Re: [asterisk-users] Stable clock with 2.6 and without Digium hardware.
How about meassuring it directly? For starters, take a look at zttest.c . (Though it could use some slightly better accuracy). Not sure how accurate is zttest.c. Will run some test to see it's accuracy. ___ --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] Stable clock with 2.6 and without Digium hardware.
Which card, BTW? TDM400 analog ___ --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] Stable clock with 2.6 and without Digium hardware.
Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable 1000 Hz for some time, but in moments stops/misses interrupts/goes away from 1000 Hz 2.6 with ztdummy USB_UHCI - don't works, needs some tweaking. Somebody knows good patch for it? 2.6 with ztdynamic as primary clock sources - some issues with 2.6 (ztdynamic not ported well to 2.6?) with the mainstream versions, somehow patches solves it. 2.6 with kernel clock - needs kernel recompiling and work stable with switched off kernel Preemption. Long time tests in progress now. 2.4 with digium card - stable 1000 Hz 2.4 with ztdummy UHCI - stable 1000 Hz 2.4 with ztdynamic clock source - stable 1000 Hz/Depends on network conditions. ___ --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] Stable clock with 2.6 and without Digium hardware.
Zoa wrote: Can you tell us how you do the testing ? 3-4 different ways. All gives same results, so test are pretty valid. 1. Interrupt counting inside the PC. 2. TDMoE packet counting on the switch. 3. External TDMoE equipment connected thru extreme network swich. The card of the PC and the device only connected to the switch. The switch filters all packets except TDMoE to de device. Calibrated oscilloscope conected to the interupt leg of the network chip. All coalescing/etc disabling. 4. Diagnostic results from firmware of the device. 5. ToDo test - oscilloscope directly conected to pads inside the PC, but needs mechanical work for each platform/type. 6. ToDo test - some driver relays the clock to simple hardware card in the PC and oscilloscope connected to it. All 4 tests reports same clock difference/clock misses etc. Tested at 3-4 types of hardware - different chipsets/processors. Same results. The are sheduled tests to around 30 more platforms, but pretty sure that the results will be similar. P.S. The device is TDMoE FXS/FXO modular channel bank currently ending development and starting production. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Congrats, Europe!
Vahan Yerkanian wrote: http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ So we're are waiting the free g729 codec for Europe 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
[Asterisk-Users] Problems using zaphfc and wct1xxp together...
Asterisk 1.0.6 Bristuff 0.2.0-RC7k When i load wct1xxp modules, zaphfc stops working. Without the module - zaphfc is working great. Tried with and without florz. More information: with wct1xxp: no debung on pri at all pri show span 1 shows that span (hfc) is Down. Any ideas, or time to buy $30 PII 233 PC for the HFC card? :) ___ 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] IP FXS channel bank
el Flynn wrote: Hi there, I'm trying to get hold of an evaluation IP-enabled FXS channel bank. I'm not sure if it's more a channel bank, or should be called a multiport-ATA. Oh well. On the surface it looks quite nice - 16 FXS ports, and since it's connected to the network I can do away with an E1/T1 connection required of the normal channel banks (if it can be called that :) Here are some features I got from the brochure: 1. MGCP, H.323 (v4) and SIP support 2. Selectable, multiple codes (g711/g723/g729A) per channel 3. G.168/165-compliant adaptive echo cancellation 4. Echo canceller jitter buffer, VAD and CNG 5. complete voice band signalling support 6. provides inband/outband DTMF generation/detection 7. provides call progress tones 8. web management interface 9. LAN (10/100) and WAN RJ-45 ports It looks like a standart VoIP box, not like a channel bank. Of course it will be worse than a T1 card with CAC Channel bank. ___ 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] Caller ID Problems after upgrading from 1.0.1 to 1.0.4
I'm using T100P with CAC AB II, only FXS ports. After upgrading, asterisk stoped sending caller IDs to the phones. Even inside - port to port. I got 2 errors in the debug: __zt_exception: Exception 23, channle 2 (i'm ringing to channel 2) zt_handle_event: Didn't finish Caller-ID spill. Canceling. ___ 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] mcedit syntax for asterisk conf files
Does anyone cool mcedit syntax for the configuration files to share? :) ___ 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
[Asterisk-Users] Anyone playing with E1 channel bank?
I looking for some compatibility information about E1 channel banks working with *. Some conf files will be great too. Thanks ___ 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
[Asterisk-Users] New card - TE110P?
Will be there new card? I'm asking it, 'couse i'm going to buy 3-4 cards? Or i should wait for the new one? ___ 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
[Asterisk-Users] TDMoE Question?
Is there any definition or reference of the TDMoE protocol? Or it is just 24*64(for T1)+ethrned overhead bits frame each 1/1000 second? ___ 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
[Asterisk-Users] Some questions about channel banks signalling?
What signaling uses asterisk to comunicate with channel banks with the T1 or the E1 boards. Is there any differences between T1 and E1 signalling, or just the number of channels? ___ 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
[Asterisk-Users] 1.0 Libs
Whicch version of zaptel and Zapata should I use with 1.0? ___ 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
Re: [Asterisk-Users] GSM to BRI ISDN Gateway
Chris Lee wrote: Miroslav Nachev wrote: Hi, I am looking for GSM to BRI ISDN Gateway. Any help? I was also looking for such things nd came across these guys: http://www.2n.cz/export they have a product or two for GSM and here is the one I found most likely to work for me (two GSM sim cards providing two ISDN channels on a BRI line): http://www.2n.cz/uploads/2/PAGES/C379.HTML But I still have to get hold of one for testing, the local supplier is moving offices and as such can not help me out in the short term. Chris. I use this. Works. ___ 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
Re: [Asterisk-Users] GSM to BRI ISDN Gateway
Chris Lee wrote: Miroslav Nachev wrote: Hi, I am looking for GSM to BRI ISDN Gateway. Any help? I was also looking for such things nd came across these guys: http://www.2n.cz/export they have a product or two for GSM and here is the one I found most likely to work for me (two GSM sim cards providing two ISDN channels on a BRI line): http://www.2n.cz/uploads/2/PAGES/C379.HTML But I still have to get hold of one for testing, the local supplier is moving offices and as such can not help me out in the short term. Chris. More Pros: - Easy instalataion. - Siemens GSM modules. - Good antennas with long cables. Cons: - Old implementation of the logic board. - No terminal mode on the RS232 interface. - Uses proprietary configuration software that works only under windows. In my case i have to ask a friend with notebook to come in the office, when i need to change something. ___ 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
[Asterisk-Users] Strange I4L troubles
I have 2 cards, using HiSax type=15 driver. The problem is that when i place outgoing calls, all calls goes through the first card. All Modem/ttyI0, Modem/ttyI1, Modem/ttyI2, Modem/ttyI3 goes thru the first catd. Attached the modem.conf [interfaces] context=incomming driver=i4l type=i4l dialtype=tone mode=immediate msn=9868620 incomingmsn=11 device = /dev/ttyI0 device = /dev/ttyI1 incomingmsn=* device = /dev/ttyI2 device = /dev/ttyI3
[Asterisk-Users] New G.729 codec and VLANS
The readme says that the license uses all network cards MACS What happens when VLANS are added or removed? Is it safe? ___ 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
Re: [Asterisk-Users] New G.729 codec and VLANS
Nicholas Bachmann wrote: Anton Tinchev wrote: The readme says that the license uses all network cards MACS The MAC address is unique a 6 byte address assigned to every 802-family (802.1 Ethernet, 802.11 wireless, etc.) network interface. What happens when VLANS are added or removed? Nothing... VLANs have absolutely no effect of MAC addresses; a VLAN is just a virtual partition within a switch. In linux VLAN appears as completely different network interface Is it safe? Completely. Adding or removing NIC? Nick ___ 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
Re: [Asterisk-Users] E100P and T1 channel banks
Andrew Kohlsmith wrote: On Monday 12 July 2004 07:36, luan au wrote: Could you kind Asterians (should we pick Asteroids then?) confirm if I can use an E100P card with a T1 channel bank via * please? I live in the UK hence the question. Yes. You''l only get 24 channels but it shoudl work fine. And I prefer the term Astericians (think electrician), myself. -A. Any signaling and framing issues? ___ 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
Re: [Asterisk-Users] Audiocodes - Asterisk Implementation
Brian J. Rathman wrote: Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 20587: Found Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:11 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:13 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:18 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:20 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:24 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:25 DEBUG[1133742896]: chan_sip.c:706 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' I am using CVS version Asterisk CVS-HEAD-06/18/04-11:53:43. I have tried changing just about every config option I can think of in both Asterisk and the Audiocodes box without any success. Any ideas? I have checked the web for documentation on this setup, and all I have found is that some people have it working, but that is about it, no details. Any help would be greatly appreciated. Thanks, Brian Firmwire version? ___ 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
[Asterisk-Users] Access Bank 2 --- T100P T1 Cable.
I just got my Access Bank 2 (of course i'm happy now:)). Just need the wiring scheme Thanks ___ 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
Re: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6
Joe Baptista wrote: I'm installing the new Slackware 10.0 distribution - but not sure if i should go with the 2.4 kernal - which i think is the default install - or the new 2.6 kernal? anyone running * and slackware 10.0 with 2.6 kernal? thanks joe Asterisk stable CVS with slack 10/2.4.25 custom kernel - it works rock stable ___ 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
Re: [Asterisk-Users] AS5300 and Asterisk
Daniel Jimenez wrote: Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. Which model - 5300 or 5350. 5300 have different DSP blades for dial-up/in and VoIP We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco equipment before. ___ 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
Re: [Asterisk-Users] max asterisk load
shabanip wrote: I have a big case: - 5 x TE410P/TE405P quad T1 (try to not sharing IRQs on them) - 20 x TA750 CB (60 FXO and 420 FXS) - echo cancellation will be used if required. - up to 200 registered sip phone - using ulaw as default codec for all connections - voicemail, ACD, IVR and MOH will be used - conference will be rarely used. whats your recommended pc? a quad xeon or even a quad opteron with xxGB of memory? is it enough for this big job? Of course, don't use FXO. T/E1 is better. Use single CPU Machine. Quad Xeon costs 20 times than a P4/3Ghz/800Mhz FSB Machines. And it is only 2 to 2.5 times fatser. P4SCi from Supermicro is a good MB. Try to avoid using of IP (SIP) phones - aastra 390/480 with digium card outperforms anything. How much of these 420FXS will be active in the same moment - this is critical to designing the system. 420FXS = 18 Channel banks. Spread the system to 6 machines: Each machine - 3 channel banks + 1 T1/E1 for Incoming in Each machine. 1 Machine for SIP incoming and 1 for all additional stuff - Voicemail ... And give some more information - how much ACD queues will be, how much of the FXS phones will be connected to the each queue ... shabanip .G, Come on now, give us more. How many concurrent calls? What's your idea of a modern PC? Processor Speed, HardDrive space, etc? How many voicemail users? Nat'd SIP clients? What's your lan setup? Codec Translation? Voice Compression, Echo Cancellation..? Before you get information, you have to give information. People aren't going to pull your teeth to help you. The information required should pop out of your head into the e-mail. Personally, I'd reccommend reading up on PBX's before making posts like this. Not trying to start a flame, but come on, let's be serious here. - Brent On Tue, 17 Feb 2004, [iso-8859-1] Gustavo Garc�a Bernardo wrote: We would like to use asterisk as a SIP phone PBX with voicemail support only. The hardware could be a typical modern PC. Thank you very much. .G -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de mattf Enviado el: martes, 17 de febrero de 2004 16:48 Para: '[EMAIL PROTECTED]' Asunto: RE: [Asterisk-Users] max asterisk load That is extremely depandant upon what you want to do. First we have to know what the job of the asterisk server is, will it be an inbound ACD, a SIP phone PBX, a T1-only IVR, a conference call system, a voicemail system, and office phone system with H323 phones, etc.. Also, we need to know what hardware you were planning on using. The load is extremely depandant upon what you are doing with it. For example, a simple IVR/Zap-T1-channels-only system can handle 10 times the number of consecutive calls of a SIPZap conference call system(at least in my experience). Let us know what you plan on doing with it and we'll give you our best guesses as to the capacity. MATT--- -Original Message- From: Gustavo Garc�a Bernardo [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 17, 2004 9:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] max asterisk load Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G ___ 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 ___ 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 ___ 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 ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] 729 licence on scsi
Mark Spencer wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? If you'll just be patient for a little while, I'm working on new G.729 code which does NOT use the voiceage code and thus does NOT have their stupid SCSI problem. The new copy protection scheme will be based upon just the MAC address of your ethernet card, and WILL NOT DO ANYTHING WITH YOUR HARD DRIVE. In addition to eliminating the crappy copy protection code, the new version is approximately twice as fast as the VoiceAge code, which DOES NOT MEAN YOU WILL BE ABLE TO DOUBLE THE NUMBER OF CHANNELS PER BOX but does mean that you should be able to get substantially more channels per box. Anyway I'll post again on here when we're ready for beta testing, and anyone that has bought a license for the voiceage code will get to upgrade to the new code free of charge, of course. Mark Hmm, which code is used for the new h.729 Codec. And which license. Here, in my NOT FREE, Ex Communist country is completely legal to have GPL-ed g.729 code. ___ 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
[Asterisk-Users] Callwaiting callerid on 390s?
Anyone got callwaitingcallerid working succesfull on nortel/aastra/.../... 390 ADSI Phone? It will be great if someone share some ADSI Scripts for these phones also. Thanks ___ 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
Re: [Asterisk-Users] Largescale Asterisk setup - 1000 external lines
[EMAIL PROTECTED] wrote: Hi there Does anyone know if it is possible to install a largescale asterisk cluster with up to 1000 external lines. Redundancy and loadbalancing would surely be a must for a such system, but which other things should be considered? Best Regards Guenther Rust How much of this 1000 lines will talk simultaneously? You will need around 11-12 TE410P cards with 45-50 channel banks, and cumpers with mainboard based on Intel 750X or Broadcom GC-XX chipsets put how many computers depends on the max talk/idle lines ratio. ___ 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
Re: [Asterisk-Users] error with asterisk -vvvvc
vozip wrote: Hi Im a new user and I do test with my hardware. I have a x100p and telephone vozip. And when I run this command asterisk c for to test it. My computer show it warning [chan_iax.so] = (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]: chan_iax.c:4828 set_config: Unable to load config iax1.conf == Parsing '/etc/asterisk/iax.conf': Found == Using TOS bits 16 == Registered channel type 'IAX1' (Inter Asterisk eXchange Drver) == Registered channel type 'IAX' (Inter Asterisk eXchange Drver) == IAX Ready and Listening on 0.0.0.0 port 5036 [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' [chan_modem_bestdata.so] = (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver) [chan_agent.so] = (Agent Proxy Channel) == Registered channel type 'Agent' (Call Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' == Parsing '/etc/asterisk/agents.conf': Found [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2427 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Apr 2 07:45:12 WARNING[16384]: chan_iax2.c:6171 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Apr 2 07:45:12 WARNING[16384]: chan_iax2.c:5586 set_config: Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] = (OSS Console Channel Driver) Apr 2 07:45:12 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [skipping chan_alsa.so] [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 2 07:45:13 ERROR[16384]: chan_zap.c:7289 setup_zap: Signalling must be specified before any channels are. == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Apr 2 07:45:13 WARNING[16384]: loader.c:312 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Apr 2 07:45:13 WARNING[16384]: loader.c:407 load_modules: Loading module chan_zap.so failed! Check zaptel drivers loading Any ideas.? Cheers..! vozip ___ 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
Re: [Asterisk-Users] IAX2 International Termination
1. Aastra 390 to Digium tdm. 2. E1 line. Redirected DID number to USA. tried with several phones - simemens s45 gsm, panasonic, ge. Everything works fine, but dtnf relaying is broken. Stephen Karrington wrote: Thanks for the feedback. What kind of phone are you using? Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Thursday, March 25, 2004 3:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 International Termination Tested from Bulgaria. The quality is great, even that the ping from here is 170ms. Some troubles with dtmf sending. Stephen Karrington wrote: Hello Everyone, We are about to launch our International IAX2 worldwide termination service from any IAX2 softphone. We would like people to make FREE calls to the USA or Canada so we can check the stability of our platform. We are allowing everyone to call the USA for free RIGHT NOW! You can make calls to any land line phone or mobile phone in the USA and Canada! The string to dial is: [EMAIL PROTECTED]/01510111 This is an example of calling a USA based number. If you want to call a San Francisco number like 1-510-333- then the string to dial is: [EMAIL PROTECTED]/01510333 We are doing this to test a few things and would like your feedback on the following: 1. Call quality. 2. Server loading. We are wondering how many simultaneous calls we can get on this server before it hits too high a load and affects the call quality. Please send any feedback on the call quality and your experience to support AT diamondcard.us. This server is located in the East coast of the USA. All users who are within that vicinity or even on the West Coast should experience very good call quality. Callers from other parts of the world will experience lesser quality depending on their location and how good their internet connection is. We will be implementing servers in Central Europe shortly for European callers to use our service. There might be some downtiime if we have to reconfigure the server to handle issues that arise when people start calling. Thanks for your feedback and have fun making calls to the USA! Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ ___ 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
Re: [Asterisk-Users] IAX2 International Termination
Tested from Bulgaria. The quality is great, even that the ping from here is 170ms. Some troubles with dtmf sending. Stephen Karrington wrote: Hello Everyone, We are about to launch our International IAX2 worldwide termination service from any IAX2 softphone. We would like people to make FREE calls to the USA or Canada so we can check the stability of our platform. We are allowing everyone to call the USA for free RIGHT NOW! You can make calls to any land line phone or mobile phone in the USA and Canada! The string to dial is: [EMAIL PROTECTED]/01510111 This is an example of calling a USA based number. If you want to call a San Francisco number like 1-510-333- then the string to dial is: [EMAIL PROTECTED]/01510333 We are doing this to test a few things and would like your feedback on the following: 1. Call quality. 2. Server loading. We are wondering how many simultaneous calls we can get on this server before it hits too high a load and affects the call quality. Please send any feedback on the call quality and your experience to support AT diamondcard.us. This server is located in the East coast of the USA. All users who are within that vicinity or even on the West Coast should experience very good call quality. Callers from other parts of the world will experience lesser quality depending on their location and how good their internet connection is. We will be implementing servers in Central Europe shortly for European callers to use our service. There might be some downtiime if we have to reconfigure the server to handle issues that arise when people start calling. Thanks for your feedback and have fun making calls to the USA! Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ 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
Re: [Asterisk-Users] PCI front mount chassis?
Steven Critchfield wrote: On Fri, 2004-03-12 at 05:26, Rich Adamson wrote: I too am running 6 cards in my system, although not in a high traffic capacity load environment. So far my (limited) high-load simulations have shown no problems. So - is it apocryphal that the Digium cards (drivers) won't share interrupts? If there is a real issue with sharing interrupts then it seems to me to be a bug that needs fixing. PCI bus supports shared interrupts, why doesn't the hardware/driver? In most cases, sharing an interrupt is not a problem at all. There have been a few cases where _some_ issue was resolved by moving cards around, however the majority of those seem to be: a) abrupt system changes with no effort to seriously identify the root-cause, b) newbie installations where the condition of the underlying system infrastructure is totally unknown, or, c) wild recommendations that might have had some basis a long time ago but no longer apply. What I was referring to in this case about sharing an IRQ was the actual wire trace in the PCI bus. As I understand the PCI spec, there are 4 interrupt lines called A,B,C, and D. In slot 1, They appear in that order. In slot 2 they shift, in slot 3 they shift and again in slot 4. By the time you get to slot 5, all interrupts have been in each of the 4 spots and now they have the choice of staying in the same as slot 4 or shifting again. Most cards use interrupt line A as if they are in slot 1, and therefore if you move them from one slot to another, they would most likely get their own access to the interrupt line. This is part of the reason why moving a card around in the chassis helps. This is also why I would be cautious of trying to run more than 4 high interrupt cards on the PCI bus. My next point would be that of, if you need more than 4 cards, you need more than 1 computer controlling it. That is way too many eggs for one basket. Most of the 3.3V Boards have more then 1 pci bus. The boards that i use have 3 PCU Busses - 3xXPCI, 3xPCI66, Internal with Adaptec, ide and E1000 on board ___ 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
Re: [Asterisk-Users] SIP 3.0
I think that there are public ata186 upgrade server 213.137.73.159:8000 Sales Department wrote: Can anyone point me to where I might obtain the SIP 3.0 image for the ATA-186 Analog adapter. I'm willing to pay for it. I have a Cisco login but am apparently not authorized for this, just trying to get my fax working with asterisk and I need SIP 3.0. Any advise appreciate. Thanks Cory ___ 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 ___ 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
Re: [Asterisk-Users] Radius
Just make a wrapper. 100 lines in perl. Derek Samford wrote: I know this has been hashed, and rehashed, but I saw that a few people had said they were going to release their code soon. Is there a working implementation of RADIUS for Asterisk out there? Not looking to start a debate on how bad it is for billing purposes, that's a given, but I need it for legacy systems. Thanks, Derek ### This message has been scanned by F-Secure Anti-Virus ___ 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
Re: [Asterisk-Users] windows alternitives to Asterisk?
Putty http://www.chiark.greenend.org.uk/~sgtatham/putty/ hank smith wrote: is there a program that I can install on my linux box so I can configure the pbx from the internet from my windows box so I don't have to work with config files? thanks hank - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 08, 2004 11:07 AM Subject: Re: [Asterisk-Users] windows alternitives to Asterisk? hank smith wrote: hello I am just curious if there is any windows alternitives to Asterisk? can I also use them with free world dialup? thanks hank No, but maybe you could port Asterisk to Windows. No, that's not a joke. The Zaptel drivers might be tough, but Asterisk's VoIP features would probably run under Cygwin without too high a mountain of work. Regards, Steve ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] E1 Red Alarm
Check the crossover cable. Nicholas Bachmann wrote: Howdy - I'm trying to get a Malaysian PRI E1 up on a TE410P, with no luck. Right now, the setup is Telco - HDSL - WorldDSL UTU801- 2 BNC E1 - balun - crossover - TE410P Right now, the CSU/DSU-ish WorldDSL box has a green light indicating E1 sync, but the TE410P shows a red alarm. I checked the card by plugging the crossover from port 1 to port 2 on the 410 (it worked fine). It I change any of the cabling (i.e. swap things around), the green light goes off. I have my suspicions about the balun (http://www.ctcu.com/catalog/datacom/balun.pdf). Would a DB15F-RJ45 converter be better the the BNC-balun-RJ45 arrangement we have now? Here's my zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 The telco line IS working; it was tested and put in a couple of days ago. Any ideas why this isn't working? Nick ___ 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 ___ 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
[Asterisk-Users] Some ADSI script programming guide?
Hi all. Just got 10 aastra 390s and searching for some page ort resources with ADSI programing guide/examples. P.S. These phones rocks :) ___ 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
Re: [Asterisk-Users] Cisco AS5350 Gateway Intergration
SIP with G.711 for local lines or SIP with gsmfr for long disctance (slow connection). I used AS5350 several months with *, then i recieved my E100Ps and moved the Cisco to History (where all proprietary solutions must go). P.S. The Cisco is for selling for around 11K Euro (New, used in rack 4 months, 4E1s, 120DSPs) Eric Merkel wrote: I am beginning a project to integrate * with a Cisco AS5350 gateway for inbound/outbound termination. If anyone has experience with this, what channel type would you recommend? H.323, SIP or MGCP? I've scoured the archives to see what channel type would be the most stable but haven't found a definitive answer. Also, if anyone has dealt with this setup before and would be willing to share an example config (cisco *), that would be much appreciated! Thanks. ___ 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
Re: [Asterisk-Users] How to best debug SIP registration failure
ngrep. There is some patch for better displaying from iptel, that works grat George Pajari wrote: I am having trouble getting SIP phones to register with Asterisk. I know that the phone can register with FWD and I have used tcpdump to see the registration packets arrive at the Asterisk server, but nothing goes back. How should I attack the problem? What debugging tools exist to try to determine why Asterisk is not accepting the registration requests? I have googled for answers but no one seems to have posted a general approach to attacking such problem, or, if they have, I missed it. g. ___ 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
Re: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system
hmm, this pages must be fixed. Looks terrible on all NGlayout based browsers Philipp von Klitzing wrote: Hi there, please comment and adjust or enhance as you find appropriate: http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning Typical questions asked on the mailing asterisk-users are: How fast/big must my machine be in order to serve my needs? How many simultaneous calls can Asterisk handle? Unfortunately there are no simple answers. You'll need work through the following checklist to at least get nearer to an answer or be able to post a meaningful question to asterisk-users: [...] Cheers, Philipp ___ 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
Re: [Asterisk-Users] X100P
Soragan wrote: Ohh.. You better give that to me then. I'll send you my Pentium 133 w/ 16 megs of ram. It works great with the X100P. LOL, can your Pentium do web server, mail server with spam and virus checking and ADSL router all together? If it can do without any performance loses compare with mine, I'd be happy to change it. ;p Yes :) thttpd, qmail(spam included), courier pop3, virus protection = just grep for outlook in mail header (easy qmail script) and then - /dev/null, router - no problem Just get some moer memory for this p133 - around 64 would be nice :) ___ 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
Re: [Asterisk-Users] Problems with ATA's locking up..
Billy Huddleston wrote: Anyone had any problems with ATA's running 3.0 software locking up? Thanks, Billy Shht, can someone send me 3.0.0 version of software ___ 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
[Asterisk-Users] Need Europian vendor for Digium hardware.
Must accepts wire transfers and ships to Sofia. Thanks ___ 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
[Asterisk-Users] Is there any plans for Digium ISDN BRI card?
Yes, i know that there are many ISDN card on the market. But when i spend money for ISDN card, i prefer to be Digiums, to get all support and help Asterisk :). ___ 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
Re: [Asterisk-Users] B-channels restart problem
Ali Mughrabi wrote: Hi , I'm having a problem that really bothers me , I have looked for similar cases but couldn't really find an answer . I keep getting messages which says that -- B-channel xx successfully restarted on span x and this causes the calls to be disconnected if somone is already using the channel that is being restarted, plz notice below the under lined text in which I was testing and got on channel 20 span 3 , and then it got disconnected after restart, I can't decide if this is a telco or configuration problem , telco says they have no problem , shall I beleive them? please I need any help or comment that might be helpful. Thanx in Advance pleaes reply here or to at [EMAIL PROTECTED] Thanx in Advance Ali Mughrabi --_ Accepting call from '065639815' to '9009170' on channel 20, span 3 _ -- Executing AGI(Zap/82-1, ../album_show/album_show.agi|--apelant=065639815) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/../album_show/album_show.agi -- B-channel 14 successfully restarted on span 3 -- B-channel 15 successfully restarted on span 3 == Spawn extension (inbound, 9009170, 2) exited non-zero on 'Zap/82-1' -- Hungup 'Zap/82-1' -- B-channel 17 successfully restarted on span 3 -- B-channel 18 successfully restarted on span 3 _-- B-channel 20 restarted on span 3_ -- B-channel 19 successfully restarted on span 3 -- B-channel 20 successfully restarted on span 3 -- B-channel 21 successfully restarted on span 3 I have similar messages, but everything works ok. No disconnects. ___ 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
Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)
Steve Underwood wrote: Anton Tinchev wrote: Richard Grinnell wrote: Dell - PowerEdge 400SC Server Under $300 with MIR Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz FSB For those of you who aren`t familiar with the 400SC, this server is an Intel i875P chipset based server with an 8x AGP slot. It is compatible with 533Mhz and 800Mhz processors (hyperthreaded too), there`s built in 2 channel S-ATA, there are 6 USB 2.0 ports, there`s built in 10/100/1000 ethernet, 4 PCI slots (with both 5.0v and 3.3v universal support), ADI 198x audio, and loads of other features. These puppies are incredibly fast and rock stable. And this deal comes with 1 year parts and Onsite labor service too! Go to gotapex.com and Find on page: 400SC Richard Grinnell Yes, it has 3.3v PCI slots, but no 64Bit.. Actually it has neither 3.3V slots or 64 bits ones. The Dell site says it has 3 x 5V 32 bit slots. I've still yet to see a board with only 32 bit slots that is 3.3V Regards, Steve http://www.msicomputer.com/product/detail_spec/product_detail.asp?model=875P_NEO-FIS2R http://www.msicomputer.com/product/detail_spec/product_detail.asp?model=875P_NEO-LSR ___ 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
Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)
Peter Brown wrote: At 11:20 11/01/04 +0800, you wrote: Anton Tinchev wrote: Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks Unless one has appeared in the last couple of weeks, there are none. In fact, the only one I know of for any kind of non-Xeon Pentium is the Dell 600SC. That one isn't an 800MHz bus machine. Regards, Steve ___ 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 Asus NRL-LS533 Server Board Single Intel Pentium 4 processors up to 2.8GHz+ North Bridge: ServerWorks CMIC-SL South Bridge: ServerWorks CSB6 Up to 4GB registered PC1600 ECC DDR RAM, 4 DIMM slots Single channel Ultra160 SCSI 1 x BroadCom 5702C 32bit PCI Gigabit Ethernet controller for NRL-LS533 5 x 64bit/33MHz/3.3V PCI slots ASUS Server Management Software included TrendMicroTM ServerProtect anti-virus software full users version for enterprises FROM TWINCUBE.COM Peter Brown ___ 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 Yes, but they all are with 533 Mhz FSB ___ 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
[Asterisk-Users] My first E1 card is running :)
Just happy. hardware information: -- Some small factor IBM Celetron (coppermine) at 1100 (11*100FSB) 256 RAM 15GB Hard. 1 x Digium E100P - E1 Line from telco with 300 Dids 1 x TDM400P for local phones --- Few small machines (mainly brand PII at 233Mhz with TDM400P Cards. --- There is a lot of SIP equipment attached: 2 x Micronet SIP Gateways 1 x ata186 1 x AudioCodes 1004 2 x Cisco AS5350 Gateways ( now it seems Obsolete :) ) Not a sign of echo problem - is this becouse all my analog phones are connected with cat5e cables? This is heavy production enviroment - Sofia's Metropolian Area Network Operation centre. Now i'm playing with the ADSI scripts. If someone has cool ADSI scripts, please send me. P.S. Should we arrange different Successfull Stories Mailing list? ___ 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
[Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)
Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks ___ 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
[Asterisk-Users] Some bugs www.voip-info.org html
It is a great site hoever. There is some bugs that makes waching the site with mozilla damn hard. If you wonna, i may send yoy screenshots. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some bugs www.voip-info.org html
Registering and changing the theme I can guess... all context moved to the left ? and squeezed ? reloading the page seems to fix the wrong displaying. matteo. Il gio, 2004-01-08 alle 20:02, Anton Tinchev ha scritto: It is a great site hoever. There is some bugs that makes waching the site with mozilla damn hard. If you wonna, i may send yoy screenshots. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing list growth
It is small. I reading it justr for prelude to the kernel mailing list :) Scott Stingel wrote: Anyone know how many people now subscribe to the asterisk-users mail list? Yes, the number of new posts is getting overwhelming! This morning at 8am California time I had something like 75 new posts, and just cleaned it out the evening before! I will have to start getting up earlier I guess... Regards Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Email: scott at evtmedia.com Web:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Thursday, January 08, 2004 10:25 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Mailing list growth Well, mailing list growth is not only a good thing. It's getting almost impossible to handle. As I've stated before, we need to change Asterisk.org so we can help people in a better way and avoid a lot of the repeating questions on the mailing list. There's a lot of people unsubscribing, just because of the amount of messages. Asterisk.org needs an FAQ, more documentation on line and ... I've offered to help in this, with no reply so far. I think it's getting urgent. And this is the end of yet another message from a dark and snowy Sweden ;-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser and Arterisk works together ?
Yes. P.S. Someone shoult set this sticky :) Jorge R. Constenla wrote: Hi, Anybody knows if Asterisk work fine with ser ? We are using SER (iptel) for VoIP and we want to use Asterisk for PSTN termination for inbound and outbound calls. Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Gateway Integration
Only SIP. In my oppinion, h.323 is obsolete. Bruce Hedreen wrote: Did you use the h323 module on asterisk? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Tuesday, December 16, 2003 12:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco Gateway Integration Bruce Hedreen wrote: Has anyone succesfully integrated * with a cisco voice gateway ? Works well with AS5350 and ATA186. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK X SER
listas iPfone wrote: Hi All I´m trying to use asterisk and ser in the same box. When i start ser my phones don´t connect with asterisk anymore. i have two nics in this machine 192.168.0.31/37 I need to set asterisk and ser to listen in diferente adresses or ports? I can use the two softwares at the same time? how? I have many problems with my nat and asterisk and i think ser can help, i read the wiki documentation on it but i´m confuse on how to do it. Somebody can point me in the rigth direction? Thanks! Miklos Just set the SIP channel of asterisk listen on different port. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Gateway Integration
Bruce Hedreen wrote: Has anyone succesfully integrated * with a cisco voice gateway ? Works well with AS5350 and ATA186. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which ADSI phones to buy?
I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
rnc Info Lists wrote: Its a free world and everyone is entitled to their opinion. Here's mine on this topic. The cards aren't so expensive (99.95 USD). If they have their own hardware then they don't have to depend on the target system having a particular configuration. Example: right now I am running * on a system that has NO USB ports so couldn't use USB for timing. That makes their programming much easier. As has already been pointed out, nothing keeps someone from writing patches to use some other timing device. Digium is nice enough to put Asterisk as public software. Lets don't screw that up. Be part of the solution, don't complain about the problem. If you have a solution suggestion then post it.. probably others would be happy to help you Robert It can be some kind of contribution ($10 card + $40 contribution) :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
Adam Hart wrote: From: Anton Tinchev [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 15, 2003 1:06 PM Subject: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing... I'm first to buy 5 pack. Even for $30. Doesn't ztdummy already do this? Only if you has the right usb chip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing CallerID
JanM wrote:
Hello,
Does anyone know how to set the outgoing CallerID properly when using
Snom200/SIP/CAPI/BRI?
Following doesn´t work:
exten = _0.,1,SetCallerID,526910
exten = _0.,2,Dial,CAPI/526980:${EXTEN:1}
Asterisk writes:
*CLI -- Executing SetCallerID(SIP/226-ada0, 526910) in new stack
-- Executing Dial(SIP/226-ada0, CAPI/526980:0408665390) in new stack
-- Called 526980:0408665390
-- CAPI[contr3/526980]/0 is ringing
My mobile is only showing some other number that my isdn line is having.
---JanM---
Telecom restrictions?
I can set only caller IDs within the set of numbers provided me from telecom.
Check with your telecom if you're allowed to set any caller ID.
___
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[Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
I'm first to buy 5 pack. Even for $30. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this Hardaware Enough for Asterisk ?
Tarun Banka wrote: Hello, We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you. 1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa 2. Wildcard T100P interface card, that will connect Asterisk server to Our Nortel Switch SL-100 3. Wildcard TDM400P that gives us 4FXS ports for 4 Analog Phones 4. Server 1.8GHz or more P 4 1GB RAM 5. T1 Cable. Please let me know if I am missing anything. Best regards, Tarun I'm using similar setup to an old IBM PII 266/128RAM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PrePaid Application!!!!!
Bartosz Jozwiak wrote: Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid application with Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart Of course. This is the meaning of releasing :) For example i have strong knowledge + expirience writing application that using PgSQL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - H323 GAteway
Mireia Munoz de jesus wrote: Hi! I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register a SIP number in a H.323 gatekeeper? I know that with NetMeeting I can make calls peer-to-peer dialing an IP... but if the softtelephone of the other terminal is an SIP UA that is not going to work, is it? Please any help will be welcome. Regards, Mireia What gatekeeper do you use. It seems that is programed to make outgoing calls only to registered h.323 users. Just program it to forward unknown number to the asteris (or switch everything to SIP) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream wallmount??
Dave Weis wrote: Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. knife, rasp, glue(strong), wrecks from plastic lighter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is the pingtime option in iax chan(iax.conf)?
Sorry for asking for it, but it is nowhere documented. There is no maches in the mailing list or the whole google. I found it just in sources - conf parser of chan_iax.c. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with GPL license of Asterisk
costas wrote: I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application) that uses Asterisk data is that also GPLd? I understand if I add code to Asterisk, but what about external interfaces? Where is the seperation here of the Cathedral and the Bazaar? Thanks http://www.gnu.org/licenses/gpl-faq.html Here are the ansfers of much questions like this ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about codecs and interoperability with cisco AS5350
Hi all. I'm going to implement some large Asterisk based solution. Maybe 4-5 PCs with 1-2 E1/T1 trunks on each. Because some of the traffic will be sended to external VoIP provider, i has some questions 1. Which is the lowest bandwidth consuming codec in Asterisk, which is compatible with Cisco Gateways. Stability is needed too. 2. Have someone allready bulded such a systems and what hardware (pc i mean) is needed Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH other than mp3 ??
For fast hacking - Mplayer plays ASFs nice. Try to make some wrapper(in perl maybe), that makes mplayer looks just like mpg123 for asterisk :) John Brown wrote: On Sat, Sep 06, 2003 at 05:40:28PM -0500, Brian West wrote: Just tell em its ASF.. like the would know the diffrence. The system they use to interface with the web is a pre-made system for the station and we can't touch it. Output is ASF and we can't make it something else. bkw On Sat, 6 Sep 2003, John Brown wrote: On Sat, Sep 06, 2003 at 04:56:22PM -0500, Brian West wrote: You didn't just say MS-ASF Yup I did, ducking under the table... Customer requirement MP3's good.. They rock, but the customer is doing something different and wants to insert ASF formated tunes. bkw On Sat, 6 Sep 2003, John Brown wrote: is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone - Asterisk - SIP LD Provider question
Peter Pauly wrote: If Asterisk registers with a SIP long distance provider and I make a call from an IP phone through Asterisk to that LD provider, does the RTP (audio) traffic flow between the two end points directly (normally the IP phone and the LD provider) or does it flow through Asterisk? I'm asking because I have Asterisk running behind a NAT firewall along with an IP Phone (software) and I'm trying to get it working with Iconnecthere (ICH). I am able to register, connect , but no audio. I have ports opened up on the firewall, but they point to the Asterisk machine and not the IP phone machine. In this scenario, any audio traffic would have to go through the asterisk box to reach the IP phone. Is that how it works? SIP control connection usualy goes thru firewall. RTP - no. Just put the Asterisk on the machine with the firewall and it will work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser vs Asterisk?
Rich Adamson wrote: Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? I'd like to implement one or the other handle a small number of local ip phones, tie a couple of asterisk (or ser) machines together across the Internet, implement a couple of FX gateways (to handle incoming pstn calls, and for outgoing pstn calls), and use features mostly common to pbx's. No immediate need for CDR. Voice mail, callerid, etc, are wanted. Would like to accept incoming sip calls from anyone on the Internet that might choose to call. Would Ser or Asterisk be the most appropriate choice? Rich I using both in heavy production enviroment. SER is the BEST SIP proxy that i found. But it is just sip proxy. And can serve a _LOT_ connections (10,000 users, 20 cals per second). Asterisk is more like telephone switch with lot of features, but far slower. In your scenario - Asterisk. SIP cannot act as a PSTN Gateway (PC with some telephone board). Mixed scenario - Voice mail, PSTN GWs, Conference ... - Asterisk. Call routing - SIP. You can implement SER only scenario, if you use Hardware Gateways - Cisco AS5350. But i don't recommend you to use HW Gateways - main problem is that this gateways still don't support the speex codec, so if you make long distance calls between, let say, AS5350 and Asterisk you can't use low bandwidth codec. . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Some questions about Asterisk and reliability
Gabe Bourque wrote: Hello Anton Tinchev, I'm writing to you in hopes you can answer a few questions regarding Asterisk/Digium and it's reliability. I saw your posting in the Asterisk mailing list (Re: [Asterisk-Users] Is Asterisk ready for real use?) and decided to write directly to you. The reason being that you are one of only a few people who have mentioned that they actually have Asterisk running in a production environment and have for some time. I There are MUCH PPLS running asterisk on production enviroment. More than you can imagine. hope you don't mind me emailing you directly. If so, I apologize. You're welcome. I guess my biggest question is simply the reliability of Asterisk and Digium cards as a whole. At the moment we are looking at implementing a production system using a quad fxs card and four fxo cards (more or less). I'm aware of some of the issues with the fxs cards and the fact that a revised version of the card is due out sometime soon. Have you had issues with these cards or the fxo cards? How has your experience been with the hardware overall and its reliability? For example demand on the pc, etc. Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards are cheaper and has more features. Any ISDN card that is supperted by isdn4linux must work, but I recommend you Sedlbauer chipset based. Digium FXS cards are great. I have looked at Asterisk for a little while now and it's reliability isn't as much a concern as the hardware's but how have you found it's reliability, performance, and management? Management is great. Reability is great too. The is some issues only with h.323, proprietary codecs, but if you don't use h.323 and/or g.729 codec everything is great. Performance? It depends of what are you doing. But sub $200 ebay machine (brand pentoium 3 1G 256 Ram) is enough. I running it under IBM 266 Pentium II/48MB with 4 ISDN channels (2 cards), 1 LineJack, some incomming SIP calls from Cisco AS5350. For FXS(office phones) I use 1 Ata 186, 2 Audicodes FXO gateway (4 ports) and several PCs with gnophone and headsets. No problems at all. I'm also interested in some of the overall issues you've faced in running Asterisk in a call center environment. Only configuring a kernel to get ztdummy.o module running (for MoH and conference). Call queues works great. Only you need little hacking to get information from call queues to the agents. (Agents a brainless peaples by default and it is hard to understand anything from text terminal :)). If you wonna make bigger call center, just get T1 Trunk. Thank you for your time. It is greatly appreciated. - Gabe Bourque [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Some questions about Asterisk and reliability
Eric Wieling wrote: On Thu, 2003-08-21 at 13:46, Anton Tinchev wrote: Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards are cheaper and has more features. Any ISDN card that is supperted by isdn4linux must work, but I recommend you Sedlbauer chipset based. Digium FXS cards are great. Price of BRI .vs. PRI .vs. Analog .vs. Channelized T-1 vary DRASTICALLY depending on your location and phone company. Here Analog = BRI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which linux soft phone is best with asterisk.
I must put working 4 sales agents. They will have PCs on the workplaces, so I thing that some Linux software phone with headset is better solution ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Some questions about Asterisk and reliability
Dave Cotton wrote: On Thu, 2003-08-21 at 20:38, Eric Wieling wrote: BRI (more correctly called ISDN BRI) is a digital service. That may be a technical answer. On Thu, 2003-08-21 at 14:12, Anton Tinchev wrote: Here Analog = BRI I mean the Price of course. That could be a financial answer? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk ready for real use?
Mike Ciholas wrote: Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. It is *not* an option to purchase a VoIP system package from Cisco, 3com, etc. Installers are getting an enormous premium for this now (rough estimate, 20 extensions $40K (!)). I am this close to committing to a solution based on Asterisk PBX, PoE LAN switches, and VoIP phones. I am absolutely sure it is the right *long term* solution, but I don't know if it is ready for reliable daily usage. I've literally read the last year's worth of posts to asterisk-users to get a feel for the situation. Since you don't see posts of the form installed it, just working, no problems very often, you could get the opinion that everyone has problems since that is what the mailing list is for. So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. For those of you who have done it, how much time did it take you to get the system running smoothly? PS: In case it matters, we're extremely Linux capable (we use it for our file serving, networking, and we built our own custom ERP on perl and mySQL, we also do embedded Linux in custom military robot controllers). For me it is ready for heavy use. I allready using it for 1 call center (call queues ...) and for the offise PBX. Now i'm waiting some hardware (channel banks ) to test it with 100+ lines (1 E1 Trunks and analog lines). Only thing that is, if you're are begginer asterisk user, you will need some more time to get whole picture and the features (3-4 days googling and reading mailing list archives) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Someone used ADIT 600 Channel Bank.
I must buy channel banks for ~120 lines. After some googling and ebay searching i see that ADIT 600 has exelant proce//... for me. Just wandering how it works with asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audiocodes fxs
Kelvin Chua wrote: hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin Can someone send me SIP firmwire for audiocodes 104. I has h.323 only and it sucks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Callout problem
-- Called ttyI0:09854433 -- Modem[i4l]/ttyI0 is busy == Everyone is busy at this time Everything works fine in minicom - i can call. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for Asterisk
Kim C. Callis wrote: I was thinking of adding QoS to my Linux based router. I thought I would add all my IP phones and my * box into a VLAN, and then would do a QoS setup for that particular VLAN. Has anyone did any QoS setups for better performance? Has it made any change to the performance? Kim C. Callis http://www.lartc.org/ - complete howto http://www.docum.org/ - Very usefull examples ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
Is the problem same if I wonna to have dynamic mail boxes? Steven J. Sobol wrote: Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be web-based. I'm looking at DynExtenDB (and have played with it). I love that it reads the dialplans out of a MySQL database - that is a critical issue for me. But it has some issues. I have a test dialplan with one call to Playback() - just plays a wav file then exits. When DynExtenDB() is called, it adds one extra step that calls DynExtenDB_Free()... --If I let the wav file play to the end, DynExtenDB_Free() is called properly. If I hang up prematurely, it isn't, and it also isn't called if I set the dialplan to dial out (for example, to forward the call to my cell phone). --If DynExtenDB_Free() *is* called properly, and I then make another call, DynExtenDB() doesn't seem to be called again. --I'm not sure that setting up a dialplan for extension 'h' is a good idea. What if I call, and then someone else calls and I hang up in the middle of the call? I am ready and willing to make changes to the source to DynExtenDB. In fact, I'd like to get it to a point where it could be used in a production environment. But I have a lot of questions before I can do that. BTW, I have looked in the archives, and it's been suggested that maybe AGI is a better way to handle this sort of thing - but wouldn't the same issues still exist?? Thanks SJS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best E1 channel bank?
I bought second hand E400P for around $450. Jeremy McNamara wrote: Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run your E-1 connection to the telco and run T-1 channel banks on the other spans. Jeremy McNamara Anton Tinchev wrote: Need to buy 2-3 channel banks for some asterisk deployments... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Is it stable enought? I mean around 30-40 Incoming SIP connections. Or i must trash the cisco and put Asteriisk/Digium/speex box? Jeremy McNamara wrote: You have to run a console with the G.729 due to the voice age library lameness. We run safe_asterisk with a TTY and it seems to be fine. Jeremy McNamara [EMAIL PROTECTED] wrote: Try launching asterisk like this: screen -d -m asterisk -vvvcn Aparently there is some bug in the codec. - Justin On Sun, 20 Jul 2003, Anton Tinchev wrote: Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco interoperability?
Hi. Does someone tried this scenario? (or like this) | Asterisk with| - -- ---| H.323 and G.729 |--| Gatekeeper(GNUGK) || Cisco AS5350/AS5400|--- E1/T1 line | Registered in GK | - --E1/T1 I know that it should work, but there is a bunch of possible showstopers like codecs interoperability, . I just wonna avoid buyng another AS5350 Gateway - is always better to use something opensource ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music while waiting for agent to free.
I has a E1 trunk to PC and 4-5 SIP phones. Can * plays some music in all calls if all the phones are busy, and when one got free, to forward the call to the agent. Excuse me, if it is newbie question, but i'm googling and reading this list 4 hours and didn't found clear answer :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
