliable. I've had messages take hours to arrive when sent by the email-to-SMS gateway. I was kinda hoping for something more direct. Rogers prioritizes internal SMS messages over e-mailed ones. What I'd like is some kind of SMSC -- or something that accomplishes the same thing. Maybe http://www.kannel.org/ provides some useful info. Kannel is a pretty mature solution, it will drive a local GSM terminal or connect through to SMSC's using standard protocols (SMPP, CIMD, UCP/EMI etc) or even http/SOAP. Terminals such as Siemens TC/MC35, Wavecomm, Falcom etc seem to work well and Kannel tends to have driver modules for them, also many phones can also work. Make sure SIM buffering isn't used or you'll wear out the SIM (they have limited writes). If I understand correctly, this means I'll need an extra SIM just to send messages -- is that right? I build a Kannel server so that it can talk to a terminal that is on the network and can send messages. (It's an awful lot of extra hardware just for messaging capacity that will only be used by a few users, though.) What if I don't want to get my own terminal? Most operators wont allow direct connectivity unless you delivering 10's of millions of SMSs per month and you'll have to go through an aggregator. Can you show me an example of an aggregator? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
Peter Bowyer wrote: On 14/02/07, Stephen Bosch [EMAIL PROTECTED] wrote: Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere (they are not interested in helping an individual customer and technical staff Tiers I and II have no idea what I am talking about). Are there SMS aggregators that I could use for sending messages to this particular phone over the Internet? There are plenty - I've used 2sms.com, clickatell.com and csoft.co.uk. bayhamsystems.com have a service tailored for Asterisk users. These are all based in the UK. What if I'm in North America? Does it matter? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
Tim Panton wrote: We've used www.Simplewire.com , they have a x86 linux executable which we wrap in a shell script and call from the dialplan with a System() call. We've been happy with them for years. Wow! Are these guys in Canada? (One of the sample numbers was a 416 area code, which is in Toronto). I tried a sample message -- it arrived in 2 seconds. That is better performance that Rogers' own web interface! More information, please! -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zoiper softphone version 1.03 now available
Hello guys! We released a new ZoIPer BIZ BETA (version 1.03). You can experience better look and more advanced features. Finally MS Vista fans can also make use of it. Zoiper BIZ BETA is available free of charge from www.zoiper.com. There you can find out more about the improvements and features. We are also offering customization packages for ZoIPeR Free Windows. Zoiper is a multiprotocol: SIP and IAX / IAX2 softphone, supporting native conferencing, g729(optionally), Call recording, callto URL protocol, autoanswer and much more. For more information please consult: www.zoiper.com www.attractel.com Greetings, Mira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
no idea? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Feb 14, 2007, at 10:17 AM, shadowym wrote: I gotta take issue with your comments that a HWEC is just software running on a DSP. In the case of Octasic, it's an ASIC. How it does EC is VERY different because.it's done completely in hardware, not firmware loaded into memory and run on a specialized CPU! Yes, the ASIC does contain an DSP but it is customized for EC. You cannot think of it as a CPU. Yes, but the math and functions involved are the same. It's just doing it on one or the other involves different types of instructions. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect a legacy PBX to an Asterisk Server
I am planing to connect a legacy PBX (Avaya Ip Office 406) to an Asterisk Server. I want to use the * as VoIP Gateway. The Avaya PBX has 3 CO ports available, so I thought buying a TDM30B with 3 FXS ports and connect then to the Avaya CO ports. Is this possible? Would this be the right way to do it? Any recommendation? Thanks in advance Housi Mueller - Need Mail bonding? Go to the Yahoo! Mail QA for great tips from Yahoo! Answers users.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4 svn doesn't compile
Is there a zaptel mailing list? Here's the error: CC [M] zaptel-1.4/xpp/xbus-core.o zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’: zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘u’___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libunicall + hashtable.c + asterisk crash
I´m having some problems with mfcr2. Asterisk crash with exit code 137 signal 11. Here is what i can get from core dumped: Loaded symbols for /lib/libgcc_s.so.1 #0 OneWordFind (tablePtr=0x32333134, key=0x80e6 Address 0x80e6 out of bounds) at hashtable.c:586 586 if (hPtr-key.oneWordValue == key) hashtable.c is a libunicall file. Asterisk goes down many times a day and the problem gets worst when many Dial commands are requesteds. Any help will be great!! Andre Dias ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Limit on SIP phones on one server
I have an application where I might need 700 SIP phones (wireless) connected to one asterisk server. Will it do this? The situation: Only a small number (less than 10) will actually be talking at one time. I presume asterisk can handle 700 SIP definitions correct? Do I need to recompile anything to handle that many phones? Thanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Call Start
Michael Collins wrote: “At times I think the wiki has grown out of control.” I hear you. I’d pay money to anyone willing to create and maintain a master index! And use a different Wiki engine! Augh! (Mediawiki, anyone?) Who runs voip-info.org? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Guide to better performance using * ?
Tim Connolly wrote: Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. Ah! A Dell! What does your 'cat /proc/interrupts' say? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Can someone comment why only Digium cards still under warranty are eligible to use this EC at no cost, versus older cards? Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
On 14/02/07, Stephen Bosch [EMAIL PROTECTED] wrote: Peter Bowyer wrote: There are plenty - I've used 2sms.com, clickatell.com and csoft.co.uk. bayhamsystems.com have a service tailored for Asterisk users. These are all based in the UK. What if I'm in North America? Does it matter? What matters is whether they can deliver to your target users - check what countries + networks each one quotes in their footprint. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
On Wed, Feb 14, 2007 at 10:29:20AM -0700, Stephen Bosch wrote: [snippage] If I understand correctly, this means I'll need an extra SIM just to send messages -- is that right? I build a Kannel server so that it can talk to a terminal that is on the network and can send messages. (It's an awful lot of extra hardware just for messaging capacity that will only be used by a few users, though.) What if I don't want to get my own terminal? Then you need to talk to someone who offers connectivity into the operators. Most operators wont allow direct connectivity unless you delivering 10's of millions of SMSs per month and you'll have to go through an aggregator. Can you show me an example of an aggregator? I don't know in the US? There are some ... they'll have an API and you then utilise that API to inject messages. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with safe_asterisk
Gordon Henderson wrote: On Tue, 13 Feb 2007, Tzafrir Cohen wrote: On Tue, Feb 13, 2007 at 01:35:50PM +0100, Andrea De Vita wrote: Hi all, I have installed some Asterisk machine, all with the same problem. My typical configuration is: - Asterisk 1.2.14 (or 1.4.0beta3) - CentOS 4.4 server. The problem is this: When I start Asterisk with the default init script (/etc/init.d/asterisk start) distributed with the source, and kill (or kill -9) Asterisk-pid, then safe_asterisk doesn't correctly work (it dies and not restart Asterisk). Instead, if I start Asterisk with safe_asterisk command from shell, after kill Asterisk-pid, safe_asterisk restart Asterisk correctly. I would use the init script because I like to use Linux-HA that require this. Edit that init.d script lightly not to use safe_asterisk. safe_asterisk is not close to robust anyway, and thus will only complicate things. Seconded. Have a look at this: http://www.drogon.net/init.d.asterisk Gordon, What is the point of reloading extensions after starting asterisk in the start section of your case statement? Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro Usage
Hello, I have the following simple application... 1. Call is answered, and Dial() function is used with a macro to dial out to a number. 2. 'Called' party answers the phone, and hears a message (this is a function of the macro) At this point I'd like for the 'Called' Party to be able to make a decision and press 1 or 2 to hear some additional information before accepting the call. The problem is that any key pressed causes the call to be bridged. Is this the only behavior, or can someone help me with an example of a script that will allow the 'Called' party to do some things before the call is bridged. I have included my macro code below ** [macro-acceptcall] exten = s,1,Answer() exten = s,2,Wait(1) exten = s,3,Background(makechoice) ;make a choice, press 1 or 2 exten = s,4,WaitExten(3) exten = s,5,Goto(s,3) exten = 1,1,Background(youchose1) exten = 2,1,Background(youchose2) *** This is what I want to happen, but it just bridges the call immediately without playing the respective messages. Thanks for the help. Thanks! J ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 SIP or IAX providers?
On 2/13/07, Dan Burwinkel [EMAIL PROTECTED] wrote: Hi Kyle, Vitelity.net does it for me... There are a few others too. I tried a half dozen, but none seem to have the elusive Customer Service, E911, and good Voice quality. I use multiple providers. Les.net is great for everything but E911. Origination-- Les.net . Termination-- Les.net, voipjet, and vitelity.net . E911-- Vitelity.net I've been using VoIP exclusively at my home for about a year. I started with an ATA and moved to TrixBox and Polycom IP501s. I tried Linksys SPA942, Snom, GXP-2000, and Aastra. The only ones I really could get a totally natural sound out of was Aastra and Polycom. I'm finally happy with the sound. I really had a hard time finding a provider that supported smaller fish like me. Dan Kyle Sexton wrote: Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dan, Do you know if vitelity has a echo test or any lines I can call for a quality test? Do you have their servers address so I can check my connection to there? If they've got a good connection and the E911 works I may have to sign up! -- Kyle Sexton ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
On Wed, Feb 14, 2007 at 03:44:25PM +, younss azzayani wrote: Thank You Cohen What card do you have? * Digium TE110P TDM400P, think the problem is with TE110P (configured as span 2) because i remark that the dchannel=20 * What is th output of: cat /proc/zaptel/* * Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 FXSKS 3 WCTDM/0/2 FXSKS 4 WCTDM/0/3 FXSKS Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS 5 WCT1/0/1 Clear 6 WCT1/0/2 Clear 7 WCT1/0/3 Clear 8 WCT1/0/4 Clear 9 WCT1/0/5 Clear 10 WCT1/0/6 Clear 11 WCT1/0/7 Clear 12 WCT1/0/8 Clear 13 WCT1/0/9 Clear 14 WCT1/0/10 Clear 15 WCT1/0/11 Clear 16 WCT1/0/12 Clear 17 WCT1/0/13 Clear 18 WCT1/0/14 Clear 19 WCT1/0/15 Clear 13 WCT1/0/9 Clear 14 WCT1/0/10 Clear 15 WCT1/0/11 Clear 16 WCT1/0/12 Clear 17 WCT1/0/13 Clear 18 WCT1/0/14 Clear 19 WCT1/0/15 Clear 20 WCT1/0/16 This is the D channel, right? Is the connection a E1 PRI? 21 WCT1/0/17 22 WCT1/0/18 23 WCT1/0/19 24 WCT1/0/20 25 WCT1/0/21 26 WCT1/0/22 27 WCT1/0/23 28 WCT1/0/24 29 WCT1/0/25 30 WCT1/0/26 31 WCT1/0/27 32 WCT1/0/28 33 WCT1/0/29 34 WCT1/0/30 35 WCT1/0/31 31 channels, as expected. What is the generated /etc/zaptel.conf ? ** # Autogenerated by ./genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 # Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS span=2,1,1,ccs,hdb3 bchan=5-19,21-35 dchan=20 # Global data loadzone= us defaultzone = us The error you get is from a place in ztcfg's code that applies some sanity checks to the signalling it sends to channel no. 16 of a span. If they are not met, that channel cannot be considered a D channel. I didn't understand those conditions exactly. In one specific case were I helped someone on #asterisk that guy eventually removed the sanity check from ztcfg and moved on. Whether or not this is a wise thing to do, I don't know. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk vendors in Houston, TX
Does anyone know of a good Asterisk/LAN/PC support company in Houston, TX? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] S101I (IAX) limitation
That little Digium adapter S101I is a nice and compact (good for travel) but has a serious limitation, it doesn't support bridging. If I have four/five of these units registered to my asterisk server (over the Internet) and with standard DSL or Cable connection all for of them connected and in use (utilizing G711); my upload bandwidth will get saturated as all of them have to go via my Asterisk server (according do Digium support). They don't support bridging mode. I think this limitation seriously cripple these devices. Why is so hard to implement bridging? Any links explaining how is it done? How FWD solved this problem? -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4 svn doesn't compile
On Wed, Feb 14, 2007 at 01:31:44PM -0500, Robert La Ferla wrote: Is there a zaptel mailing list? Not really. Here's the error: CC [M] zaptel-1.4/xpp/xbus-core.o zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’: zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘u’___ Look a bit above line 171 in xpp/xbus-core.c . There's a condition there regarding that struct indoe. Replace 2.6.18 with 2.6.19 (because you don't really have a Fedora) ;-) Already fixed in branch 1.2 . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Wednesday 14 February 2007 11:17 am, shadowym wrote: I gotta take issue with your comments that a HWEC is just software running on a DSP. In the case of Octasic, it's an ASIC. How it does EC is VERY different because.it's done completely in hardware, not firmware loaded into memory and run on a specialized CPU! Yes, the ASIC does contain an DSP but it is customized for EC. You cannot think of it as a CPU. Why not? A DSP is a CPU which has been designed to do mathematical functions very quickly, generally especially with respect to matrix math. I mean think of what you just said. You could just as easily have said A CPU ... it's an ASIC. Everything it does is completely in hardware. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Wednesday 14 February 2007 11:19 am, Matthew Fredrickson wrote: We noticed that it has slightly better performance characteristics than the Octasic, particularly in double talk scenarios, at least from our internal lab testing. How has the testing been with respect to its use on FXO ports (such as those on the TDM400 FXO modules) ?? I'm *very* interested in any real test data, including any comparisons with MG2 and the Octasic cancellers available on Digium products. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Guide to better performance using * ?
We had a similar issue with a 'new' server here. We had a trixbox install and the kernel didn't support the particular type of motherboard/drive combination and the disk was not in DMA mode. There was nothing we could do to get it to work and eventually put in an older motherboard. Since then, its been working beautifully. Run a hdparm /dev/hda (or whatever your disk is) and make sure its in dma mode. -- Kevin Withnall ILB Computing PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4253 0001 http://www.ilb.com.au/ http://kevin.withnall.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Thursday, 15 February 2007 5:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Guide to better performance using * ? Tim Connolly wrote: Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. Ah! A Dell! What does your 'cat /proc/interrupts' say? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] The High Performance Echo Canceller (HPEC)
The algorithms may be similar but EC is an infinitely variable non-linear(analog) process. A CPU cannot do that. You can fake it by performing cpu intensive rapid calculations one after another but it is fundamentally not an analog processor. HWEC is designed to deal with the analog process on an instant by instant basis performing parallel computations. A CPU cannot do that at ANY clock speed. -Original Message- From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 14, 2007 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) On Feb 14, 2007, at 10:17 AM, shadowym wrote: I gotta take issue with your comments that a HWEC is just software running on a DSP. In the case of Octasic, it's an ASIC. How it does EC is VERY different because.it's done completely in hardware, not firmware loaded into memory and run on a specialized CPU! Yes, the ASIC does contain an DSP but it is customized for EC. You cannot think of it as a CPU. Yes, but the math and functions involved are the same. It's just doing it on one or the other involves different types of instructions. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with safe_asterisk
On Wed, Feb 14, 2007 at 08:39:36AM +, Gordon Henderson wrote: On Tue, 13 Feb 2007, Tzafrir Cohen wrote: On Tue, Feb 13, 2007 at 01:35:50PM +0100, Andrea De Vita wrote: Hi all, I have installed some Asterisk machine, all with the same problem. My typical configuration is: - Asterisk 1.2.14 (or 1.4.0beta3) - CentOS 4.4 server. The problem is this: When I start Asterisk with the default init script (/etc/init.d/asterisk start) distributed with the source, and kill (or kill -9) Asterisk-pid, then safe_asterisk doesn't correctly work (it dies and not restart Asterisk). Instead, if I start Asterisk with safe_asterisk command from shell, after kill Asterisk-pid, safe_asterisk restart Asterisk correctly. I would use the init script because I like to use Linux-HA that require this. Edit that init.d script lightly not to use safe_asterisk. safe_asterisk is not close to robust anyway, and thus will only complicate things. Seconded. Have a look at this: http://www.drogon.net/init.d.asterisk Here is one bad case many people seem to miss: Run 'asterisk -c' Now press ctrl-z (SIGSTOP) to freeze it. Now try running your script's stop target. It will hang forever on the write to the asterisk.ctl socket. A hung stop target can freeze a shutdown of a server. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
