[Asterisk-Users] problem configuring a digium quad E1 card
Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France Télécom) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -, it seems that all channels are configured... So can someone give me an advice on that matter... maybe someone in France who already configured that type of access. Also, I would like that you confirm the type of cable which can be used to connect the card to the Telco : can I use a straight cable or use a crossed cable with pair 1-2 connect to 4-5 ? Thanks Best regards! David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem configuring a digium quad E1 card
Hi again, Can you specify the pin order for each end ? thanks -Message d'origine-De: JOSE MANUEL CORTES DAVID [mailto:[EMAIL PROTECTED]De la part de JOSE MANUEL CORTES DAVIDEnvoyé: mercredi 15 mars 2006 16:28À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: RE: [Asterisk-Users] problem configuring a digium quad E1 card Hi Youneed touse a cross-over E1 cable(not an ethernet cross-over one) Good luck Jose Manuel Cortes David XSemestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA De: [EMAIL PROTECTED] en nombre de David MasureEnviado el: Mié 15/03/2006 9:41Para: asterisk-users@lists.digium.comAsunto: [Asterisk-Users] problem configuring a digium quad E1 card Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France Télécom) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -, it seems that all channels are configured... So can someone give me an advice on that matter... maybe someone in France who already configured that type of access. Also, I would like that you confirm the type of cable which can be used to connect the card to the Telco : can I use a straight cable or use a crossed cable with pair 1-2 connect to 4-5 ? Thanks Best regards! David ___ --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] Hardware combination and type of asterisk configuration
Hi all, I'd like to set up a box with asterisk and the following cards in it : - one E1 card (from digium) - one Junghanns OctoBRI My question is : 1) Is it possible such a configuration ? 2) Because of the Junghanns card, I will have to use the bristuff package, but I'd like to know if this package will also work for the digium card ? Thanks Best regards David Masure ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with a second incoming call on a BRI ZapChannel
Second Post !!! Please help ! Hi, I'm using Asterisk with a BRI Card (HFC Chipset) using the zaphfc driver. I'm encountering the following problem : when the first line is in use and a second incoming call arrive, the console shows the following message : Dec 5 14:40:52 WARNING[2323]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! Does anyone have an idea of what this is ??? FYI : the Asterisk is located in France (so France Telecom as carrier) The worst is that it doesn't do it all the time Any help will be welcome Best regards David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with a second incoming call on a BRIZapChannel
Here are the versions : Asterisk 1.0.6 Bristuff 0.2.0-RC7k Kernel 2.4.20-8 on RedHat 9 I must also tell that I have 8 identical configurations running and I have only one computer doing this problem... I'm facing the problem when at least one of the two lines is already on call Best regards David Here comes more details from the command line interface : Verbosity was 0 and is now 8 received TEI check request for TEI = 127 -- Device 'POSTEDU02' successfuly registered -- Executing Answer(Zap/1-1, ) in new stack -- Accepting voice call from '359577352' to '6203' on channel 0/1, span 1 -- Executing SetVar(Zap/1-1, CALLFILENAME=DU-asterisk-2326-1133864928.0) in new stack -- Executing Monitor(Zap/1-1, wav|DU-asterisk-2326-1133864928.0|m) in new stack -- Executing Dial(Zap/1-1, SIP/PosteDU01||rm) in new stack -- Called PosteDU01 -- Started music on hold, class 'default', on Zap/1-1 -- SIP/PosteDU01-85d1 is ringing -- SIP/PosteDU01-85d1 answered Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Starting simple switch on '[EMAIL PROTECTED]' -- Executing SetVar(USTM/[EMAIL PROTECTED], CALLFILENAME=DU-asterisk-2326-1133864955.2) in new stack -- Executing Monitor(USTM/[EMAIL PROTECTED], wav|DU-asterisk-2326-1133864955.2|m) in new stack -- Executing Dial(USTM/[EMAIL PROTECTED], Zap/g1/0328242018) in new stack -- Called g1/0328242018 == Spawn extension (default, 6203, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/DU-asterisk-2326-1133864928.0-in.wav /var/spool/asteris k/monitor/DU-asterisk-2326-1133864928.0-out.wav /var/spool/asterisk/monitor/DU-asterisk-2326-1133864928.0.wav rm -f /var/spool/asterisk/monitor/DU-asterisk-2326-1133864928.0-* ) -- Zap/2-1 answered USTM/[EMAIL PROTECTED] -- Executing SetVar(SIP/PosteDU01-409e, CALLFILENAME=DU-asterisk-2326-1133864969.4) in new stack -- Executing Monitor(SIP/PosteDU01-409e, wav|DU-asterisk-2326-1133864969.4|m) in new stack -- Executing Dial(SIP/PosteDU01-409e, Zap/g1/0663482134) in new stack -- Called g1/0663482134 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/PosteDU01-409e -- Ignoring callwaiting SETUP on channel 0/0 span 1 0 Dec 6 11:30:30 WARNING[2327]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wi cked!!! -- Hungup 'Zap/1-1' == Spawn extension (default, 00663482134, 3) exited non-zero on 'SIP/PosteDU01-409e' monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/DU-asterisk-2326-1133864969.4-in.wav /var/spool/asteris k/monitor/DU-asterisk-2326-1133864969.4-out.wav /var/spool/asterisk/monitor/DU-asterisk-2326-1133864969.4.wav rm -f /var/spool/asterisk/monitor/DU-asterisk-2326-1133864969.4-* ) -- Hungup 'Zap/2-1' == Spawn extension (default, 00328242018, 3) exited non-zero on 'USTM/[EMAIL PROTECTED]' monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/DU-asterisk-2326-1133864955.2-in.wav /var/spool/asteris k/monitor/DU-asterisk-2326-1133864955.2-out.wav /var/spool/asterisk/monitor/DU-asterisk-2326-1133864955.2.wav rm -f /var/spool/asterisk/monitor/DU-asterisk-2326-1133864955.2-* ) == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up -Message d'origine- De : Tzafrir Cohen [mailto:[EMAIL PROTECTED] Envoyé : mardi 6 décembre 2005 11:07 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] Problem with a second incoming call on a BRIZapChannel On Tue, Dec 06, 2005 at 10:42:11AM +0100, David Masure wrote: Second Post !!! Please help ! Hi, I'm using Asterisk with a BRI Card (HFC Chipset) using the zaphfc driver. What versions of asterisk, bristuff, kernel and linux? I'm encountering the following problem : when the first line is in use and a second incoming call arrive, the console shows the following message : Dec 5 14:40:52 WARNING[2323]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! Could you provide some more context? set verbose 5 and provide some lines from the CLI around that message. Does anyone have an idea of what this is ??? FYI : the Asterisk is located in France (so France Telecom as carrier) The worst is that it doesn't do it all the time Any help will be welcome Any ideas how to reproduce the problem? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth
[Asterisk-Users] Problem with a second incoming call on a BRI Zap Channel
Hi, I'm using Asterisk with a BRI Card (HFC Chipset) using the zaphfc driver. I'm encountering the following problem : when the first line is in use and a second incoming call arrive, the console shows the following message : Dec 5 14:40:52 WARNING[2323]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! Does anyone have an idea of what this is ??? FYI : the Asterisk is located in France (so France Telecom as carrier) The worst is that it doesn't do it all the time Any help will be welcome Best regards David ___ --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] S0 - T0 interfaces question
Hi, In fact, a S0 is like a T0 interface except the fact that it is internal. Normally, the S0 should be powered by a PBX or something. So, normally, you should be ablt to connect to it in TE mode. At our office, I have a PBX (Nortel) with a S0 bus on which I have connected Asterisk...it works without problem... Bets regards David Masure -Message d'origine- De : Antonio eleuterio [mailto:[EMAIL PROTECTED] Envoyé : vendredi 7 octobre 2005 14:20 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] S0 - T0 interfaces question Hello. I am new in the area, and I have a customer asking if asterisk can connect directly to an S0 interface. Usually I only connect it to an operator using T0 or T2 interfaces. This customer has a private ISDN network, and has ISDN telephones. Then he is asking me if asterisk can connect directly to them, using an S0 interface. I don't see any reference to S0 in the digium adapters site, so I don't know if it is supported and how. Can someone help on this please ? What do I need to add to asterisk to support such isdn phones ? Thanks. Regards. Antonio Eleuterio ALCIDIAN Solutions Tél:06 86 68 17 41 Email: [EMAIL PROTECTED] Web:http://www.alcidian.net ___ --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 ___ --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
RE: [Asterisk-Users] monitor using incorrect path
Hi, I'm using Bristuff 0.2.0-RC7k and asterisk 1.0.6 and I'm facing something similar... Nearly all my monitor files are in 2 parts, soxmix doesn't compile them into one file. But I don't think soxmix is to blame because when I run it from the command line, everything is ok... the problem seems to originate from the command line executing soxmix. Help would be appreciate on that matter. Best regards David -Message d'origine- De : Tzafrir Cohen [mailto:[EMAIL PROTECTED] Envoyé : mardi 12 juillet 2005 10:45 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] monitor using incorrect path On Tue, Jul 12, 2005 at 09:52:09AM +0200, Kristof Hardy wrote: Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix //var/spool/asterisk/monitor/SIP-242-027e_1-in.wav //var/spool/asterisk/monitor/SIP-242-027e_1-out.wav //var/spool/asterisk/monitor/SIP-242-027e_1.wav rm -f //var/spool/asterisk/monitor/SIP-242-027e_1-* ) Logging from what exactly? That shouldn't be a problem on any posix system (except cygwin) . '//' is simply translated to '/' . I suspect you have a different problem. And when it is correct, it does: monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/SIP-220-c400_0-in.wav /var/spool/asterisk/monitor/SIP-220-c400_0-out.wav /var/spool/asterisk/monitor/SIP-220-c400_0.wav rm -f /var/spool/asterisk/monitor/SIP-220-c400_0-* ) For the record, I am using bristuff-RC8h (that is, quadBRI and asterisk-1.0.8) on a Debian 3.1. Any ideas on what I might be doing wrong, or does anyone see the same behaviour? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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 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
RE: [Asterisk-Users] Asterisk and DHCP
Hi, I don't think it's goonna work that way ! The phones need to register with the asterisk server. So when you configure the phones, you specify the ip of the asterisk server... so... Another way, if you want to work with dynamic ip address would be to set up a DNS (name resolver) which handles dynamic IP and so, you could configure your IP phone with the name of your asterisk server instead of the IP address... I hope I've helped you. Best regards David Masure -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Envoyé : vendredi 1 juillet 2005 09:15 À : 'asterisk-users@lists.digium.com' Objet : [Asterisk-Users] Asterisk and DHCP Hi all! I am working with asterisk and trying to get it work in DHCP network, where asterisk gets a DHCP address as well as other computers (IP-phones). So far, I've have got asterisk working with static IP address where phones are getting their IP from DHCP server. Is it possible at all to phones to find asterisk server if it gets random IP address from the DHCP server also? I mean, is there some settings in asterisk how I can bypass IP settings and force it to work with dynamic IP address? Thanks for any help and guidance! This mail sent through L-secure: http://www.l-secure.net/ ___ 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 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
RE: [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card oni4l vs Fritz ISDN BRI card on CAPI
Hi, Bristuff works great with HFC card... your compilation problem may come from your kernel configuration... You should check this doc, at least, for the redhat config : http://www.automated.it/guidetoasterisk.htm Then, installation of Bristuff works like as charm ! Bye David Masure -Message d'origine- De : vdasilva [mailto:[EMAIL PROTECTED] Envoyé : mardi 28 juin 2005 09:01 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card oni4l vs Fritz ISDN BRI card on CAPI Hello I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have choppy sound problems sometimes, and echo problems often. I am using a 2 port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000 I read that changing to BriStuff will fix the echo problems, but have also read other users say that the only way they solved the echo/choppy sound problems was using a Fritz ISDN card with the CAPI drivers... I have tried using bristuff on RH9 but couldn't get my zaptel to compile... Then there is the issue of timing, ztdummy or zaprtcand QoS setup on the Linux box... Can anyone who has a 100% working Asterisk implementation using any of the techniques described above tell me more... I will happily upgrade to the Fritz card if it will solve all the problems... Thanks Vicente -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: 07 April 2005 09:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Choppy sounds after transferring to ISDN clientor after a time Hi, Michiel van Baak schrieb: I had the exact same thing when using a cheap HFC-S card connected to my outside ISDN line. Replacing the card with an AVM Fritz!PCI fixed this issue for me. I tried a lot with the HFC-S card, different archs, SMP, uniprocessor, nolapic, noapic, dual channel ram setup, 1 dimm only, removed all cards but the HFC-S, 4 different versions of bristuffed. Nothing solved it. What do you mean? The choppy sound or the log message? Trouble is, my HFC-S is internal ISDN. I can't use a Fritz!PCI for that because it isn't capable of NT mode... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ 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 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 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] zaphfc: empty HDLC frame or bad CRC received
Hi all, I'm using asterisk 1.0.6 with bristuff-0.2.0-rc7k. I've already set up 3 boxes with the same config, but I'm facing something strange with the fourth one : In my messages log, I've got thoses lines : kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 17, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 10, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 7, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 10, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 35, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 10, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 23, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 8, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 13, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 11, stat = 0xff).kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff). This problem causes asterisk to crash... The only thing to do is a nice reboot :-) I read some informations about it in the list telling that's because of a wrong signalling type... But as, I am using the same config everywhere, I don't believe the problem is on the box ... Questions 1) The asterisk box is located in France...so could this be that the telco (France Telecom) is powering it's lines with different signalling (for example mutli point and point to point ?) 2) Does anyone ever face the same problem ? 3) Does someone know how to cope with this problem ? Thanks for your help ...I'm really don't know what to do !!! David ___ 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] Is there a problem when we want to transfer an incoming call to an external phone number
Hi, I'm facing something strange but maybe I haven't the right solution. What I want ot do is : Someone from outside call my phone number, I check some informations using an IVR script and then I want to transfer the callto an externalphone number. The point is that when I'm doing that using the Dial command, I've got a message telling me no is able to answer the call. But ifI configure it to ring aninternal extension, it works ! Is there an issue when we want to transfer an incoming call toan externalline ? I'm using the bristuff package and I'm using a HFC pciisdn card. If you can enlighten me.. thanks a lot David Masure ___ 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
RE: [Asterisk-Users] Is there a problem when we want to transfer anincoming call to an external phone number
What do you want to knwo exactly ? The way I use to transfer the call is the same if it was an internal extension, I use the Dial command using the trunk to go outside ... Ask me the info you need ... Thanks David -Message d'origine- De : Bob Goddard [mailto:[EMAIL PROTECTED] Envoy : mardi 14 juin 2005 12:52 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Is there a problem when we want to transfer anincoming call to an external phone number On Tuesday 14 Jun 2005 11:45, Bob Goddard wrote: On Tuesday 14 Jun 2005 09:16, David Masure wrote: Hi, I'm facing something strange but maybe I haven't the right solution. What I want ot do is : Someone from outside call my phone number, I check some informations using an IVR script and then I want to transfer the call to an external phone number. The point is that when I'm doing that using the Dial command, I've got a message telling me no is able to answer the call. But if I configure it to ring an internal extension, it works ! Is there an issue when we want to transfer an incoming call to an external line ? I'm using the bristuff package and I'm using a HFC pci isdn card. If you can enlighten me.. thanks a lot Little info to go on, but I'll hazard a guess and say you only have a single analogue line. B*ll*cks, next time I'll read the complete message Still, we need more info. B ___ 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 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] Sipura 841 and headset
Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to do ? Anyone could help me on that matter ? Orcan anyone advise me on headset working with the sipura 841 ? Thanks Best regards David Masure ___ 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
RE: [Asterisk-Users] bri error
Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoyé : vendredi 29 avril 2005 12:08 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ 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 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
RE: [Asterisk-Users] bri error
The problem may then originate from the NT of your telco -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:21 : David Masure Cc : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] bri error if I do a zttool it shows TE mode On Fri, 2005-04-29 at 12:14, David Masure wrote: Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ 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 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] Bristuff and Belgium
Hi all, Does anyone has any experience with bristuff in Belgium... ??? I have an ISDN line at home and I try to install asterisk (the bristuff version 1.0.6). It works although I received a lot of messages from zaphfc telling me it didn't receive the correct number of frames for both lines. Sometimes too much, sometimes too few... Probably a buffer overrun it says Anyone hava encounter that problem??? is there a cure doc ? Thanks Regards David ___ 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] CDR and TDS
Hi, I wantto use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds but when I compile, I've got this message Does anyone have the same problem and/or know how to solve it ? Thanks Baste regards David Masure make[1]: Entering directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr'gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"1.0.6-BRIstuffed-0.2.0-RC7k\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -fPIC -c -o cdr_tds.o cdr_tds.ccdr_tds.c: In function `mssql_connect':cdr_tds.c:415: `TDSCONNECTINFO' undeclared (first use in this function)cdr_tds.c:415: (Each undeclared identifier is reported only oncecdr_tds.c:415: for each function it appears in.)cdr_tds.c:415: `connection' undeclared (first use in this function)cdr_tds.c:460: warning: implicit declaration of function `tds_free_connect'/usr/include/ctype.h: At top level:cdr_tds.c:71: warning: `connect_time' defined but not usedmake[1]: *** [cdr_tds.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr'make: *** [subdirs] Error 1 ___ 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] Bristuff and startup scripts
Hi, I'm not the kind of Linux guru and I was wondering how I could start automatically the Zaphfc script. What I mean is that before starting asterisk, I have to type : make load from the zaphfc directory in order to load the zaptel driver. How can I do that automatically. This can be very useful in case of unattended reboot, Thanks Best regards David Masure ___ 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
RE: [Asterisk-Users] HFC-S
Hi, Look at this page page http://www.junghanns.net/asterisk/downloads/ and get the latest version of bristuff which should be : http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC7k.tar.gz This package is specific for BRI adapter using the cologne chipset. It works great ! Best regards David Masure -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Envoyé : mardi 29 mars 2005 14:04 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] HFC-S Hi! I have just installed Redhat 9 and Asterisk to my computer, and now i have problems with my non-zaptel Card, I don't know how to set it up since all instructions are for digium's hardware. I have searched from the Internet for hours now, can you help me to understand all this HFC-s thing and how it is related to CAPI, ISDN4Linux, bristuff and so on. I have to say that I am not so familiar with Linux. Thank you in advance This mail sent through L-secure: http://www.l-secure.net/ ___ 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 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] Unable to create Zap channel when dialing using a bri cellular gateway
Hi all, I have an asterisk box set up with a bri card (using zaphfc). I have a bri cellular gateway connected to it beacuse I'd like to route all my cellular calls through that gateway. The probel I encounter is that when trying to dial a phone number, I've the message : unable to create a zap channel. My card is normally well configured because when connected to the NT, It works perfectly... The gateway is configured as a NT as well so no worry... Has anyone an idea of what I should look for ? Thank you David Masure ___ 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] Multiple lines
Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ 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
RE: [Asterisk-Users] Multiple lines
Dean, Thank you for your answer but fromwhat I know meetme is able to solde the conference problem, but how can I for example receive 2 phone calls at the same time on 1 phone and just switching from one line to another ? In my current config, I make a phone call and the SIP phone is answering, when trying to make a second call, I've got the music to hold me till first conversation has ended. Meanwhile, the sip phone user doesn't know there is a call waiting and so, he won't answer the line Is there a solution to that problem ? Thanks David -Message d'origine-De: dean collins [mailto:[EMAIL PROTECTED]Envoyé: mercredi 2 mars 2005 17:02À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: RE: [Asterisk-Users] Multiple lines David, please search the wiki for meetme rooms; this is a standard feature. If you want to be able to the control those calls from a web interface do a search for meetme2 If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David MasureSent: Wednesday, March 02, 2005 9:58 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Multiple lines Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ 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
RE: [Asterisk-Users] Asterisk and #
Don't forget to restart your * bos. A simple reload won't work... David Masure -Message d'origine- De : Marco Ziglioli [mailto:[EMAIL PROTECTED] Envoyé : jeudi 24 février 2005 18:51 À : Asterisk ml post Objet : [Asterisk-Users] Asterisk and # Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco ___ 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 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
RE: [Asterisk-Users] How to monitor Agen Voice channal?
Hi, In your agents.conf file you just have to add the following entries : recordagentcalls=yes recordformat=gsm (or wav,...) createlinks=yes savecallsin=/var/spool/... (the directory you want ot use) Best regards David Masure -Message d'origine- De : Aram Ter-Martirosyan [mailto:[EMAIL PROTECTED] Envoyé : jeudi 24 février 2005 22:50 À : 'Asterisk Developers Mailing List'; asterisk-users@lists.digium.com Objet : [Asterisk-Users] How to monitor Agen Voice channal? Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in conference room - after the customer hangs up the agent session stays connected and there is no way to disconnect agent session but to restart Asterisk - is this a know problem? Is there a solution for this? But in any case if possible to monitor voice channel of the agent without placing them in conference room we will prefer to use that option. Thank you in advance for help. Aram Ter-Martirosyan ___ 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 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] Monitor and Record : audio quality
Hi Folks, I've been experiencing something very strange... When I want to listen to call between a SIP phone and a Zap Channel, I can listen a with a nice audio quality. When it comes to record using the monitor command, I just have a wav file which is completely noisy and I can't hear the conversation. Questions ? 1) Is there a way to specifiy the audio quality using monitor ? 2) Am I missing something else ? The SIP phone is configured with alaw codec (which should be fairly good) and the other party is calling through the Zap Channel which is a HFC chip-based ISDN Card. Any idea ? Best regards David Masure [EMAIL PROTECTED] ___ 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] Nortel Phones.
Hi, I saw your messages related to the Nortel I 2004. I downloaded the code but I have some trouble with the installation, could you give me some more informations... The trick is that I cannot compile (the make). It must be because I didn't make the changes in db.c but after reading your doc, I don't get what I have to do ! Could you tell me which line I have to change and what do i have to put in there ? A sample would be appreciate... Thank you for your help David Masure ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
