[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
[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] 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
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
[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
