Re: [asterisk-users] Sysmaster and Asterisk
On 9/6/07, Mani Nair [EMAIL PROTECTED] wrote: Hello Guys, I am unable to make calls to outside number from some of my extensions. Internally I am able to make and receive calls between extensions and also I am able to receive call from outside number. Any suggestions? Then in am thinking of getting rid of Sysmaster and configure Trixbox to do the entire job that currently my Sysmaster is doing. Any suggestions..? Suggestion is check the dialplan check asterisk cli check network trace with ngrep you have sysmaster and want to move to Trixbox ? ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Debugging to separate log file
Dear Jared; I would like to ask if there is a method to let the output of set sip debug ip to be sent for a file? Regards Bilal Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? I always find it easier to extract the SIP messaging traffic by using tcpdump or ngrep. If you use tcpdump, you can always pass the traffic through ngrep later, as well as passing it through Wireshark to get the pretty SIP traffic graphs, etc. -- Jared Smith Community Relations Manager Digium, Inc. Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Debugging to separate log file
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Jared; I would like to ask if there is a method to let the output of set sip debug ip to be sent for a file? hi when iam doing this i see the server is load is very high how can i send this traffic or mirror traffic to other server and grep the reports ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special kind of billing
Dear Guillermo; Is there an english link that help me in configuration other than: http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos Also, what about ASTCC? Another issue: a2billing support prepaid billing (so it can be used for calling cards)? Regards, ITS - Telecom Group IP Telephony And Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460 Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Yes, We are using a2billing [1]. You can define serveral trunks and add rates for the destinations, the a2billing can use low cost routing and gives to you a detailed call detail record with the ammount of sell, buy, profit, margin and markup. You can learn to use with this small guide (spanish): http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos [1] www.asterisk2billing.org Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Udev issue on zaptel install
On Sat, Sep 08, 2007 at 02:58:40PM -0400, Hariharan Veerappan wrote: On 9/6/07, Tzafrir Cohen [EMAIL PROTECTED], rcom.com wrote: udev is not a prerequirement for zaptel. Debian Sarge uses devfs by default, and Zaptel supports devfs as well. since the udev not installed in by the sequence, that may not supported in your distribution, use the correct version of udev for linux kernel version. i got the same problem with another device, udev wont create the device node automatically, if yours seems to be the same, this approach may solve the problem Again: 1. Devfs does create the device files automatically. Thus udev is not required on Debian Sarge. 2. On Debian Sarge with kernel 2.6 installing the package udev will give you a working udev version. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu Feisty
On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote: Hi all, Have just installed v1.4.11 of Asterisk, but I am trying to have it start at boot but with no luck. I have used the make config command but it doesn't start. Any help would be apreciated, many thanks! use the command update-rc.d Also, as always in the case of software that has already been packaged, it may help to look at the existing package. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu Feisty
Hi, On 2007-09-09 at 13:30 Tzafrir Cohen wrote: use the command update-rc.d Also, as always in the case of software that has already been packaged, it may help to look at the existing package. What parameter should I use with that command? Many thanks, Christian -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu Feisty
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote: Hi all, Have just installed v1.4.11 of Asterisk, but I am trying to have it start at boot but with no luck. I have used the make config command but it doesn't start. Any help would be apreciated, many thanks! use the command update-rc.d Also, as always in the case of software that has already been packaged, it may help to look at the existing package. I used update-rc.d asterisk 30 to ensure that it started after zaptel and mysql (which by default start at 20). - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.136111 Linux Counter No. 202120 Ekiga: 645022 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRuPqzktP/KMNOfRbAQL99ggAjRNda5TZxEatmd9aSrI6COtqNGhTgyXb p3+BTtHxMsVxB7B/L/GSbxi5swBxTRgO1egiILYfEkA3rjtDo9fMPUSW/YLgyGdf vpKpiKCPziuw9OeHyumFbIi7ORlNwWwAANVcSngCRELwv8wczj7upHRPJOeGFGRU vEfWDgilUpuhep6DhCtFjO90Qs9icbGgRn7gAegYBNVb6QCExywRljhsKSxEkF5q VOqdLKLL2I4Im1YZChG4BXt1U0mUdIJwp/Q+g+s/ahBZ7ivFFobOU7PVU7QW8puN Mp5MmZz0A1DN80qBlhNBT0Elu11MWL2qr9I6R41jJftyWDK+rcA1PA== =bwb3 -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [mythtv-users] Real Time Clock Alarm Broken with 2.6.22+ kernel
Ok, the script is attached... I'll post it on www.generationd.com when I have a chance. If you have any updates improvements please email them to me! (The command line parameter handle is pretty stupid - just grew from testing to production without cleanup). MD _ From: Craig Huff [mailto:[EMAIL PROTECTED] Sent: Saturday, September 08, 2007 11:34 AM To: Michelle Dupuis Subject: Re: [mythtv-users] Real Time Clock Alarm Broken with 2.6.22+ kernel On 9/7/07, Michelle Dupuis [EMAIL PROTECTED] wrote: Craig, I wrote an nvram-wakeup replacement call acpi-wakeup. It is a bash script that is command line compatible with nvram-wakeup, but uses the proc acpi interface and handles epoch data etc. The acpi path is just a constant you can change. It would need a tweak to work for you situation, but your welcome to play with it. I can put it on www.generationd.com if you want it. MD Michelle, If you would, please. You could e-mail it to the list as an attachment instead, if you like. Then it would be in the archives for others to reference. Craig. acpi-wakeup Description: Binary data ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what is the usable feature in DUNDi?
hi: i create a dundi environment by the caveman can do it dundi guide. it works fine.but i want to extend the example for my own need, so i follow the sample dundi.conf config file comes with asterisk 1.4.11 source. i try to use precache and failed, and there seems no one know how to use it after googling. i try to setup dynamic dundi with [*] and failed, and google tell me that feature is not implement yet. there is a patch to fix this: http://bugs.digium.com/view.php?id=10546nbn=1 i can only find one kind of syntax example about the mappings keyword. there seems no other way about how to using mappings. so i wonder what's the situation of DUNDi today. what's the usable feature in asterisk 1.4? will it become mature in the 1.4 release? or we should wait for 1.6? thanks for help!! Regards, tbskyd ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special kind of billing
On Sun, 2007-09-09 at 02:44 -0700, bilal ghayyad wrote: Dear Guillermo; Is there an english link that help me in configuration other than: http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos Check the www.asterisk2billing.org documentation page. Also, what about ASTCC? I've not used it yet. Another issue: a2billing support prepaid billing (so it can be used for calling cards)? Yes. Check the features list at http://trac.asterisk2billing.org/cgi-bin/trac.cgi . Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue static agents
--- Mark Michelson [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi,
I setup a queue (number 4050) with one static
agent
(extension 4054).
What I would like is that when someone calls the
4050
queue and there are neither dynamic agents
logged in
nor is the static agent 4054 on-line then the
caller
gets out of the queue and falls into another
context
(eg. voicemail or anything). Not on-line means
that
either the SIP extension 4054 has not registered
with
Asterisk or has activated DND.
There is another option for joinempty that might
help you out. The
strict option will not allow a caller to join the
queue if agents are
either not logged in, or are logged in but
unavailable. The no option
will only not allow a caller to enter if there are
no agents logged in.
Thanks Mark.
Unfortunately, strict alone doesn't work with static
agents unless I add the extensions to the queue
programmatically (with the use of
{Add,Remove}QueueMember).
I could call the AddQueueMember function **each time**
someone calls the queue number but wouldn't that be
excess work on behalf of Asterisk?
Suggestions appreciated.
Vieri
Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel
and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7
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[asterisk-users] Softkeys wrong with chan_skinny
Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Regards, Andreas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softkeys wrong with chan_skinny
On Sun, 2007-09-09 at 15:45 +, Andreas Anderson wrote: Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Hi, asterisk-trunk also allows the redial to work. The other buttons are not supported 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? signature.asc Description: This is a digitally signed message part ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, September 05, 2007 2:38 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: I have a client setup where a T1 is terminated into a Cisco IAD2430 Series device which then interfaces with a Digium Wildcard TE110P card in a server running Asterisk 1.2.23. I am having a problem with the DTMF tones being passed to the Asterisk server. Wrong tones are being passed to the server especially during the digital receptionist menu selections. Setting relaxdtmf=yes does not seem to address the situation. Any pointers? Try the patch at http://bugs.digium.com/view.php?id=10535 and see if it helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote: I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. A make install will not overwrite any configfile. It will install the asterisk binary and the modules (thus overwriting the existing files) but configfiles will only be overwritten when you run: make samples -- 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? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Build your own appliance concept
Jeremy P wrote: Thanks for all the good info. If you're looking for a cheaper version of the thin client you could try the t5530. It's about $300 US but it only has 64 MB Flash. A 1GB flash module is $70 US but sounds like overkill for your application. Frankly, the 70 clams is the worth time saved on stripping down your install to make it fit. Flash is so cheap nowadays that it's hard to justify the effort. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
G B wrote: Hi, I appreciate the help. I called the vendor of the card and they recommended removing all of the PCI cards on the system (including the video card), and moving the card to a new PCI slot. I did all of them together, ran the system headless, and ssh'ed in remotely. It worked! haha... This must be proof that I have purchased a real piece of @#$. Glad you said it without us having to. At least it was cheap, right? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
Steve Totaro wrote: I am the last one to pickup a manual or call tech support but yesterday I was working on a very large industrial ShopBot (It is a robot so that is cool and it does really awesome things but why I was working on it don't ask.. http://www.shopbottools.com/applications.htm ) After trying a million things, briefly looking at the manual, I called the tech support line. The guy had me check two things, change one thing and everything was joyful. Had I done that from the start, I would have saved three or four hours (I bill by the hour so it's not so bad, but I couldn't bill the full rate since conscience told me not to) Show us your Asterisk configs for the ShopBot. Can I build a dresser from payphone? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meridian S1 to Asterisk via T1
David Gomillion wrote: On 9/7/07, *Michelle Dupuis* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: This is going into an emergency response facility...where they currently have a Nortel Option 61 (I think). They want to slowly phase into VoIP. They will need 1000 phone set capacity (assuming full migration). This can be done, and I am a proponent of Asterisk. But I don't think I would recommend it in this situation. Frankly, having a big company like Nortel to blame if/when downtime occurs would be worth the money difference to me! Would it? Having someone to blame doesn't mean you didn't have a massive outage, and also doesn't mean that the vendor you are blaming is actually going to fix the problem. Which is not to say that there aren't good commercial products that are appropriate in certain circumstances... just that people place an awful lot of faith in their service agreements, probably more than they should. What you need in a situation like the above is some engineering depth and people with lots of deployment experience. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which cause less CPU usage: GSM or wav??
I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram TIA -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference in show channels
'show channels' shows only running calls while 'sip show channels' shows all running sip sessions including phones trying to register . On 09/09/2007, ram [EMAIL PROTECTED] wrote: Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
On 10:28, Sun 09 Sep 07, Barton Fisher wrote: I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram depends on what codec the T1 is using. Best to transcode the ivr sounds to the same codec to prevent on-the-fly transcoding by asterisk. -- 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? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Behaviour
Hi, my ATA has two phones attached and the possibility to set different accounts. I put two account of my asterisk server, however, it is able to call only with the second one in order to the sip.conf and the first it gives me 403. And idea how to solve it? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference in show channels
On 9/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote: 'show channels' shows only running calls while 'sip show channels' shows all running sip sessions including phones trying to register . thanks but after my 30 channels of show channels i see lot of vice break and choppy voice doing passthrough codecs Xeon 2.0GHZ with 2 GG Ram centos 4.4 1.2.17 any suggestions ram On 09/09/2007, ram [EMAIL PROTECTED] wrote: Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Michiel van Baak wrote: On 10:28, Sun 09 Sep 07, Barton Fisher wrote: I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram depends on what codec the T1 is using. Best to transcode the ivr sounds to the same codec to prevent on-the-fly transcoding by asterisk. The answer is going to ulaw or alaw depending where you live. T1 should most likely be using ulaw so make everything ulaw, end to end. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF bug in dsp.c and 1.4.11
I was wondering if this bug: http://bugs.digium.com/view.php?id=10535 would affect a PRI connection. I seem to be dropping DTMF digits on the PRI. The company says they have test the line and they way the PRI is fine as far as they are concerned. So will this bug and patch help me? I am running 1.4.11 Jerry ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu Feisty
Hi, What parameter should I use to that command? On 2007-09-09 at 13:45 Ron Wellsted wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote: Hi all, Have just installed v1.4.11 of Asterisk, but I am trying to have it start at boot but with no luck. I have used the make config command but it doesn't start. Any help would be apreciated, many thanks! use the command update-rc.d Also, as always in the case of software that has already been packaged, it may help to look at the existing package. I used update-rc.d asterisk 30 to ensure that it started after zaptel and mysql (which by default start at 20). - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.136111 Linux Counter No. 202120 Ekiga: 645022 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRuPqzktP/KMNOfRbAQL99ggAjRNda5TZxEatmd9aSrI6COtqNGhTgyXb p3+BTtHxMsVxB7B/L/GSbxi5swBxTRgO1egiILYfEkA3rjtDo9fMPUSW/YLgyGdf vpKpiKCPziuw9OeHyumFbIi7ORlNwWwAANVcSngCRELwv8wczj7upHRPJOeGFGRU vEfWDgilUpuhep6DhCtFjO90Qs9icbGgRn7gAegYBNVb6QCExywRljhsKSxEkF5q VOqdLKLL2I4Im1YZChG4BXt1U0mUdIJwp/Q+g+s/ahBZ7ivFFobOU7PVU7QW8puN Mp5MmZz0A1DN80qBlhNBT0Elu11MWL2qr9I6R41jJftyWDK+rcA1PA== =bwb3 -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? Bart Steve Totaro wrote: Michiel van Baak wrote: On 10:28, Sun 09 Sep 07, Barton Fisher wrote: I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram depends on what codec the T1 is using. Best to transcode the ivr sounds to the same codec to prevent on-the-fly transcoding by asterisk. The answer is going to ulaw or alaw depending where you live. T1 should most likely be using ulaw so make everything ulaw, end to end. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2516 (20070909) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Debugging to separate log file
Dear Ram; You are able to send it for a file? Regards Bilal Dear Jared; I would like to ask if there is a method to let the output of set sip debug ip to be sent for a file? hi when iam doing this i see the server is load is very high how can i send this traffic or mirror traffic to other server and grep the reports ram Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu Feisty
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Christian wrote: Hi, What parameter should I use to that command? On 2007-09-09 at 13:45 Ron Wellsted wrote: Tzafrir Cohen wrote: On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote: Hi all, Have just installed v1.4.11 of Asterisk, but I am trying to have it start at boot but with no luck. I have used the make config command but it doesn't start. Any help would be apreciated, many thanks! use the command update-rc.d Also, as always in the case of software that has already been packaged, it may help to look at the existing package. I used update-rc.d asterisk 30 to ensure that it started after zaptel and mysql (which by default start at 20). Sorry, it should have read sudo update-rc.d asterisk defaults 30 - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.136111 Linux Counter No. 202120 Ekiga: 645022 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRuRnQktP/KMNOfRbAQLvyQf+N16Y9W3cuDdySMkZImPh7BEMfoHmmr++ ghZulTdrwQ436MoEQx0AOu/ZemxJoXarRSg6ju7ehpHyIk7+PebujH8UWdxO9HzW pnMJSDB5NexKYtRWjW8uHwxPbTX45LugYQWKSMFuQ94SmVwAO2vhL9mppnlJpJGy 5WZ3AOwpNW/lHIjojGWxXO9FuzBGWQDX6evWSOWWO4St69PSk2HnAYFGIvZ+G4Zl HyNgLkGmX7khDQpJc9olle7Rilr19kLWmKY0kERrE+M7L9k09nZyM9ZWAz4AaZsx LT/AMx1UB9iXR9G8HxHaGdUyXpa+BjaI7YREw8O7ZeJORJtsXwfA7Q== =/9iA -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special kind of billing
You can try MOR: www.kolmisoft.com/mor It does what you need. It does it even in FREE version. PRO version costs _many_ times less then other not free solutions mentioned in this thread. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kate Kretz Sent: Wednesday, September 05, 2007 7:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] special kind of billing Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Cheers, Kate ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Behaviour
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Hi, my ATA has two phones attached and the possibility to set different accounts. I put two account of my asterisk server, however, it is able to call only with the second one in order to the sip.conf and the first it gives me 403. And idea how to solve it? Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. If those are not the source of trouble, _I_ probably would switch the accounts in the ATA (port A versus port B) and try if the problem sticks with the port or with the account. I would also google if there are known problems with my ATA, look if a newer firmware is available, if there are informative messages that are worth a verbatim quote, and get another bottle of beer to keep the sunday relaxation at a proper level. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu Feisty
Hello, On 2007-09-09 at 22:36 Ron Wellsted wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Christian wrote: Hi, What parameter should I use to that command? On 2007-09-09 at 13:45 Ron Wellsted wrote: Tzafrir Cohen wrote: On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote: Hi all, Have just installed v1.4.11 of Asterisk, but I am trying to have it start at boot but with no luck. I have used the make config command but it doesn't start. Any help would be apreciated, many thanks! use the command update-rc.d Also, as always in the case of software that has already been packaged, it may help to look at the existing package. I used update-rc.d asterisk 30 to ensure that it started after zaptel and mysql (which by default start at 20). Sorry, it should have read sudo update-rc.d asterisk defaults 30 Many thanks, will try that. Is Zaptel already loaded or will I need to do another command for that? Still learning. Many thanks, Christian - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.136111 Linux Counter No. 202120 Ekiga: 645022 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRuRnQktP/KMNOfRbAQLvyQf+N16Y9W3cuDdySMkZImPh7BEMfoHmmr++ ghZulTdrwQ436MoEQx0AOu/ZemxJoXarRSg6ju7ehpHyIk7+PebujH8UWdxO9HzW pnMJSDB5NexKYtRWjW8uHwxPbTX45LugYQWKSMFuQ94SmVwAO2vhL9mppnlJpJGy 5WZ3AOwpNW/lHIjojGWxXO9FuzBGWQDX6evWSOWWO4St69PSk2HnAYFGIvZ+G4Zl HyNgLkGmX7khDQpJc9olle7Rilr19kLWmKY0kERrE+M7L9k09nZyM9ZWAz4AaZsx LT/AMx1UB9iXR9G8HxHaGdUyXpa+BjaI7YREw8O7ZeJORJtsXwfA7Q== =/9iA -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maximum retries exceeded on transmission
I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call 778f89593967725f0abe40eb1752504c no reply to our critical packet. What is the critical packet that is not being responded to? Please help. - Pinpoint customers who are looking for what you sell. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is the difference between increasing the verbose level and the debug level?
Hi List; What is the difference between increasing the verbose level and the debug level? By increasing the verbose level, then I will get more traces messages and by increasing the debug level, I will also get more traces messages. So what is the difference? Any help? Regards Bilal Ghayad Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nat=yes
Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canreinvite
Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP endpoint that does not register
Hi List; Did any one see a SIP endpoint that can work without need to do registeration on the gatekeeper side? If this SIP endpoint existed, then I can configure the host=static, but I am not able to find any SIP endpoint accept to not register (all sip endpoints request to register), but most of H.323 endpoints are working without need to registeration (it is called direct mode). Any one can help and advise if he know any SIP endpoint that does not request a SIP registeraion o I can configure the host=static? Also, configuring the host=static is something different than configuring host=192.168.8.2 (formate: host=ip), correct? Any help? Regards Bilal Ghayad Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP endpoint that does not register
Mediatrix 1204, and i assume their other models as well On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; Did any one see a SIP endpoint that can work without need to do registeration on the gatekeeper side? If this SIP endpoint existed, then I can configure the host=static, but I am not able to find any SIP endpoint accept to not register (all sip endpoints request to register), but most of H.323 endpoints are working without need to registeration (it is called direct mode). Any one can help and advise if he know any SIP endpoint that does not request a SIP registeraion o I can configure the host=static? Also, configuring the host=static is something different than configuring host=192.168.8.2 (formate: host=ip), correct? Any help? Regards Bilal Ghayad Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite
By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the difference between increasing the verbose level and the debug level?
Check it out in the CLI and you will see for yourself. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; What is the difference between increasing the verbose level and the debug level? By increasing the verbose level, then I will get more traces messages and by increasing the debug level, I will also get more traces messages. So what is the difference? Any help? Regards Bilal Ghayad Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the difference between increasing the verbose level and the debug level?
In general keep in mind, asterisk is very user friendly and wont bite :). Trial and error is a good friend to get to know asterisk so that you know what all of these mean. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; What is the difference between increasing the verbose level and the debug level? By increasing the verbose level, then I will get more traces messages and by increasing the debug level, I will also get more traces messages. So what is the difference? Any help? Regards Bilal Ghayad Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
BTW, AFAIK, there is no such thing as host=static it's either dynamic or an IP/Name. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
I suspect if you remove the callerid entry from this device's sip.confdefinition things will work better. On 9/9/07, Apa Minerala [EMAIL PROTECTED] wrote: I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call 778f89593967725f0abe40eb1752504c no reply to our critical packet. What is the critical packet that is not being responded to? Please help. -- Pinpoint customers http://us.rd.yahoo.com/evt=48250/*http://searchmarketing.yahoo.com/arp/sponsoredsearch_v9.php?o=US2226cmp=Yahooctv=AprNIs=Ys2=EMb=50who are looking for what you sell. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
Thanks. My results after applying the patch and recompiling are that the problem can only be replicated with calls from mobile networks. Digits like 160 entered in the digital receptionist by a caller are received by the asterisk server as 16660 sometimes. Other times it is received as 1660. Digits like 1234 are received as 1222334 etc... From fixed lines, there is no problem. Digits are received as they have been sent. Any other pointers? Thanks a lot. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, September 09, 2007 12:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote: I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. A make install will not overwrite any configfile. It will install the asterisk binary and the modules (thus overwriting the existing files) but configfiles will only be overwritten when you run: make samples -- 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? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Behaviour
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. First of all, thank you for you reply The ATA is the Fritz!Box and I tried with different FW version but I have the same behaviour this is part of the sip.conf [180] type=peer username=180 secret=aa callerid=First180 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm [181] type=peer username=181 secret=bb callerid=Second181 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm If those are not the source of trouble, _I_ probably would switch the accounts in the ATA (port A versus port B) and try if the problem sticks with the port or with the account. I tried to switch the account for the two ports but what it is important is only the order in the sip.conf I would also google if there are known problems with my ATA, look if a newer firmware is available, if there are informative messages that are worth a verbatim quote, and get another bottle of beer to keep the sunday relaxation at a proper level. I found some information in german and I do not know it ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. My Uniden phone here uses the stutter dial tone to discover if a message is waiting, and lights up a red light on the phone if there is. How does an analogue phone differentiate between a half ring and a call where someone hangs up quickly? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFG5L2iDQNt8rg0Kp4RApDwAKCakhiLuqAIClqS7M9d7pgq2N0jNQCYuIX0 UwvJNiTkC/544IajMONE+w== =nqBw -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager connection timeout
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rizwan Hisham wrote: hi all, Is there any default timeout for manager connection. If its configurable then plz tell me how. In the sample manager.conf file there is the following: ; If the device connected via this user accepts input slowly, ; the timeout for writes to it can be increased to keep it ; from being disconnected (value is in milliseconds) ; ; writetimeout = 100 - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5MD3DQNt8rg0Kp4RAnShAJ9IZ9tUhocoHMzbm41IjoPeWLKZNACgvdYb CZMv/swKj60dp17jLnkKKjo= =b9Jn -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barton Fisher wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? Basically its going to be g711.ulaw for T1 (USA) and g711.alaw for E1 (rest of world) 99.9% of the time. Unless you have something strange or different, I'd record in ulaw for T1. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5MI9DQNt8rg0Kp4RAgRWAKCL2l8egvLV2Xu3T754KJMzGXrKnQCfboCx aFwrtGNKZ0EbZr176MDZUkY= =HvDo -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by the msg/recv from address. It is the NAT's headache on the endpoints network periphery to send the response from Asterisk to the endpoint. C F wrote: If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
Sir, I am having Asterisk pbx which is running without any problem now i want to connect this with Panasonic pbx with FXS port so, if any body want to call panasonic users than he will call or vise-versa. i want to connect only two extension with Asterisk so, all communication done only on these two line. what is the process and what is the setting in sip.conf and extensions.conf to communicate with Asterisk and Panasonic pbx. Rajeev. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Behaviour
Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita: On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. First of all, thank you for you reply The ATA is the Fritz!Box and I tried with different FW version but I have the same behaviour I have been using FritzBoxes for quite a while, and have not found such strange bugs - except after a Firmware Upgrade. It seems after some upgrades you need to do a factory reset (via the web interface) and enter your data again, else they behave stupidly. this is part of the sip.conf [180] type=peer username=180 secret=aa callerid=First180 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm [181] type=peer username=181 secret=bb callerid=Second181 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm Looks pretty OK to me. Just a stupid idea: Do you have a [general] section before those two? And then, I use type=friend, not type=peer, that _might_ make a difference in how asterisk matches sip.conf contexts to registered clients. 8 From my sip.conf: [sip501] mailbox=01 callerid=501 type=friend username=sip501 secret=lk1j2eu89 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw [sip502] mailbox=02 callerid=502 type=friend username=sip502 secret=1092jd0 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw =8 Note: Those two accounts belong to the same FritzBox. I tried to switch the account for the two ports but what it is important is only the order in the sip.conf That made me think about that friend/peer thingy. I found some information in german and I do not know it The FritzBoxes are popular here in Germany - no wonder, being a German manufactured product and being given away for (nearly) free with any 2-year DSL contract... I like them nevertheless :) BR, HTH Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
