Re: [Asterisk-Users] pickup a call in queue
Time Bandit wrote: If you have FOP, and if the call come in thru a ZAP channel, you can drag the ZAP channel to your extension. This should work. As for a way to make this happen from the manager API, I don't know. Okay, thanks. We found a way to do it through the manager (i suppose FOP does it that way also) by using: Redirect($channel, $extrachannel, $to, $context, $priority); cheers! ___ --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 a call in queue
Hello, We are faced with a problem concerning queues. When we have several calls in different queues, is there some sort of way to open a channel between a (sip-)phone and a SPECIFIC call in a queue using the Asterisk manager api? We would like to do this even when we are not a member of that specific queue. Thanks in advance for any suggestions! cheers ___ --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] Search for Links for Communicating PC to PC in the same lan through Asterisk
John Joseph wrote: I am trying to do some simple experiment with Asterisk . my intention is to communicated two PC in start your experiment with [EMAIL PROTECTED] (http://asteriskathome.sourceforge.net/), it's very good to start with. cheers ___ --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] which gui for asterisk on web
Toygun Mavinil wrote: I tried AMPortal, it added extensions to mysql but asterisk did not find users i added İ installed asterisk 1.2.2 on FC4 this also looks promising: http://www.voiceone.it/ cheers ___ --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] GXP-2000 fw 1.0.1.13 and NTP
Was there a resolution to this issue? The GXP-2000 seems to be a very popular phone, so I can't imagine others on the list not experiencing this? Or is this part of a batch with unresolvable problems that I need to send back to the seller? Well, I'm using dozens of these phones without this problem. What kind of DHCP/ntp server are you using? I'm using dnsmasq on a Debian box, together with the ntp-server. I'm using a mixture of 1.0.1.13 beta and .12 firmwares, both working correct. Thanks! TGIF! :') Hell yeah! :D cheers ___ --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] Fritz card technology German *
Chris Earle (CBL) wrote: I have also heard about BERONET isdn cards? a single Beronet 4-channel card would suffice I think? Yes. Beronet and Junghanns both have the same cards. (they just 'work' different, junghanns uses zap interfaces, beronet mISDN) So, as already mentioned, you have 2 good options: - 4x BRI card (Beronet or junghanns) - 2x HFC PCI card (uses zap, and are cheap!) Regarding the phones, I only use sip phones, so no idea on that.. ___ --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] Fritz card technology German *
Mimmus wrote: Eicon DIVA cards rocks but a quad-BRI costs 1500€ Other models cost this price/100. Or not? A Junghanns quadbri is approx 640€. And 2x HFC-pci ISDN card is 2x30€ or so.. haven't tried this in production yet :) ___ --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 1.2 bristuff and sms
hi there, I've been using sms a few months ago with * v1.0.9, but now I need it, so I'm testing it out again. But for some reason the SMS receiving doesn't work like it should. It receives the call from the telecom operator, and it starts the SMS application, but then i get the following error.. Any idea if this is due to bristuff or my implementation of SMS ? (I used voip-info.org as a reference) thanks in advance log: VERBOSE[7905] logger.c: -- Goto (custom-smsrx,0,1) VERBOSE[7905] logger.c: -- Executing Verbose(Zap/1-1, Receiving SMS from 0171701) in new stack VERBOSE[7905] logger.c: Receiving SMS from 0171701 VERBOSE[7905] logger.c: -- Executing SMS(Zap/1-1, default|a) in new stack VERBOSE[7905] logger.c: -- SMS TX 93 00 6D DEBUG[7710] channel.c: Avoiding initial deadlock for 'Zap/1-1' DEBUG[7710] channel.c: Avoiding initial deadlock for 'Zap/1-1' DEBUG[7905] chan_zap.c: Engaged echo training on channel 1 DEBUG[7905] channel.c: Scheduling timer at 160 sample intervals VERBOSE[7905] logger.c: -- Silence suppression is enabled (option_silence_sup ___ --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] D-Link announces Asterisk on Router/DSL-Modem
Philipp von Klitzing wrote: G3342SB will be provided by Asterisk, and it should become available sometime around Q2/2006 for aprox. 450 EUR (let me add that this is twice as much as the Fritz!Box). Thanks for sharing! for this price, it seems a very good thing. Let's wait untill it's out. I think the idea is not new, but the price indication is nice. ___ --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] Presence support on GrandStream GXP-2000
[EMAIL PROTECTED] wrote: It does with the latest BETA firmware. But it dosn't seem to work to well. It stops working and the phones have to be rebooted. works good, as long as asterisk doesn't get restarted. then you need to reboot the phone. it's a bug. http://www.voip-info.org/wiki/view/GXP-2000 ___ --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] Presence support on GrandStream GXP-2000
trixter aka Bret McDanel wrote: I havent looked, I am sure that its there somewhere on grandstreams site but where is the latest beta located? all info can be found on http://www.voip-info.org/wiki/view/GXP-2000 ___ --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] Recommendations on a WiFi phone for *?
Philip Edelbrock wrote: We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? depending on the needs, Kirk phones are very nice. They are DECT, but they've got an own dect-to-ip system, and that works very nice. The IP-600 together with some repeaters and 4020 or 4040 work very well with chan_sccp. ___ --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] Echo after asterisk has been running for severaldays
Matt wrote: I had read somewhere (but now can't find) that instead of a reboot I can just unload the zap module (after stopping asterisk) and reload it? Can anyone confirm this? I do a nightly shutdown of asterisk, do a ztcfg -s, unload the modules, and then fire it all up again. cheers, Kristof. ___ --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] GXP-2000 any good with * ?
Michiel van Baak wrote: Hinting works fine for me with the latest firmware. What version are you running? We use 1.0.1.9 but the leds next to the speeddials wont use latest * and latest gxp firmware, have a look here on how to do it: http://www.voip-info.org/wiki/view/GXP-2000 ___ --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] GXP-2000 fw 1.0.1.13 and NTP
Peter Bowyer wrote: side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. No problem here. Using the 1.0.1.13 (very beta:)) also, synching with an internal time server on our network. (wich then syncs to pool.ntp.org) cheers ___ --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] spandsp fax
Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well? ___ --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 1.2 mysql cdr garbage
hi, Just wanted to know if anyone else is experiencing 'garbage' mysql call detail records on asterisk v1.2? So, where the à is, tehre should be a number.. Example: 1. 2005-12-16 10:07:08 Local/[EMAIL PROTECTED] à 400Tech: à 210 ANSWERED 00:00 2. 2005-12-16 10:02:43 SIP/206-69... à 400Tech: à 210 ANSWERED 04:25 If I restart asterisk, this is solved for a few hours.. Cheers, Kristof. ___ --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] IP Phone Recommendation
Anders Svensson wrote: We use Grandstream GPX2000 for this. It works ok. Support 11 lines in basic. Anders I also use this phone, have read about the 11 lines, but how does one 'manage' these lines? The first 4 are easy, you have buttons for that, but how can you use the 'others' ? (incoming/outgoing) ___ --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] Long and variable echo
The problem has been consistent from 1.0 through CVS to 1.2, and across different machines and distributions. Does anyone have any suggestions on how I can deal with this? I have had echo cancellation happening, but half-duplex speech is not acceptable. You're not using zaptel, what are you using to connect to the outside? (and is it PSTN/ISDN ?) Can you advice on what hardware you're running, that would help.. Regards, Kristof ___ --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] Need advice on BRI
Pedro Nunes wrote: I need to install a production server with BRI support. I know that exists bristuff, misdn, chan_capi ... I have hcfpci based cards. For a very stable environment, what driver should I use?? My personal experience is only on using zaptel, it's also the most 'mature' environment. That's why I'm using bristuff, it works with hfc-based cards and quad/octo BRI cards. The advantage is, you can use the more advanced echo cancellers of zaptel. cheers.. ___ --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] Make list of incoming and outgoing calls
I would like to make a list of all incoming and outgoing calls. From, to, date, duration and for incoming, whether the calls were taken or not and if yes, by which extension. best is to use the built-in cdr options. (search voip-info.org with asterisk cdr) You can try the mysql cdr, it uses mysql to log all what you need. cheers ___ --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] ZapHFC cards not maintaining sync?!
Remco Barende wrote: Weird, I checked with KPJ before and he mentioned it is normal behaviour for ISDN. My console is filled with messages like this : == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down Well, just wanted to share my experiences, over here in Belgium. We do not have this behaviour on ISDN lines. (I'm using a quadBRI from junghanns, but have also used plain hfc pci cards) My signalling type (for the quadBRI) in zapata.conf is: ; p2p TE mode (for connecting ISDN lines in point-to-point mode) signalling = bri_cpe cheers ___ --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] Restore logging functionality...
Chuck Bunn wrote: drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log you can: delete your logfiles, * will re-create them I think or: change the owner to asterisk. (chown asterisk.asterisk /var/log/asterisk/ -R) cheers ___ --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] Grandstream NTP
Rod Bacon wrote: It now appears to be server specific. The shipped default, time.nist.gov, appears to work OK. Does anyone know of anything specific required by these grandstream phones as far as NTP server support goes? I also have GXP2000 pones and use a 'standard' ntp.conf, nothing fancy at all: driftfile /etc/ntp/drift server pool.ntp.org server 127.127.1.1 fudge 127.127.1.1 stratum 10 restrict 10.10.0.0 mask 255.255.255.0 nomodify nopeer notrap ; allow local lan to use the ntp server restrict 127.0.0.1 ___ --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] what is your echo solution
Patrick Fortin wrote: Just wandering what solution worked to eliminate echo on your setup. I am trying every solutions I can find on the wiki and none is working perfectly. I have been (since 1 1/2 weeks) using the ECHO_CAN_MG2. We have got a different setup (quadBRI, 12 GXP-2000's), and untill now it seems to be much better. cheers ___ --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] v1.2 and cdr badly written
Has anyone encountered 'bad' cdr logging in * 1.2? Since upgrading to 1.2 (bristuffed) and asterisk-addons 1.2, sometimes the clid is 'messed' up. I use AMP to look at the reports, but when I look in the cdr database, it's the same, here's an example: 2/12/2005 15:06:02 Tech: ÀB ÀB 2 ext-group Zap/1-1 SIP/211-f379 Dial SIP/209SIP/200SIP/210SIP/211|30|tr 582 582 ANSWERED 3 asterisk-2559-1133532362.58 So, why would the numver be saved as ÀB ?? It's a correct number, but gets written incorrect.. another one: 2/12/2005 16:07:50 ÀB ÀB s aa_4 Zap/1-1 Hangup 3 2 ANSWERED 3 asterisk-2559-1133536070.98 Any suggestions on this? cheers ___ --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] ZapHFC cards not maintaining sync?!
Francesco Peeters wrote: Does anybody have any experience in this? I am using * 1.2 BRIstuffed 0.3.0 Pre1 No experience on that, but there's an updated bristuff (0.3.0pre1b), maybe try that one? This is 1 issue that's fixed: - chan_zap/libpri fixes (stuck B channels) cheers ___ --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] Blind transfer question
Sean Kennedy wrote: Is this a known thing? Can anybody give me an idea of how to change the Blind Transfer key sequence to something else? I assume you're using v1.2. If you change anything in features.conf and then restart asterisk, you can connect to the CLI and do show features to see your current feature list. (that way you are sure * has used your config) cheers ___ --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] How to exit from Asterisk console.
gc wrote: I started * like this: asterisk -vgc now I am in CLI mode: *CLI How do I get out this CLI mode to linux shell without kill asterisk process? if you want to run it like this, first do a screen (more info: man screen) so you can run it in a background shell. But I recommend on running it with the appropriate init script.. cheers ___ --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] Would DECT cordless phones work with Asterisk and VOIP?
Richard Malcolm-Smith wrote: I am so keen on getting the Kirk telecomms 600 system to hook to asterisk, anyone know where to get one from that will ship to New Zealand? You could try asking in the asterisk irc channel, there regularly are some people from New Zealand in there, they might have an idea where to get one from.. If not, I can always have a look on shipping one to you.. (contact me off-list then) cheers ___ --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] Would DECT cordless phones work with Asterisk and VOIP?
Larry Alkoff wrote: In my house, a Uniden 5.8 and Panasonic 2.4 cordless system would only work over about 35 feet indoors - not enough for a large house. Does anyone have any hands-on experience with DECT? I have recently discovered kirk (kirktelecom.com) wich also uses something like DECT, but according to their website, a modified version. Now, what makes this interesting is, you can use 'repeaters' so you can even bridge larger distances. (you can use multiple repeaters, but there is a maximum, check their website) To summarize, I'm using: IP600 and a Kirk4020 and 4040 and a repeater-box. The only drawback is, you can't use regular dect-phones with this system. I found them to be very reliable, and when using chan_sccp, you can interface very smoothly with asterisk. cheers! ___ --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] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
Francesco Peeters wrote: Reshuffled the cards in my machine (actually inverted the order of the PCI cards) and problem solved! Indeed, if the bios is is not working 'with' you, but more against you, then shuffling could solve the problem, glad it did :-) It appears the bottom 2 slots of the MoBo share an IRQ line, period... Now the cards have different IRQs, and both work, *with* the sync connection in place (i.e. the NT running of the TE clock) Brilliant! Now to play with the dialplan and integrate a zap-channel phone (or actually: 4 phones... G) Excellent, now the 'really' fun part starts out :-) 1.2 has 'soo' many possibilities :) cheers! ___ --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] misdn, 2x HFC cards
Denny Schierz wrote: Most howtos discribes asterisk with capi or bristuff. How does the extensions.conf and misdn.conf looks like, so that ISDN (basics) works, for example phone internal from sip to isdn, isdn to sip or isdn to isdn. Is there anybody, who has asterisk 1.2 with misdn running? I had asterisk 1.2 on a testbox with chan_misdn, I used the howto you can find beronet.com (a PDF that can be found in the download section). This was with a quadBRI card from junghanns, but I've now switched that testmachine to bristuff 0.3.0pre1. Cheers ___ --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] Distinctive ring?
Kerry Garrison wrote: pain to configure) have 4 ring types. I am guessing that I would need to figure out how to tell this particular phone to use a different ring tone unless there is a way to send a stutter type ring to the phones. Hi Kerry, I'm also using grandstreams on a few places, have the 'same' issue/question. Afaik it can't be done with the current Grandsteam firmware. (at least, you can't command the phone to use tone X, like you can do with Cisco's) You can use the phone's built-in Distinctive Ring Tone: setting (Advanced settings), but I'm not aware of any 'wildcard' you can fill in there, I only got it to work when filling in an 'exact' number. It could be that the next firmware (should have arrived end of oct) gives us distinctive ring tones and working hint leds.. Let's hope.. If you do find a way to get any working, please report back to the list, meanwhile, i'm eagerly waiting for the firmware :) cheers! ___ --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] Re: Not receiving fax
Wayne Gemmell wrote: Yes, make a 'default' to go directly to your fax-receive macro. (rxfax witht the parameters) At least you should hear a 'fax' answering. Yes, I hear a fax answering, so at least I know its working. Okay, so the ring detection goes wrong. Now at least you know what direction to look in. Hope it'll help.. cheers! ___ --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] Camping-on-busy
Chris Bagnall wrote: Hopefully this will be of use to others on the list once it's working correctly. Here's what I've got so far: Looks very nice, just have to find a way to get it integrated with AMP, but looks promising! It works fine with one exception - when the caller hits 1 whilst camping (to fallback to another extension such as a global hunt group), then hangs Hm, I will try it out on my asterisk system also, after the weekend. cheers! ___ --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] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
Francesco Peeters wrote: I compiled 1.2 and bristuff 0.3.0 Pre1 yesterday late and that now seems to work! * is up and running *with* 2nd card in NT mode... Nice to hear *1.2 and bristuff 0.3pre1 makes a difference.. cheers ___ --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] Not receiving fax
Wayne Gemmell wrote: I'm having trouble receiving faxes using rxfax. Could somebody please browse my log file and give me a swift kick in the right direction? I've also added my zapata.conf file at the end. have you tried using direct indialing, to see if rxfax works? (I assume you are now using fax-detection) That way we know if the detection is failing or the receiving itself.. (or both :)) cheers ___ --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] Re: Not receiving fax
Wayne Gemmell wrote: I'm not sure what you mean, are you saying that I should some how circumvent the menu system to make calls go directly to the fax? Then I should listen for noises? Yes, make a 'default' to go directly to your fax-receive macro. (rxfax witht the parameters) At least you should hear a 'fax' answering. Hm, you could try enabling the busy detection in your zapata file.. ___ --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] [Asterrik-Users] Bristuff for Asterisk 1.2 error
asterisk183 wrote: 5. modprobe zaptel 6. But when I doing insmod qozap.o and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why? what is the output you get 'after' you do: modprobe qozap After this, what is the output you get after: ztcfg -v Also, the qozap.ko file should have been copied to your /lib/modules/your kernel/misc directory. (this is being done by the install script) Cheers ___ --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] [Asterrik-Users] Bristuff for Asterisk 1.2 error
asterisk183 wrote: and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why? If you don't have qozap.o files, then your qozap is not compiled correctly. Try (in qozap dir) a 'make clean' and 'make all' and see if this produces an error. ___ --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] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
My asterisk server breaks as soon as I turn one of the two cards in to an NT card, which I suspect is an issue in the (Florz-patched) BriStuffed chan_zap.so module, but as my cards aren't original Junghanns', they obviously aren't supported by Junghanns... (ZapHFC btw shows both cards active, one as TE Master, the other as NT slave, so that seems to be OK! I therefore doubt it is the Florz patch!) Maybe it's worth a try, using chan_mISDN (experimental, but works!).. You can find the how-to pdf (for beronet, hfc, etc.. cards) on http://www.beronet.com/downloads/. There also is an install-script that helps you through the installation, I have gotten it to work with a junghanns card and 1x HFC pci card. Didn't have a 2nd hfc around to try back then.. I you have results (good/bad) keep the list (or me) posted :) Cheers. ___ --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] ver1.2 installation problem
Bartosz Wegrzyn - asterisk wrote: this is the loop: else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp maybe you don't have the correct user rights on your source tree? try a chown and chmod on the complete source tree. (or do the make install as root) Cheers. ___ --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] Bristuff for Asterisk 1.2
Andrew Nowrot wrote: Does anyone know if there is a new version of Bristuff for Asterisk 1.2 stable. If yes, where can I find it and of course download it :). Yes, it has just been released since last sunday-evening. You can find it (as always) at junghanns.net. Don't forget to check http://www.junghanns.net/downloads/ also, because sometimes new versions don't make it to the homepage :-) Oh, and as the readme in the tgz says, it's not for production use yet.. :) Cheers ___ --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] mISDN and chan_isdn for 1.2
John Martin wrote: Can anyone recommend a version of mISDN and mISDNuser (dates of CVS or archive held on someone’s server) that will work with the chan_isdn in Asterisk 1.2. I have used the install-misdn script on http://www.beronet.com/download/ and that seems to work.. ___ --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] v1.2 and features.conf
Hi, I'm trying to get the 'xfersound' working with v1.2. I enabled all in features.conf (like: xfersound = beep), but I can't get the beep when transferring a call. I'm trying this with * v1.2, the bristuff-version, but I'm not sure if that matters? (does it only work with SIP-to-SIP calls?) Cheers, Kristof. ___ --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] Wireless SIP Phones with Asterisk
Juan Janczuk wrote: Any tried some Wireless IP phones with Asterisk? Comments, recomendations? You could try Kirk phones (http://www.kirktelecom.com/), using chan_sccp you can hook them up to *.. All you need is an IP600 and a few handsets.. Cheers.. ___ --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] Bristuff / Junghanns / Customer Service
Frederic Steinfels wrote: Nethertheless I have found SEVERAL errors that lead to complete hangups and core dumps. I was running gdb for KPJ, writing extensive bug reports and some of those bugs were fixed. Last January I told KPJ that I can still not use my Simens Gigagaset cordless phones and sent him some bug Hi, I've been using some quadBRI's, never had to much problems with them. (I have some echo issues, but I hope these'll be resolved on the next driver update, when we can use new echo can's..) So, I can't confirm the bug(s) you're experiencing, but I must say that I didn't have any hangups or dumps ever.. The driver seems to be stable, but maybe we are using it in a very different situation.. Maybe it's a possibility to try the chan_misdn driver (www.beronet.com has the instructions), it could help you.. they produce approx the same cards but address it through mISDN instead of zaptel.. Cheers! ___ --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] HFC ISDN card and mISDN driver
Hamish Whittal wrote: I am fighting with my ISDN HFC card to get the necessary compiled and working. I'd say, go ahead and try the install-misdn script from beronet (www.beronet.com/downloads), it might solve your problems. Then you'll use mISDN (and not zaptel) to use your card with *. Cheers, Kristof. ___ --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] misdn for BRI
hi all, I just configured a junghanns card like described on beronet. (using mISDN) All seems to work, but I'm more concerned about the 'differences' in echo cancellation. If one uses the junghanns drivers, you're stuck with asterisk1.0.9 at the moment; using chan_mISDN I can use asterisk 1.2.0rc2, but I'm not using the zaptel drivers, and thus not using the new (KB1 or MG2) echo cancellers.. (wich are supposed to be *much* better) Now I'm wondering if anyone has experience or knows of any (echo) differences between the junghanns-way or the chan_misdn way. Cheers! ___ --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] ISDN card required
Lee Archer wrote: Can anyone point me in the direction of a quality, works with Asterisk, BRI card. I need minimum 2 port/4 channel. Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine. Cheers. ___ --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] groupware + unified messagerie +Asterisk
harry gaillac wrote: Is it possible to add a frontend groupware with All is possible, you're only limited by your imagination. (always wanted to say this :p) I'm not sure there's a(n Open-source) project like this already. Cheers.. ___ --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] Multiple zaphfc cards (for ISDN BRI) in a single machine
Steve Davies wrote: I would suggest using a pair of 4-port cards. The interrupts alone from 5 PCI cards would kill most boxes. There is also an octo-card, You also have the BeroNet cards (http://www.beronet.com), exists in 4 and 8 ports. (BN4S0 - 4 S0 interface card) They are almost identical to the Junghanns cards. At the moment I have got experience with the Junghanns cards only. The 'big' difference is the way the cards work. You can use the BeroNet driver, wich uses (2.6 only) mISDN and chan_capi, or the Junghanss-way, wich uses Zaptel. (a modified version, you should use their patches) The only disadvantage I find the Junghanns card has, the driver is not updated to *-v1.2 (or cvs). But this will change when 1.2 is released of course. A + for both cards: you can get them pretty easy in europe, prices are almost the same for the 2. Please let us know your personal choice/results. Cheers! ___ --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] Multiple zaphfc cards (for ISDN BRI) in a singlemachine
Erik wrote: Sorry for this shameless hijack, is there a version of brisuff/zaphfc for 1.2 ? Steve Davies wrote: bad boy :-) there is none as of 'yet'. I guess Junghanns will deliver one as soon as 1.2 is stable. You could try the beronet-way (mISDN and chan_capi). Cheers ___ --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] Open Source Content Management System - Joomla
Kanuri, Seshu (Company IT) wrote: This is relevant where Administrative users wanted to manage their Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc Now you have made me curious :-) In what way could a CMS contribute to this? (just curious because I'm using some CMS for other purposes) Cheers ___ --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] asterisk.org beta site up!
Matt Brooks wrote: I am just emailing to inform you guys that a new website has been created for asterisk.org. You can find the beta site up at http://beta.asterisk.org. It utilizes the drupal portal framework and Looking very good and much easier to navigate! Great work! Cheers, Kristof ___ 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] Should this work?
Angus Comber wrote:
; for dialing outbound - over ISDN line - this bit does not work
exten = _9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten = _9XX.,2,Hangup
Error I get is:
-- Executing Dial(SIP/200-e433, ZAP/g1/902088787367|60) in new stack
Jul 25 11:56:33 NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create
channel of type 'ZAP'
== Everyone is busy/congested at this time
-- Executing Hangup(SIP/200-e433, ) in new stack
== Spawn extension (default, 902088787367, 2) exited non-zero on
'SIP/200-e433'
You should rewrite the calling rule so that the 'outgoing' extension is
not ZAP/g1/902088787367|60 but ZAP/g1/02088787367|60. The 9 should
only be used to 'indicate an outgoing line'. Samples can be found on the
wiki. I'm not 100% sure how to rewrite it so best check voip-info.org.
Cheers,
Kristf.
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Re: [Asterisk-Users] Asterisk 1.2 is getting closer - please help
Olle E. Johansson wrote: There's no good documentation on that out there... yet. Read all the sample configuration files, the READMEs and especially the updating file. Any documentation people with time to write out there? What would be the best place for (this kind of) documentation? On the wiki voip-info.org or do you think other ways are better? Cheers, Kristof. ___ 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] Junghanns quadBRI on Dell PowerEdge
David Hajek wrote: Yes, I tried signalling = bri_cpe_ptmp. When I put the card into older system and use same cables, same ISDN units, same Asterisk configs (but older bristuff!) it works fine. When I put the card into Dell, I got the CRC errors as I wrote before. Maybe someone from Junghanns is watching this thread and can give some help? Can you try the 'older' bristuff on that DELL ? That seems to be the difference between your 2 systems.. You can also try swapping the cable (as in the other post) and try a different PCI slot in your Dell. Cheers Kristof. ___ 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] Junghanns quadBRI on Dell PowerEdge
David Hajek wrote: We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 What are your settings in /etc/zaptel.conf ? I once had the same issue, make sure your signalling is: signalling = bri_cpe Also, check with # cat /proc/interrupts that the quadBRI is not sharing an irq with something else. Jul 19 17:15:55 ustredna kernel: qozap: S/T ports: 4 [ TE TE TE TE ] Jul 19 17:15:55 ustredna kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Your card is found, so hardware seems to be okay.. Best regards, Kristof. ___ 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] Extension Lights Patch
Anton Krall wrote: How does this work? And will it work only on certain phones or can it work with the gxp2000? I think for the GXP-2000 the firmware also needs support for this. (and it's planned for a future version) Cheers, Kristof. ___ 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] Junghanns quadBRI on Dell PowerEdge
David Hajek wrote: [channels] switchtype = euroisdn ; BRI CARD nationalprefix = 0 internationalprefix = 00 signalling = bri_cpe Ouch, we are talking about zapata.conf I guess :-) But, that all seems okay.. Have you tried signalling = bri_cpe_ptmp instead of bri_cpe? Do you use this card in Dell PE 2800? I suspect that this card can't work in 64-bit PCI slots? We do not use it in a Dell, but on SuperMicro with pci-x slots, so that should work I guess. Keep us posted, in my opinion it has to work :-) Cheers, Kristof. ___ 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] SpanDSP rxfax, no tiff
Rob Danz wrote:
If I leave the mailfax step out entirely, then there should be a .tif
file, right? But there’s not. No tif file gets created at all.
Permissions on the fax folder are 777 at the moment.
Are the permissions okay for getting TO that folder? (do you have r+x on
the directories 'above' the fax folder?)
Thanks for the responses so far.
Just trying to help out, I'm using ISDN lines with a DID, for receiving
I do this..
If one calls number X, then Goto(custom-fax,s,1) and then..
[custom-fax]
exten = s,1,Answer
exten = s,2,Macro(faxreceive)
exten = s,3,SetVar(ONZENID=${UNIQUEID})
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE}
[EMAIL PROTECTED] ${CALLERIDNUM} ${CALLERIDNAME} ${ONZENID})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,104,Goto(3)
The [macro-faxrecevive] is from AMP (wich I'm using for managing asterisk)
Hope you can do something with this.. :-)
Cheers
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[Asterisk-Users] SMS in Belgium
Hiya,
I've been doing some testing with SMS in Belgium (Belgacom), sending SMS
seems to work fine. (with .call file and a context that handles the sending)
The problem is however, receiving.. Hardware and software is quadBRI
from Junghanns, bristuff RC8h (asterisk 1.0.8), Debian 3.1.
For receiving SMS in asterisk, I am using:
[custom-smsrx]
exten = s/0171701,1,Verbose(Receiving SMS from ${CALLERIDNUM})
exten = s/0171701,2,Answer
exten = s/0171701,3,Wait(1)
exten = s/0171701,4,SMS(from-pstn,a)
exten = s/0171701,5,Hangup
I do get some logging that shows me it seems to work, but nothing gets
written to the /var/spool/asterisk/sms dir.. Here's the log:
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Goto(Zap/5-1,
custom-smsrx|s/0171701|1) in new stack
Jul 14 15:17:30 VERBOSE[12055]: -- Goto (custom-smsrx,s/0171701,1)
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Verbose(Zap/5-1,
Receiving SMS from 0171701) in new stack
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Answer(Zap/5-1, )
in new stack
Jul 14 15:17:30 WARNING[12055]: Unable to request echo training on channel 5
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Wait(Zap/5-1, 1) in
new stack
Jul 14 15:17:31 DEBUG[12055]: Scheduling timer at 160 sample intervals
Jul 14 15:17:31 DEBUG[12055]: Generator got voice, switching to phase
locked mode
Jul 14 15:17:31 DEBUG[12055]: Scheduling timer at 0 sample intervals
Jul 14 15:17:33 VERBOSE[12055]: -- SMS RX 91 8D 00 0A 81 20...
Jul 14 15:17:33 VERBOSE[12055]: -- SMS TX 95 02 00 00 69 00...
Jul 14 15:17:35 VERBOSE[12055]: -- SMS RX 91 8D 04 0A 81 20...
Jul 14 15:17:35 VERBOSE[12055]: -- SMS TX 95 02 00 00 69 00...
Jul 14 15:17:36 VERBOSE[12055]: -- SMS RX 94 00 6C 0A 81 20...
Jul 14 15:17:36 VERBOSE[12055]: -- Executing Hangup(Zap/5-1, )
in new stack
Any idea.. anyone?
Cheers!
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Re: [Asterisk-Users] OT: proliant fedora asterisk
Freddi Hansen wrote: HP doesn't support Fedora on Proliant hw so you can't just install their ILO and get access to hw info like cpu/mb/temperature,powersupply status,fan info aso. Thanks for sharing the link.. Btw, the same counts for Debian on HP servers.. For that you could visit the howto's at: http://debian.catsanddogs.com/forum/ cheers. ___ 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 using incorrect path
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-* ) 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? Cheers, Kristof ___ 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
Tzafrir Cohen wrote: 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? The output is from /var/log/asterisk/full That shouldn't be a problem on any posix system (except cygwin) . '//' is simply translated to '/' . I suspect you have a different problem. Indeed, when I do this manually (by using soxmix on command line) this works. Any idea why this doesn't work when this gets executed after the hangup? Cheers, Kristof. ___ 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] Soft-switch.org is out?
[EMAIL PROTECTED] wrote: I've been trying to access www.soft-swtch.org for a couple days, but it seems to be down. Anyone got the same? Indeed, same issue here.. If we can help, let us know. (dns/website hosting) Cheers, Kristof. ___ 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] quadBRI form junghanns.net
Ivan Meic (Vox Mundi) wrote: I had quite a lot of experience with it ... it works fine, the only problem I got was that I couldn't transmit fax (data) calls through it reliably ... although this was some time ago, so it is possible that the kernel modules for them improved lately. I can confirm the 'sending' is a bit problematic indeed. (on SuperMicro mainboard, no other issues) Last time we tested was, eh, a few days ago :-) Receiving works perfect. Cheers, Kristof. ___ 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] quadBRI form junghanns.net
Bartosz Jozwiak wrote: Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. I have been testing out with chan_capi and some cheap PCI cards in the past, but since we've used the quadBRI cards, ... I don't want anything else. :-) It's 'very' easy to install, you only need a basic system (we use Debian stable) with the right packages (development, kernel sources, ..) and you can compile the whole package at once. The junghanns.net site gives you the bristuff download, with that you have the most recent quadBRI drivers, and a current asterisk version (1.0.8) in 1 easy to compile and install package.. Just my personal opinion.. But I'm happy with the way these cards work.. Regards, Kristof. ___ 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] Weird ring back
David Wilson wrote: I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN. Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds. When the call is answered there is no one there. What phone are you using? A GXP-2000 or other model? I received notice of someone experiencing a sporadic 'ring' on his phone, but I'm not sure it was after haning up a call.. Do you use CDR logging (to mysql?), if yes, do you find anything in there about this call? Maybe try searching through the asterisk logfile for records of this call? Cheers, Kristof. ___ 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] Junghanns 4 port BRI problem
Doug Reid - Stormcorp wrote: I have a Junghanns BRI 4 port installed where only the first channel of each line is working i.e. channels 1 and 4 work but 2 and 5 don't. Our config is the same on this box as 15 other similar installations where all works well. the only error I see is in /var/log/messages: Not sure if it's related, but I had the same error once when I used a wrong parameter in zapata.conf. For signalling I had something incorrect, but after i changed it to the following, it was okay: signalling = bri_cpe Cheers ___ 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] gxp-2000 tftp cfg
The VoIP Connection wrote: It's here: http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/gxp2000_config.txt Very interesting, it wasn't available at Grandstream's site :-) Thanks! I will adjust some things on the page now I have the new template.. http://voip-info.org/tiki-index.php?page=GXP-2000 Cheers ___ 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] voip-info.org unreliable lately?
Damon Estep wrote: I assume the bandwidth is being donated or something, but surely someone would be willing to donate reliable bandwidth as the knowledge hosted on the site (which is also donated!) is worth way more than the bandwidth. Sure it's the bandwidth? If the wiki is loaded, I see Server load on the bottom of the page, the numbers sometimes go as high as 80-100..? Not sure if it's a Linux (guess so? :p) but if that represents the system load.. 80 is a 'bit' high indeed. Cheers ___ 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] Grandstream 100 pricing question
Francesco Peeters wrote: I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? Prices I know are around 99 EUR, incl VAT. But if you ask me, depending on how many you need, you should take a look into the GXP-2000. (+- 125 EUR incl VAT) The difference in quality (and features) between these is big enough to justify the difference in price. Cheers.. ___ 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 7912G DST
Hi, a small question.. I'm using NTP to synch our phones with an ntp server, but it seems the Cisco 7912G (with SIP image) does not handle daylight savings time very well? Am I overlooking something or is this a known feature? I'm using GMT+1 and minutes are correct but it doesn't respect DST. SIP software seems to be: v1.02.00(040406A). Cheers, Kristof. ___ 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] cisco7940 upgrading problem
Betl Gzlkolu wrote: I have Cisco 7940 with version 3.2 sccp and want to upgrade it to sipI got the firmware P0S3-07-4-00When the phone tries to reach the tftp server gives conf. error and in status messages It says CFG File Not foundIs it because of version problem or something else? Does anybody have any idea? That is described on www.voip-info.org, search that site for cisco 7940, you will find some details about the upgade process. If it still doesn't work just let us know.. ___ 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] music on hold on R key not working.
Jason Williams wrote: The release is Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k 1.0.6 had a broken hold music you need 1.0.7. and then bristuff it or get the newest bristuff, it's updated to 0.2.0-rc8a at the moment.. ___ 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] 7960 multi-line configuration
Joseph wrote: Is call waiting turned on? What do you see if you telnet to the phone, and do a show config? Are the lines all set the same? Sorry to barge in like this, here it's working like you're telling. If I do a show config, I see call_waiting: 1 and all the line's are set to the same SIP login. Chgeers, Kristof. ___ 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: [SPAM] - Re: [Asterisk-Users] cisco7940 upgrading problem - Email found in subject
Betl Gzlkolu wrote: I found the necessary documents on www.voip-info.org and create XMLDefault.cnf.xml it starts downloading but now it says Protocol Application Invalid ...It tries periodically and gives that error... Do you have any idea what should I do now? Can you post your XMLDefault.cnf.xml ? What other files do you have in your tftp dir on the server? ___ 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] 7960 'multi-line' configuration
Corey S. McFadden wrote: I added some content to the Wiki on this feature. I don't think it's well documented anywhere. Please expand upon what I put in there if you have more details. http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx Looks very nice! I have it working on a 7960 like described in the Call waiting section. I expect 2x 7940 phones and will try and use it on these also. If needed (but it looks complete to me) I'll add info to the wiki. Cheers. ___ 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] 7960 multi-line configuration
Patrick M. Gray, Jr. wrote: In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should I don't think that is possible, at least not the way one thinks it would work. I have also done some reading on this, maybe this thread gives a solution: http://lists.digium.com/pipermail/asterisk-users/2004-March/039271.html But, I am also curious on how other people have solved this, especially with using AMP for example. Cheers.. ___ 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] quadbri bristuff ztcfg fail
Sander wrote: I compiled the bristuff drivers and then I do -- When doing lsmod I can see qozap is loaded with zaptel but no entry in /proc/zaptel/ Did the compiling go correct? What version of bristuff are you using? (latest? 0.2.0rc8a) What linux distro are you running and wich kernel? Your zaptel.conf looks okay, but you should have something in /proc/zaptel indeed.. Cheers. ___ 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] QuadBRI card on Suse 9.2 Unable to load qozap.ko
Massimo wrote: Hi, I successfully installed zaptel,libpri,asterisk and qozap in a Suse 9.2. I removed the old modules loaded as default by Suse. Now I'm triying to load qozap.ko but I receive this error: Did you do the install with bristuff-0.2.0-RC8a ? Works nice on debian, I guess it will be the same on Suse. Cheers, Kristof. ___ 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] No music on hold when transferring call
Bam wrote: MOH is working in that a defined extension works just fine: exten = 6000,1,Answer exten = 6000,2,MusicOnHold() I had some similar problems with asterisk v1.0.6, 1.0.7 solved this. (it had something to do with SIP and MOH) Cheers. ___ 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] Bristuff and Belgium
David Masure wrote: Does anyone has any experience with bristuff in Belgium... ??? Yes. 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 I have not received any problems of that kind 'yet', could you show some logging? What card are you using? Maybe let know what version of 'other' software you are using? and what distribution do you use? cheers. ___ 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] Fax and spandsp
Julian J. M. wrote: I want asterisk to receive incoming faxes (via rxfax application) and send them by mail. The problem is that, although the fax machine and the asterisk log report a succesful transfer, the tiff file is just I have not experienced this before, but I am using spandsp-0.0.2pre10, perhaps you could try this, to see if this matters? I will surely (in the near future) rebuild a box and try out the new spandsp (pre15) but maybe you can try downgrading as a test. Cheers Kristof ___ 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 POE
Paul Hales wrote: From our experience (with a HP 2626-PWR) it hasn't been an issue. PaulH Paul, Would you like to share wich phones you're using on this switch? Cheers, Kristof. ___ 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] Junghans QuadBRI and fax detection
Marc Storck wrote: does the Junghans QuadBRI Card and qozap module support Fax detection? You could have a look at the fax detection code in AMP, maybe that helps? But I think it should work. We're not using it, because we're using a fixed number for fax, so if that number gets a call, the (software-) fax always picks up. We didn't like the small delay you need for fax detection, and this way it always works perfect. cheers Kristof ___ 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] Junghans QuadBRI and fax detection
Eugenio De Vena wrote: I have QuadBRI and asterisk 1.0.6 bristuffed but fax reception works ugly. My faxes are missing many rasters and even sending does not work well. Can you tell me with version of asterisk , spandsp, app_sndfax etc you use to have a good result? I have used bristuff-0.2.0-RC7k (as on junghanns.net ). This downloads v1.0.6 of asterisk, zaptel and libpri. I'm not doing fax-detection, I'm using DID to dedicate 1 number to fax receiving. Os is Debian Sarge. Let me know if it works out now ;) ___ 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] AMP 1.10.007 problem on cdr_mysql_table.sql
Colin Anderson wrote: So, AMP 1.10.007 from SourceForge seems to have this problem, anyone upgrading won't run into this problem but a new install you will. Just wondering, did you download AMP-1.10.007a bugfix release ? I have installed it a few days ago and it went fine. (somewhere beginning this week I guess) Cheers. Kristof. ___ 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] Linux Asterisk
Asterisk Pbx wrote: sorts of good things about it. The question im asking my self is what linux distribution is best to use? Do you know what distribution they use for their asterisk training? As in most questions, the one that you're used to use. I myself use Debian sarge, but I'm used to this distro.. I guess the developers use different flavors also.. Cheers, Kristof. ___ 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 CDR Locations
Aaron Daniel wrote: Does anyone know of a way to have asterisk save multiple cdr records in different places (i.e. the same record in a database locally and in another database on another system, or database and csv, or some other strange combination)? Well, I've been using mysql replication for a while now, but just not for asterisk (yet). You could do that, but it all depends on what you want to accomplish. Would be a nice way to get a realtime copy on a remote server/location indeed.. You can find more about it on the mysql site, just search for replication. Kristof. ___ 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] Dial from a URL - Possible?
Julius Kidubuka wrote: Is it possible to initiate/receive calls from a url (that is without having to install and configure a PC soft phone) using asterisk? If yes, may I please get some sites, pointers, HOWTOs on how its done? I think you need asterisk call manager, that can initiate calls for you. (try looking for call manager at voip-info.org) cheers. ___ 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
