[asterisk-users] Call drop and strange CDR records
Hi, I have some (5-10 per day on an average 250 calls/day) incoming calls dropped after 25 to 60 seconds. Asterisk is 1.2.10 + BriStuff 0.3.0-PRE1s on one hand (with 4 ISDN lines...) Snom 320 SIP IP Phone (release 6.2.3) on the other. With SIP Debug on, it *_looks_* like a normal call clearing, but the users are complaining, stating that no one on either end had hanged up. Doing a "sip debug" on asterisk shows a normal call clear from the snom (Receiving a SIP BYE frame). But, the CDR record looks strange (and this is the only common point between those calls): Both the session timer and the talk timer are the same, but according to the log, the call are all answered after 3 to 5 seconds ringing (so those timers should show this difference). Thanks, Fabrice Cahen -- Fabrice Cahen Consultant - Bluesat SARL 20-22, rue de Nantes 75019 Paris T: 01.40.18.75.75 F: 01.40.18.75.71 P: 06.80.70.97.60 M: fabrice DoOoT cahen AoT bluesat DoOoT fr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT
Le Vendredi 3 Février 2006 13:54, Dave Cotton a écrit : > On Fri, 2006-02-03 at 09:52 +0100, Wilson Pickett wrote: > > > Have you seen that 3 Asterisk servers were running during this show ? > > > > François, > > > > I was there (had a coffee with Dave in fact) but was wondering, there > > was no official asterisk presence, was there? Maybe we should have > > helped organize this as * is a "Linux Solution" > > Good idea, and we've got 362 days to organise it. I'd be ready to do it. > It could be in the village or even a proper stand, what do the rest of > the French users think? Hello, It's A Good Idea . We have allready made some Asterisk Presentation on OpenSource Day In Alsace It Fun to Discuss Open Source , Linux and Asterisk. fabrice ___ --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 hardware.
Hello all, Just a question, on asterisk box : I looking on the web , for asterisk at large , and 'asterisk future of telephonie' ... If we would like to change our OLD PABX 600 phone with 4 E1, to install a asterisk with full ip phone in SIP, Could we use 1 Box for asterisk with voicemail, zap channels and some agi script ? thanks Fabrice ___ --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] Memory PB.
Hello, I have some Pb with my asterisk . The box runs out of memory We have 1 Gb of Memory , try on different box, we different version of asterisk ( 1.0.7 => 1.2.1) unable to stabilise memory. the mmlogs : 1136224256 - WARNING: Freeing unused memory at 0x679600, in dial_exec_full of app_dial.c, line 1062 1136224303 - WARNING: Freeing unused memory at (nil), in ast_yyfree of ast_expr2f.c, line 2533 1136224303 - WARNING: Freeing unused memory at (nil), in ast_yyfree of ast_expr2f.c, line 2533 1136224303 - WARNING: Freeing unused memory at (nil), in ast_yyfree of ast_expr2f.c, line 2533 1136224303 - WARNING: Freeing unused memory at (nil), in ast_yyfree of ast_expr2f.c, line 2533 1136224304 - WARNING: Freeing unused memory at (nil), in ast_yyfree of ast_expr2f.c, line 2533 Is there a solution ??? Thanks and HAPPY NEW YEAR ALL ___ --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] Pb musiconhold with G729 codec
Hi, When i place a call on hold, and then return to it, the caller then hears my voice in a delay usually equal to the amount of time i put them on hold. I have the problem only with G729 codec and with my voip provider (i live in france and i use wengo) My configuration : - Pentium III 550 Mhz + 256 Mo Ram - [EMAIL PROTECTED] 1.5 - Grandstream 102 IP Phone - TDM400p card (2FXO + 1 FXS) - 3 licences G729 Codec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie X100P question
Hello, I just bought a X100P from digitnetworks. It is supposed to be a FXO card, but there are 2 rj-11 plug on the card. One is labelled "phone" and the other "pstn". When i plug the "pstn" on the wall and the "phone" on my analog phone, everything (incoming and outgoing calls) works like before (without asterisk). AFAIU, i should have an FXS card in my box to be able to use my analog phone, so why does it work this way ? Second question, what is the cheapest card to use one analog phone only (TDM400 is too expensive). I read there's a S100U which seems to be a single FXS card, but I can't find a webshop selling it. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP phone recommendation
Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ------ Fabrice Tereszkiewicz Sawadka.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to get SetMusicOnHold working...
There are my logs :
Asterisk Ready.
*CLI> -- Starting simple switch on 'Zap/1-1'
-- Executing SetMusicOnHold("Zap/1-1", "default") in new stack
-- Executing ResponseTimeout("Zap/1-1", "20") in new stack
-- Set Response Timeout to 20
-- Executing Dial("Zap/1-1", "OH323/192.168.1.215") in new stack
-- Called 192.168.1.215
Everything seems to be ok...
On Tue, 2003-07-01 at 12:58, Fabrice Tereszkiewicz wrote:
> Hello,
>
> I'm trying to do something really easy : transfer a PSTN call to a H323
> client. This works great. Now I'm trying to use the SetMusicOnHold
> function. I din't find any doc about it, I've just seen some mails in
> the list archive, but it still doesn't work.
>
> That's my extension.conf :
>
> [incoming]
> exten => s,1,SetMusicOnHold,default
> exten => s,2,Dial(OH323/192.168.1.215)
>
> really short...
> my musiconhold.conf :
>
> [classes]
> default => mp3:/var/lib/asterisk/mohmp3/
>
> and there are mp3's in my /var/lib/asterisk/mohmp3/ directory. I can
> hear music when I use :
>
> exten => s,4,MP3Player(/var/lib/asterisk/mohmp3/rem.mp3)
>
> So I don't know what to try, thanks for your help
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to get SetMusicOnHold working...
Hello, I'm trying to do something really easy : transfer a PSTN call to a H323 client. This works great. Now I'm trying to use the SetMusicOnHold function. I din't find any doc about it, I've just seen some mails in the list archive, but it still doesn't work. That's my extension.conf : [incoming] exten => s,1,SetMusicOnHold,default exten => s,2,Dial(OH323/192.168.1.215) really short... my musiconhold.conf : [classes] default => mp3:/var/lib/asterisk/mohmp3/ and there are mp3's in my /var/lib/asterisk/mohmp3/ directory. I can hear music when I use : exten => s,4,MP3Player(/var/lib/asterisk/mohmp3/rem.mp3) So I don't know what to try, thanks for your help -- Fabrice Tereszkiewicz <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Duplicate numbers with outbounding calls(RESOLVED)
It was just an echo problem... reducing volume on the quiknet device does the tricks ! thanks Fabrice Tereszkiewicz On Tue, 2003-05-27 at 18:45, Fabrice Tereszkiewicz wrote: > To test it, I've add this in my extensions.conf : > > exten => 0684357917,1,Playback,demo-thanks > > And I've test it many times without any error... > > On Tue, 2003-05-27 at 18:26, Michael Manousos wrote: > > Fabrice Tereszkiewicz wrote: > > > I've a problem with my X100P card. > > > > > > I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I > > > call an PSTN phone number, some digits are duplicated, so I'm unable to > > > call the right person. > > > > > > Not very clear ? I'll try to do better (sorry, I'm french...) > > > example : > > > I use ohphone (with quicknet hardware), I call asterisk > > > (*192*168*1*204#), asterisk answers, I choose "9" (to do an extern call, > > > see my extensions.conf below), so I've a dial tone. Now I call > > > "0684357917" with my OH323 client but asterisk calls something like > > > "06884335779117" > > > > Can you verify, from the ohphone console, that the numbers > > are typed correctly (and not duplicated)? > > > > > > > > Is it a problem with the french PSTN network ? > > > > > > This are my conf files : > > > > > > -> extensions.conf > > > > > > [incoming] > > > > > > ; jouer un fichier des le debut.. > > > exten => s,1,Playback,demo-thanks ;for playing a file > > > > > > ; 9 pour appeler un num exterieur (VoIP->PSTN) > > > ignorepat => 9 > > > exten => 9,1,Dial(Zap/1-1/) > > > exten => 9,2,Congestion > > > > > > -> zapate.conf > > > > > > [channels] > > > signalling=fxs_ks > > > context=incoming > > > channel=>1 ;X100P > > > > > > > > > > > > Michael. > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Duplicate numbers with outbounding calls
To test it, I've add this in my extensions.conf : exten => 0684357917,1,Playback,demo-thanks And I've test it many times without any error... On Tue, 2003-05-27 at 18:26, Michael Manousos wrote: > Fabrice Tereszkiewicz wrote: > > I've a problem with my X100P card. > > > > I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I > > call an PSTN phone number, some digits are duplicated, so I'm unable to > > call the right person. > > > > Not very clear ? I'll try to do better (sorry, I'm french...) > > example : > > I use ohphone (with quicknet hardware), I call asterisk > > (*192*168*1*204#), asterisk answers, I choose "9" (to do an extern call, > > see my extensions.conf below), so I've a dial tone. Now I call > > "0684357917" with my OH323 client but asterisk calls something like > > "06884335779117" > > Can you verify, from the ohphone console, that the numbers > are typed correctly (and not duplicated)? > > > > > Is it a problem with the french PSTN network ? > > > > This are my conf files : > > > > -> extensions.conf > > > > [incoming] > > > > ; jouer un fichier des le debut.. > > exten => s,1,Playback,demo-thanks ;for playing a file > > > > ; 9 pour appeler un num exterieur (VoIP->PSTN) > > ignorepat => 9 > > exten => 9,1,Dial(Zap/1-1/) > > exten => 9,2,Congestion > > > > -> zapate.conf > > > > [channels] > > signalling=fxs_ks > > context=incoming > > channel=>1 ;X100P > > > > > > > Michael. > > > > > > > > > > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Duplicate numbers with outbounding calls
I've a problem with my X100P card. I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I call an PSTN phone number, some digits are duplicated, so I'm unable to call the right person. Not very clear ? I'll try to do better (sorry, I'm french...) example : I use ohphone (with quicknet hardware), I call asterisk (*192*168*1*204#), asterisk answers, I choose "9" (to do an extern call, see my extensions.conf below), so I've a dial tone. Now I call "0684357917" with my OH323 client but asterisk calls something like "06884335779117" Is it a problem with the french PSTN network ? This are my conf files : -> extensions.conf [incoming] ; jouer un fichier des le debut.. exten => s,1,Playback,demo-thanks ;for playing a file ; 9 pour appeler un num exterieur (VoIP->PSTN) ignorepat => 9 exten => 9,1,Dial(Zap/1-1/) exten => 9,2,Congestion -> zapate.conf [channels] signalling=fxs_ks context=incoming channel=>1 ;X100P ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
