[Asterisk-Users] beronet bristuff
Hi. Im trying to get a Beronet QuadBRI card work with bristuff drivers. Though qozap module loads right, all card spans are in deactivated status. Im quite sure my configuration is correct and using a single BRI card instead of the quadBRI the status is active and I can place and receive calls. On Beronet installation manual I read that Beronet and Junghanns cards are identical in their construction but Junghanns made bristuff so that only their cards can work with their drivers. In the same document and googling around I found that bristuff source files must be patched to recognize other cards. Anyone has experienced with this? The alternative would be using mISDN, but Im not sure its as fine as bristuff. Thanks, _fangi_ ___ --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
R: [Asterisk-Users] Re: courtesy message calling mobile phones
Calling any unreachable mobile from Fastweb or Telecom PSTN I don't get any courtesy message. The same happens when calling an inexistent number. I'm configuring two PBX's, connected to two different phone lines, both behave this way. Perhaps there's some missing zapata parameter? Regards, _fangi_ Well, it's funny because here, now (Italy; Telecom Italia PSTN calling Wind mobile), I do get the courtesy message saying that they're moving me to voicemail, if I call myself from the office PBX to my mobile Wind number, and the cellphone is switched off. ___ --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] pickup problem on Asterisk 1.2.4
I was wrong. The problem was with chan_sccp library and was solved downgrading from version 20060207 to 20060204. _fangi_ -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Francesco Angi Inviato: venerdì 24 febbraio 2006 10.00 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [Asterisk-Users] pickup problem on Asterisk 1.2.4 Solved the problem downgrading zaptel 1.2.4 to 1.2.3. Mantaining the same configurations now everything works fine. Regards, _fangi_ ___ --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] courtesy message calling mobile phones
Hi everybody. Just noticed that when calling a mobile phone, Asterisk doesnt bridge the voice message by telco if mobile is unreachable, but keeps on ringing till it receives a hangup signal. I think this is due to the fact that the message is played without the call has been answered, but Im wondering if theres some way to let Asterisk realize it. All I see in the CLI is the line PROGRESS with cause code 0 received. Thank you, _fangi_ ___ --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] pickup problem on Asterisk 1.2.4
Solved the problem downgrading zaptel 1.2.4 to 1.2.3. Mantaining the same configurations now everything works fine. Regards, _fangi_ -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Francesco Angi Inviato: martedì 21 febbraio 2006 14.35 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [Asterisk-Users] pickup problem on Asterisk 1.2.4 Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply Asterisk doesn't notice a phone picked up! Upgrading Asterisk I only did some changes to my dialplan, according to the upgrade page. My card is a TE110P, this is my zapata file: [channels] language=it context=default signalling=pri_cpe switchtype=euroisdn overlapdial=yes pridialplan = unknown prilocaldialplan = unknown nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=no group=1 language=it musiconhold=default channel = 1-15,17-31 Thanks for help, _fangi_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] queue behaviour
That's exactly what I was looking for. By the way, I discovered Local channels to fork into dialplan. I also discovered that roundrobin policy does not work as I expected, but that's another story. Thanks for help, _fangi_ -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Chris Bagnall Inviato: lunedì 20 febbraio 2006 20.21 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] queue behaviour What I'm trying to do is accepting a call from pstn, put it into a queue, while callee is waiting contact some numbers till one responds, then bridge the two calls. What I can't manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I've no idea if this'll work in practice, but the theory seems sound: 1) Create some extensions in your dialplan which dial the numbers you want the queue to try: exten = 1000,1,Dial(dialstring here) exten = 1001,1,Dial(second dialstring here) etc. 2) Assign members to your queue as follows: member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] etc. 3) Set the queue to ringall or round robin as required. 4) let the list know whether it worked or not :-) Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] pickup problem on Asterisk 1.2.4
Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply Asterisk doesn't notice a phone picked up! Upgrading Asterisk I only did some changes to my dialplan, according to the upgrade page. My card is a TE110P, this is my zapata file: [channels] language=it context=default signalling=pri_cpe switchtype=euroisdn overlapdial=yes pridialplan = unknown prilocaldialplan = unknown nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=no group=1 language=it musiconhold=default channel = 1-15,17-31 Thanks for help, _fangi_ ___ --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] queue behaviour
Hi folks, need some help on queue behaviour. What Im trying to do is accepting a call from pstn, put it into a queue, while callee is waiting contact some numbers till one responds, then bridge the two calls. What I cant manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I also tried AGI and Asterisk Manager, with the same result. I think Id need some kind of multi-threading. Any ideas? Thanks, _fangi_ ___ --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] presence and Asterisk crash
Hi all. Ive got Asterisk CVS Head running on Fedora Core 3. It has been running for 4 months with no particular problem. Recently I tried to enable presence. On dialplan I added hint extensions for all my SIP users and on my Eyebeam clients (v. 1.1 3008q) I set Peer-to-Peer presence mode. Presence works right, but when an incoming or outogoing call is answered, Asterisk crashes with the following message: Ouch ... error while writing audio data: : Broken pipe Segmentation fault I tried to restart Asterisk many times but it always stop with this message. As I disable presence (on Eyebeam clients, not even in Asterisk dial plan) Asterisk stays on. Is this a bug or do I miss something with presence? Thank you, _fangi_ ___ --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] Cisco 7905 sccp Hold and Message buttons
Francesco Angi ha scritto: Two simple questions about Cisco 7905 on Asterisk using chan_sccp. 1) using both sccp firmware 5.0 and 6.1 I cannot put a call in hold, because there's no Hold Button at all! Is there a way to configure The 7905 has an hard button for the hold stuff, the button is the one on the top of the button 1 It wasnt hard. Next time Id rather read user guide first. But in SIP there was a Hold button on display. element in SEPmac_address.cnf.xml, I also put it into sccp.conf, but pushing Message always dials 8500. vmnum = 123456 in the line section Set it on sccp.conf SergioThanks,_fangi_ ___ --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] Cisco 7905 sccp Hold and Message buttons
Hi all. Two simple questions about Cisco 7905 on Asterisk using chan_sccp. 1) using both sccp firmware 5.0 and 6.1 I cannot put a call in hold, because there's no Hold Button at all! Is there a way to configure buttons? Perhaps through XML? 2) How can I configure Message button to dial my voicemail number (and not default 8500)? I tried to enter the number into the messagesURL element in SEPmac_address.cnf.xml, I also put it into sccp.conf, but pushing Message always dials 8500. Thank you, _fangi_ ___ --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] unloading TE110P bristuffed module cause kernel panic
Hi folks, I've already searched the mailing list but no one else seems to have my same problem. I'm using Asterisk with the following configuration: Fedora Core 4 (but I also tried Fedora 3) 1 Digium TE110P 1 TDM40B 1 HFC-S 'Cologne' bristuff 0.2.0-RC8o (zaptel 1.0.9.2) I compiled right, I can load kernel modules but when I try to unload wcte11xp module (the one for TE110P card) I get a kernel panic: Kernel panic - not syncing: /usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:333: spin_lock(usr/src/zbristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c:d9640004) already locked by /usr/src/bristuff-0.2.0-RC8o/zaptel-1.0.9.2/wcte11xp.c/887. (Not tainted) This happens if I load and unload by zaptel script or if modprobe or insmod 'by hand', then run ztcfg and the unload the module. No bristuffed zaptel works right and bristuffed zaptel module for TDM40B works right. The card does not share IRQ with other devices, anyway I tried to have only TE110P mounted on PCI slot and to change PCI slot where card is mounted. Nothing to do. I really don't know what else I can try. Thanks for help, _fangi_ ___ --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] unloading TE110P bristuffed module cause kernelpanic
Same problem with debian sarge on a dell and asterisk 1.0.7 from packages, unloading the module freezes the system, (rebooting the machine worked right), I installed zaptel 1.2beta and it seems to work, but I haven't really tested it, only loaded/unloaded/loaded and placed a couple of calls. Interesting. The zaptel part of the bristuff patch is rather small and does not seem to have much to do with locks or with the init code, at first glance. Also note that the zaptel patch actually applies cleanly to 1.2 . Contact me by email for packages. Though I suspect that it is not going to solve the problem. Patched zaptel 1.2.0beta1 with bristuff 0.2.0-RC8o and now modules load and unload well. Now have to try Asterisk, but this is another story... Thanks for help, _fangi_ ___ --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
