Re: [Asterisk-Users] * For Call Center
On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote: sounds like one of those pesky auto dialers the simpsons make fun of. It sure does... -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ 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
AW: [Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP
Hi, [...] There should also be a digitally signed version of that file (cmterm_7920.*.sbn), which the phone probably requires. nope. no sbn. according to my cisco source the file is not signed. Funny, that would be the first phone with unsigned firmware. But I'll double-check after the next firmware update. At least for my other phones, Cisco introduced signed binaries for versions = 5.0; looks like the 7920 firmware is still below that. Bye, Martin ___ 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] grandstream asterisk configuration
On Thursday 15 January 2004 01:30 am, Chandra wrote: hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT. Why leave a host to defend for itself? At least behind a firewall you got some layers of protection. -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ 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] Asterisk on FreeBSD 4.9?
Jason T. Nelson wrote: I have already started playing with trying to figure out why Asterisk runs so badly under FreeBSD, such as eating 100% of the CPU without warning unload pbx-wilcalu.so, see http://www.voip-info.org/wiki-Asterisk+freebsd /O ___ 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] Sending voicemail with qmail
Dear all, Is * capable to use qmail as a MTA? If so, how can I set it? Thanks Isianto ___ 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 bind RTP when IP alias are configured
Hi Robert, Yes, this fixed the RTP issue for me. Do you need a bug note created on this ??? Cheers SW Date: Mon, 12 Jan 2004 14:03:14 -0800 (PST) Subject: Re: [Asterisk-Users] How to bind RTP when IP alias are configured From: Robert Hajime Lanning [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] quote who=SW Hi Folks, I have a situation where my Colo insists on a particular IP setup for my * server box. They allocate two blocks of IPs to my colo server. One set as my own (ex 20.20.20.20.4/30 - 4 ips) and the other as a transit lan (es 10.10.10.0/29). These are all public IP addresses and there is no NAT involved in. So essentially I have to set-up IP aliases in my Linux box as follows; Example: TRANSIT LAN: 10.10.10.0/29 CUSTOMER LAN: 20.20.20.20.4/30 RedHat LINUX FILE: /etc/sysconfig/network-scripts/ifcfg-eth0 DEVICE=eth0 IPADDR=20.20.20.20.4 NETMASK=255.255.255.255 ONBOOT=yes FILE: /etc/sysconfig/network-scripts/ifcfg-eth0:99 ## TRANSIT IP: DO NOT UNCONFIGURE ## DEVICE=eth0:99 IPADDR=10.10.10.4 NETMASK=255.255.255.248 NETWORK=10.10.10.0 BROADCAST=10.10.10.3 GATEWAY=10.10.10.1 ONBOOT=0 First of all. I can ping to customer lan and telnet to it, therefore IP routing (at least for unicast traffic) works fine. Now question arises when asterisk start to work on this box. Since the IP that I am supposed to use is 20.20.20.4, I set that as bindaddress in my sip.conf file. As far as SIP messages are concern * users that IP address, no problem. However for RTP stream * users 10.10.10.4 as it's source address. Because of this obviously calls will not go through asterisk, as the ip phone is expecting RTP packets from the SIP server which is bound to IP 20.20.20.4. Is there a way to tell * to use the same bind address in SIP.conf (h323.conf, iax.conf) for RTP ? I read rtp.conf file but that does not show any bind address. It seems like LINUX always select it's src address as the interface (alias) which has the gateway tied to it unless otherwise an application specifically asks Linux to use a particular ip address. rtp.c uses 0.0.0.0 (hardcoded, well kindof, the whole struct is initialized to zeros, so, it is just not set) Since, to get around NAT issues, I have a host route on my firewall (Linux IPTables), I have the same problem. In rtp.c - function ast_rtp_new()... rtp-us.sin_family = AF_INET; /* the next line was added to fix host route hack instead of NATing */ inet_aton(20.20.20.4,rtp-us.sin_addr); rtp-s = socket(AF_INET, SOCK_DGRAM, 0); Sorry, no context diffs. When I get around to adding an rtp.conf keyword, I will provide context diffs to bugtrack. -- END OF LINE -MCP ___ 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] SNOM IAX image
Christian, There are a couple of images at http://snom.com/download/share. We are not really happy with the latest image yet; hopefully we can fix the remaining issues in a couple of days. Input appreciated (but no new feature requests until we have this stuff stable!). I could not find any image with IAX in the name. So just one direct question: Do you have an IAX image for any of your hardphones? TIA rgds pos ___ 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] SIP clipping sound
Hello all, Is there a way of setting the sound level at which * starts to transmit silence. It appears when an external call comes in the caller speaks silently you hear a lot of lost bits as it drops in and out. This only seems to have been introduced when I upgraded to the latest version of *. Regards Nick
[Asterisk-Users] GSM connection for asterisk
Hi Has anyone managed to connect a GSM modem to Asterisk ? I have a Siemens M20 Terminal that can do voice/fax/data and want to connect it. Are there someone on the list that has experience with the Siemens M20 and Asterisk? Terje _ Hotmail snakker ditt språk! http://www.hotmail.msn.com/cgi-bin/sbox?rru=dasp/lang.asp - Få Hotmail på norsk i dag ___ 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] newbie ISDN question
Hi All, Have just checked kapejod's quadBRI specs and looks wonderful. I am not an expert on ISDN either, but seems to me that features and functionally worth the 600 EUR (almost US$600,right??) suggested price. However, from * stand point it seems --pls pardon me in advance if I am wrong, that kapejod's quadBRI card provides much more 'horse power' than * really needs in standard applications In our case (my partners and I), we are looking for an internal plain multiBRI EuroISDN card capable of taking the B channels and deliver them to the * in a duplex mode, so that * take care of everything else: Caller ID, Call Routing (switching), Conferencing, Least Cost Routing, Protocol Conversions, etc, etc .Our ideal multiBRi card needs to operate in TE mode only but would be great if it was capable to accept either U or S/T BRI interfaces --with S/T required only in situations where a NT1 + 2POTS box should go before * to provide dial tone even during shutdowns. Given that our * servers -- P4 Dell pe400SC, costs about US $300 here, it would be great if this ideal multiBRI would cost no more than the server, which has (we guess) plenty power to run the BRI's card load and a 4x12, 6x18 or 8x24 small office application. Does any one uses or knows any BRI card, like the one of our dreams... (hope this is not a silly dream). I know that some of you are successfully using a single port BRI card. It is a kapejod's card too, right?? Thanks a lot! Sam p.s. probably this should go in a separate thread... - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 4:54 AM Subject: Re: [Asterisk-Users] newbie ISDN question Hi Thorsten, the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect phones to that card. The quadBRI card has 4 BRI ports that can individually be configured for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones). Please find the details at: http://www.junghanns.net/asterisk/page17.html best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ hi everybody, sorry for posting such a stupid question ;) i've managed to run asterisk* with my AVM fritz2.0 card and a some VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me ;))) now i want to run asterisk* istead of our old PBX. but it would be great to connect some phones directly to my box. how does a E100P from digium work. can i connect it to my ISDN-line and my internal phones (ISDN)? it would look like this: [PHONE2] / [PC]-[E100P] - [PHONE1] \ [ISDN-LINE] thank you for your help!!! thorsten ___ 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] H.323 protocol security vulnerability
Hello, two days ago *NISCC* released a security advisory reference about a security vulnerability in H323 protocol. http://www.uniras.gov.uk/vuls/2004/006489/h323.htm According to Graig Southeren from OpenH323 development team: OpenH323 is affected by SOME of the problems Both asterisk H323 channels are OpenH323 based. Here is a link to the Cisco advisory about this vulnerability: http://www.cisco.com/warp/public/707/cisco-sa-20040113-h323.shtml Lubo ___ 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] ISDN CAPI and anonymous callers
Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten = 5150/0,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ 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] ISDN CAPI and anonymous callers
Walter Doerr wrote: Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten = 5150/0,1,Congestion use: exten = 5150/,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ 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 -- Best regards Detlef Wengorz [EMAIL PROTECTED] Detlef Wengorz [EMAIL PROTECTED] ___ 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] Parking extension:700
Hi all, From Andy Powells Getting Started With Asterisk (V 0.1a) http://www.automated.it/guidetoasterisk.htm parking.conf file has this number set at 700. I've changed mine to 701 because I was having an issue with Asterisk - although it would 'see' (looking at the console) I had tried to transfer to 700 it appeared not to believe that I had dialed it. This was essentially due to the 00 in the 700, changing it to 701 eliminates the problem completely. The extension 700 is working for me. I am able to park the call. I am using SJPhone here. As stated in that guide Is there any problem with using 00? If yes, can anyone explain that? Regards... Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Join BharatMatrimony.com and get married. ___ 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] ISDN CAPI and anonymous callers
Walter Doerr schrieb: Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten = 5150/0,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? Do it the other way round, and just identify all callers with a number. Then do the congestion for the rest. I am not to good at extensions.conf, but the following should identify all callers with a number exten = 5150/_X.,1,Answer exten = 5150/_X.,999,Hangup exten = 5150,1,Congestion hth rgds pos ___ 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] * For Call Center
Hello, I think you need to do a little more looking around on the Asterisk resources and on Google. What you are trying to do is mostly possible if you have the time, patience and money to follow through with it. One thing you need to learn is that a great many on this list despise telemarketers of any kind, and even more so they hate it when people don't do their own research for questions that have been answered many times before. Asterisk cannot easily be turned into a predictive dialer(which is a good thing) there are a few people that have created simple auto-dialer programs that simply dial one number after another one-at-a-time(Myself included) which is not very efficient for bulk cold-calling, but works wonderfully to call back your customers on a quarterly basis for customer checkups like my company does with it's customers. There is a group of Asterisk users that decided to modify the code of Asterisk to try to make it a predictive dialer, called shady_dial I believe, but I haven't heard anything about it lately. Back to your questions: - yes, Asterisk can be used for a fractional voice/data T1. there are several people that have done this. - you will need a decent powered x86 computer with at least a digium single T1 card - yes Asterisk can be used as an auto-dialer, if you program it to do so yourself. The easiest way is to use the manager interface or generate .call files, but it is not very fast and will not piss people off at a high rate like a the predictive dialer you want will - maybe Asterisk can be turned into a predictive dialer, but you'd have to do that yourself or find out if shady_dial has succeeded in their project - yes you can use a screen popup system on win32 or Unix there are resources out there to do it and it's not terribly hard to do if you understand the APIs involved. - no, Asterisk will not work out of the box for what you want to do, it will take effort and time to get it all working and you may end up spending as much as you would on a predictive dialer to get it all done. - don't expect much sympathy from the people here on this list as the the cost of a predictive dialer. Thank you and good luck, MATT--- -Original Message- From: empire underground [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 11:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * For Call Center Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to do Voip at all... but that will change in the future. I have a T1 with 12 anolog lines and the rest for data (768k). I need to know what cards I should buy? I would also like to setup the box with 12-16 lines for outbound calling, and im nto trying to do (IVR). What I would like to do is make * either a predictive/auto dialer only. I read about a few people doing this when searching google but cant find the links anymore :( Aslo someone made a win32 program to log into * and get screen pops of all the info that was dialed for that # such as address, name, phone, ect... I dont realy care if I have to write an agi for it in linux because I hate winblows and would rather stay far far away from it ;) If anyone can help or point me in the right direction it would be much help ;) Also I have checked wiki allready... I cant really find anything there for this. Also is it even possible for this? I know I would have to write a agi for the screen pops to popup in web browser and rout that info to the person logged in and waiting in that queue, I was thinking about using sql backend for the db and maybe writing agi to import the .csv file? Also I was thinking about flying someone down here to Florida if all else fails (unless you already live here) to maybe help setup this type of box, or even giving root access to the box and configuring it? because a commercial dialer costs WAY too much! they want anywhere from $3500-30,000 for dialers... and then even pay another $1,500 for a license per agent that wil be using it! talk about getting raped! thanx for all you help in advance chris _ Scope out the new MSN Plus Internet Software - optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1 ___ 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]
Re: [Asterisk-Users] grandstream asterisk configuration
On Thu, Jan 15, 2004 at 02:45:22AM -0500, Steve said: On Thursday 15 January 2004 01:30 am, Chandra wrote: hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT. Why leave a host to defend for itself? At least behind a firewall you got some layers of protection. Um, just becasue a host isn't NATed, doesn't meant that there isn't a firewall protecting it. Nat is an evil hack used to allow more hosts to use the internet than you have real IP addresses. NAT itself is NOT a firewall, but it does have a side effect by nature that gives NATed hosts some protection. ___ 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: Re: [Asterisk-Users] How to park and pickup a call
By the way, I have an other question, are there any way to implement MeetMe conference If I haven't zaptel device? Zhang Peihao 2004-01-15 Yes, there is. Modify the Makefile in the /usr/src/zaptel directory, ie, change #ztdummy to ztdummy and run make clean; make install Find more information at: http://www.automated.it/guidetoasterisk.htm Regards... Girish _ Get head-hunted by 10,000 recruiters. http://go.msnserver.com/IN/35984.asp Post your CV on naukri.com today. ___ 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] Sending voicemail with qmail
Is * capable to use qmail as a MTA? If so, how can I set it? It shouldn't be an issue, as qmail has the standard 'sendmail' binary included. Regards, Andrew ___ 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] newbie ISDN question
hello klaus-peter this sounds great ; will the phones that are connected to a bri in nt-mode still allow all isdn-functions (in special : caller id-display)? thanks... - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 12:46 PM Subject: Re: [Asterisk-Users] newbie ISDN question Thorsten, theoretically you can connect 8 phones per port, but only 2 can be used at the same time. We advise to use 2 per port and in some scenarios 3 might be an option. So you can connect 8 ISDN phones to the quadBRI card. The drivers are still released as experimental and have some bugs. We are planning to be stable in about 2 weeks. The cards are in stock, so delivery will be fast. We ship with worldwide with UPS. best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/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
[Asterisk-Users] best SIP-softphone?
hi everybody. i'm currently looking for a good SIP-softphone. i tried x-lite but i'm not happy with it's soundquality as there is a very high noiselevel. do you know a better softphone? and would a hardware SIP-phone offer (almost) the same audioquality as my isdn-phone? if so what phone would you recommend (and where can i get it in germany)? thanks!! bye thorsten
[Asterisk-Users] Skinny behind NAT?
Can skinny work behind NAT? I have a Cisco 7910 using SCCP behind NAT that has one way audio. The called party cannot hear the calling party who's using the 7910. skinny.conf ; ; Skinny Configuration for Asterisk ; [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 0.0.0.0 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 ; allow = all disallow = all ;allow=g723.1 allow=g729 allow=ulaw allow=alaw ; Carlos' 7910 used for testing [7910] ; Device name device=SEP0007EBC7F3DE ; Offical identifier nat=1 callerid=TS Test 7910 954-937-8081 ;mailbox=2109 callwaiting=1 transfer=1 threewaycalling=1 context=skinny-longdistance line = 2109 -- David A. Lauer Network Engineer Tristar Communications [EMAIL PROTECTED] 954.977.8081 ext. 21 ___ 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] Sending voicemail with qmail
Is there someone here who could offer some assistance to a Linux noobie in getting an mta configured for a new * server. Off list of course. Michael On Thu, 15 Jan 2004 07:14:00 -0500, Andrew Kohlsmith wrote: Is * capable to use qmail as a MTA? If so, how can I set it? It shouldn't be an issue, as qmail has the standard 'sendmail' binary included. Regards, Andrew ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] If you ever reach total enlightenment while drinking beer, I bet it makes the beer shoot out your nose. - Deep Thought, Jack Handy ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] B-channels restart problem
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 me [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 _ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail ___ 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] B-channels restart problem
Hi ,
I'm having a problem that really bothers me , I havelooked for similar cases but couldn't really
find an answer .
I keep getting messages whichsays 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 textin 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 toat [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
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Re: [Asterisk-Users] Skinny behind NAT?
this is answered in the FAQ: http://www.voip-info.org/wiki-Asterisk+FAQ RTP based protocols (using RTP to transfer the voice in a separate UDP session) are generally hard to NAT. On Thu, 2004-01-15 at 13:29, David A. Lauer wrote: Can skinny work behind NAT? I have a Cisco 7910 using SCCP behind NAT that has one way audio. The called party cannot hear the calling party who's using the 7910. skinny.conf ; ; Skinny Configuration for Asterisk ; [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 0.0.0.0 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 ; allow = all disallow = all ;allow=g723.1 allow=g729 allow=ulaw allow=alaw ; Carlos' 7910 used for testing [7910] ; Device name device=SEP0007EBC7F3DE ; Offical identifier nat=1 callerid=TS Test 7910 954-937-8081 ;mailbox=2109 callwaiting=1 transfer=1 threewaycalling=1 context=skinny-longdistance line = 2109 ___ 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] re hardware requirement - asterisk
fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 For fxp0, the internal interface although the nic can do full-duplex it seems to me that it is only running simplex!! Same for xl0, the external interface. It is running 10BaseT but again it is simplex. Does that affect my voip performance? Is it true that every step of the way the network has to be full-duplex? There are no RFC standards on how duplex settings are negotiated across a cat 5 cable, etc. Most vendors support auto-negotiate, but somewhere near 50% of the time, its negotiated incorrectly. Part of the problem is that both ends of the cable attempt to negotiate at roughly the same time, one end locks into full while the other locks into half. When that happens, the end that thinks full duplex is fine steps all over the packets being sent from the half-duplex end, causing damaged packets, etc. Since we're talking about UDP traffic, that's Not A Good Thing. The system will run fine if both ends are operating at half duplex, however bandwidth (and performance) will be limited to something below about 30% utilization. In many systems, that is more then adequate. However, on a heavily loaded system, statically locking the interfaces (at both ends) to full duplex will allow utilizations up towards 90% without degradation. Rich ___ 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] asterisk.org webpage
hi all for new users, finding asterisk info is unneccesary troublesome. the asterisk.org page has very little information about the product and using google for 'asterisk' is like using google for 'linux'. you get all too many hits that has nothing to do with the product. perhaps the asterisk.org page at least should point to voip-info.org? or perhaps it's time someone rewrote the page? roy ___ 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
Hi - I noticed this as well a while ago, and spoke with Mark Spencer at Digium. I think he said it is normal for the channels to occasionally restart themselves, however I didn't think they were supposed to do it if they are in use. Perhaps you should send a message to [EMAIL PROTECTED] and see what they say. Do you have any other errors in your /var/log/asterisk/messages flle? regards Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ali Mughrabi Sent: Thursday, January 15, 2004 12:54 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] B-channels restart problem 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 me [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 _ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail ___ 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] B-channels restart problem
Ali,
If
Zap/82 is channel 20 on Span 3, then it looks like it's hanging up before the
channel restarts as this line indicates.
== Spawn extension (inbound, 9009170,
2) exited non-zero on 'Zap/82-1'
Maybe
there is a problem with your agi script.
B
channels only restart when the PRI line isn't busy.
-sb
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Ali
MughrabiSent: Thursday, January 15, 2004 7:48 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] B-channels
restart problem
Hi ,
I'm having a problem that really bothers me , I havelooked
for similar cases but couldn't really
find an answer .
I keep getting messages whichsays 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
textin 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 toat [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
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[Asterisk-Users] Strange sound when fax answers (app_rxfax)
Hi! I am struggling around with * and the spandsp library (app_rxfax) for a couple of days. I'm trying to receive faxes which come via a SIP gateway. The rxfax-module answers the call, the reception of faxes, however, still does not work correctly, the received file is only about 300 bytes of size, because the sending fax machine is terminating the transmission. Now I've figured out (by listening to the fax signal of the app_rxfax module) that the sound is somewhat different to the one of our regular fax machine (hylafax). Can that really be true? Did anyone experience the same problem in the past? Peter ___ 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] Problem at compiling zaptel
Hi all!
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS, but
now the next step at zaptel clean; make install fails.
Please have a look at the error-log below.
There must be a fundamental mis-configuration I suppose, but I am
unfortunately not an expert in this area.
Franz
-- error log -
lpc:/usr/src # cd zaptel
lpc:/usr/src/zaptel # make clean; make install rm -f torisatool makefw
tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f
zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h rm
-f libtonezone* rm -f tor2ee rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB
-I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes
-fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I
/usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c
zaptel.c
In file included from /usr/include/linux/module.h:20,
from zaptel.c:44:
/usr/include/asm/module.h:54:2: #error unknown processor family In file
included from /usr/include/linux/mm.h:205,
from /usr/include/asm/pci.h:7,
from /usr/include/linux/pci.h:677,
from zaptel.c:46:
/usr/include/linux/page-flags.h:119: error: `CONFIG_X86_L1_CACHE_SHIFT'
undeclared here (not in a function)
/usr/include/linux/page-flags.h:119: error: requested alignment is not a
constant In file included from zaptel.c:48:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2: #error These files are glibc internal and
may not match the
/usr/include/linux/version.h:9:2: #error currently running kernel. They
should only be
/usr/include/linux/version.h:10:2: #error included via other system header
files - user space
/usr/include/linux/version.h:11:2: #error programs should not directly
include linux/*.h or
/usr/include/linux/version.h:12:2: #error asm/*.h as well.
/usr/include/linux/version.h:13:2: #error
/usr/include/linux/version.h:14:2: #error To build kernel modules please do
the following:
/usr/include/linux/version.h:15:2: #error
/usr/include/linux/version.h:16:2: #error o Have the kernel sources
installed
/usr/include/linux/version.h:17:2: #error
/usr/include/linux/version.h:18:2: #error o Make sure that the symbolic
link
/usr/include/linux/version.h:19:2: #error/lib/modules/`uname -r`/build
exists and points to
/usr/include/linux/version.h:20:2: #errorthe matching kernel source
directory
/usr/include/linux/version.h:21:2: #error
/usr/include/linux/version.h:22:2: #error o Configure kernel sources:
/usr/include/linux/version.h:23:2: #error- cd /usr/src/linux
/usr/include/linux/version.h:24:2: #error- make mrproper
/usr/include/linux/version.h:25:2: #error- make cloneconfig
/usr/include/linux/version.h:26:2: #error- make dep
/usr/include/linux/version.h:27:2: #error
/usr/include/linux/version.h:28:2: #error o When compiling, make sure to
use the following
/usr/include/linux/version.h:29:2: #errorcompiler option to use the
correct include files:
/usr/include/linux/version.h:30:2: #error
/usr/include/linux/version.h:31:2: #error-I/lib/modules/`uname
-r`/build/include
/usr/include/linux/version.h:32:2: #error
/usr/include/linux/version.h:33:2: #errorinstead of
/usr/include/linux/version.h:34:2: #error
/usr/include/linux/version.h:35:2: #error-I/usr/include/linux
/usr/include/linux/version.h:36:2: #error
/usr/include/linux/version.h:37:2: #errorPlease adjust the Makefile
accordingly.
/usr/include/linux/version.h:38:2: #error
===
In file included from zaptel.h:36,
from zaptel.c:82:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2:
Re: [Asterisk-Users] grandstream asterisk configuration
I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable Chandra, I have this _exact_ same problem and it's the Netgear router corrupting the UDP checksums in the RTP packets. Specifically, the checksums come out of the phone unset and the router is setting them to incorrect values. Netgear has not yet responded to my support requets. Ethereal will confirm if you're getting the same thing. Swap out the Netgear with a Linksys or other router and I bet it 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
[Asterisk-Users] Problem at compiling zaptel (again)
Hi all!
Sorry, the error-log in my previous mail was disturbed.
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS, but
now the next step at zaptel clean; make install fails.
Please have a look at the error-log below.
There must be a fundamental mis-configuration I suppose, but I am
unfortunately not an expert in this area.
Franz
-- error-log -
lpc:/usr/src # cd zaptel
lpc:/usr/src/zaptel # make clean; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB
-I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes
-fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I
/usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c
zaptel.c
In file included from /usr/include/linux/module.h:20,
from zaptel.c:44:
/usr/include/asm/module.h:54:2: #error unknown processor family
In file included from /usr/include/linux/mm.h:205,
from /usr/include/asm/pci.h:7,
from /usr/include/linux/pci.h:677,
from zaptel.c:46:
/usr/include/linux/page-flags.h:119: error: `CONFIG_X86_L1_CACHE_SHIFT'
undeclared here (not in a function)
/usr/include/linux/page-flags.h:119: error: requested alignment is not a
constant
In file included from zaptel.c:48:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2: #error These files are glibc internal and
may not match the
/usr/include/linux/version.h:9:2: #error currently running kernel. They
should only be
/usr/include/linux/version.h:10:2: #error included via other system header
files - user space
/usr/include/linux/version.h:11:2: #error programs should not directly
include linux/*.h or
/usr/include/linux/version.h:12:2: #error asm/*.h as well.
/usr/include/linux/version.h:13:2: #error
/usr/include/linux/version.h:14:2: #error To build kernel modules please do
the following:
/usr/include/linux/version.h:15:2: #error
/usr/include/linux/version.h:16:2: #error o Have the kernel sources
installed
/usr/include/linux/version.h:17:2: #error
/usr/include/linux/version.h:18:2: #error o Make sure that the symbolic
link
/usr/include/linux/version.h:19:2: #error/lib/modules/`uname -r`/build
exists and points to
/usr/include/linux/version.h:20:2: #errorthe matching kernel source
directory
/usr/include/linux/version.h:21:2: #error
/usr/include/linux/version.h:22:2: #error o Configure kernel sources:
/usr/include/linux/version.h:23:2: #error- cd /usr/src/linux
/usr/include/linux/version.h:24:2: #error- make mrproper
/usr/include/linux/version.h:25:2: #error- make cloneconfig
/usr/include/linux/version.h:26:2: #error- make dep
/usr/include/linux/version.h:27:2: #error
/usr/include/linux/version.h:28:2: #error o When compiling, make sure to
use the following
/usr/include/linux/version.h:29:2: #errorcompiler option to use the
correct include files:
/usr/include/linux/version.h:30:2: #error
/usr/include/linux/version.h:31:2: #error-I/lib/modules/`uname
-r`/build/include
/usr/include/linux/version.h:32:2: #error
/usr/include/linux/version.h:33:2: #errorinstead of
/usr/include/linux/version.h:34:2: #error
/usr/include/linux/version.h:35:2: #error-I/usr/include/linux
/usr/include/linux/version.h:36:2: #error
/usr/include/linux/version.h:37:2: #errorPlease adjust the Makefile
accordingly.
/usr/include/linux/version.h:38:2: #error
===
In file included from zaptel.h:36,
from zaptel.c:82:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
Re: [Asterisk-Users] Re Hardware requirement -Asterisk
On my Linux box mii-tool yeilds the following which shows 100mbs full duplex. [EMAIL PROTECTED] gford]# mii-tool eth0: negotiated 100baseTx-FD, link ok /glen [EMAIL PROTECTED] wrote: My ADSL speed is Uplink 128kbit and Downstream 512kbit. The mii-tool does not tell whether eth0 is in full-duplexed mode. It just say that it is 100baseTx. David Kwok -- Glen Ford [EMAIL PROTECTED] ___ 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] Problem at compiling zaptel
Hello,
I'm also new to * but I think that this is what we had to do:
You need to make sure that the following packages are installed on your
system:
-OpenSSL-Devel
-Ncurses
-Ncurses-Devel (C++)
-sox
-kernel sourses
-kernel development
-bison
-newt
-newt-devel
-readline
-readline-devel
If you already have all of these I know that there were some bugs in the
latest CVS but I'm not sure if that's the problem. I'm also curious
which distribution you're running. If you do have these please do the
following and then try to compile again:
In /usr/src/linux
#make mrproper
#make menuconfig (exit without changing anything)
#make dep
then in /usr/src/zaptel
make clean install
Hope this helps
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: Franz Edler [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 9:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem at compiling zaptel
Hi all!
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS,
but
now the next step at zaptel clean; make install fails.
Please have a look at the error-log below.
There must be a fundamental mis-configuration I suppose, but I am
unfortunately not an expert in this area.
Franz
-- error log -
lpc:/usr/src # cd zaptel
lpc:/usr/src/zaptel # make clean; make install rm -f torisatool makefw
tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f
zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h
rm
-f libtonezone* rm -f tor2ee rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__
-DEXPORT_SYMTAB
-I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes
-fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I
/usr/src/linux/include -I/usr/src/linux/include/net
-DSTANDALONE_ZAPATA -c
zaptel.c
In file included from /usr/include/linux/module.h:20,
from zaptel.c:44:
/usr/include/asm/module.h:54:2: #error unknown processor family In file
included from /usr/include/linux/mm.h:205,
from /usr/include/asm/pci.h:7,
from /usr/include/linux/pci.h:677,
from zaptel.c:46:
/usr/include/linux/page-flags.h:119: error: `CONFIG_X86_L1_CACHE_SHIFT'
undeclared here (not in a function)
/usr/include/linux/page-flags.h:119: error: requested alignment is not a
constant In file included from zaptel.c:48:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the
compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header
files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2: #error These files are glibc internal
and
may not match the
/usr/include/linux/version.h:9:2: #error currently running kernel. They
should only be
/usr/include/linux/version.h:10:2: #error included via other system
header
files - user space
/usr/include/linux/version.h:11:2: #error programs should not directly
include linux/*.h or
/usr/include/linux/version.h:12:2: #error asm/*.h as well.
/usr/include/linux/version.h:13:2: #error
/usr/include/linux/version.h:14:2: #error To build kernel modules
please do
the following:
/usr/include/linux/version.h:15:2: #error
/usr/include/linux/version.h:16:2: #error o Have the kernel sources
installed
/usr/include/linux/version.h:17:2: #error
/usr/include/linux/version.h:18:2: #error o Make sure that the
symbolic
link
/usr/include/linux/version.h:19:2: #error/lib/modules/`uname
-r`/build
exists and points to
/usr/include/linux/version.h:20:2: #errorthe matching kernel source
directory
/usr/include/linux/version.h:21:2: #error
/usr/include/linux/version.h:22:2: #error o Configure kernel sources:
/usr/include/linux/version.h:23:2: #error- cd /usr/src/linux
/usr/include/linux/version.h:24:2: #error- make mrproper
/usr/include/linux/version.h:25:2: #error- make cloneconfig
/usr/include/linux/version.h:26:2: #error- make dep
/usr/include/linux/version.h:27:2: #error
/usr/include/linux/version.h:28:2: #error o When compiling, make sure
to
use the following
/usr/include/linux/version.h:29:2: #errorcompiler option to use the
correct include files:
/usr/include/linux/version.h:30:2: #error
/usr/include/linux/version.h:31:2: #error-I/lib/modules/`uname
-r`/build/include
/usr/include/linux/version.h:32:2: #error
/usr/include/linux/version.h:33:2: #errorinstead of
Re: [Asterisk-Users] Strange sound when fax answers (app_rxfax)
Have you confirmed that the call is using the ulaw or alaw codec? It won't work otherwise. On Thu, 2004-01-15 at 08:13, Peter Bittner wrote: Hi! I am struggling around with * and the spandsp library (app_rxfax) for a couple of days. I'm trying to receive faxes which come via a SIP gateway. The rxfax-module answers the call, the reception of faxes, however, still does not work correctly, the received file is only about 300 bytes of size, because the sending fax machine is terminating the transmission. Now I've figured out (by listening to the fax signal of the app_rxfax module) that the sound is somewhat different to the one of our regular fax machine (hylafax). Can that really be true? Did anyone experience the same problem in the past? Peter ___ 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 -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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] ISDN newbie
Mine is a small company with 2 incoming analog lines that we use for voice. One line rolls over to the other if the first is busy. I started using an */grandstream combo a while ago, and besides a couple of bugs that I have yet to work out (echo, ringing in the earpiece) its has been good for the company. I have been thinking that perhaps getting some digital voice channels, instead of analog lines, will help with the echo problem. It also seems like the next logical step in expanding to more lines. Can ISDN do this? I have heard that ISDN cannot do hunting; or I cannot have one number for multiple lines with 'ringdown'. Is that true? Is there something else that is preferable to just getting more analog lines? I think a T1 is a bit out of our price range at this point, but I don't know. How much is a T1 anyway? Thanks for any info, Sean R. ___ 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] Problem at compiling zaptel (again)
On Thursday 15 January 2004 08:37, Franz Edler wrote: Can anybody please give me some advice, what is wrong at my first try to compile Asterisk. I have successfully downloaded the sources from CVS, but now the next step at zaptel clean; make install fails. Please have a look at the error-log below. There must be a fundamental mis-configuration I suppose, but I am unfortunately not an expert in this area. snip You don't have kernel-source and kernel-headers installed. -Tilghman ___ 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] ATA186 SIP Outbound Fax Calls
All, I was wondering if anyone has any experience with the Cisco ATA186 (SIP image) and outbound faxing with Asterisk. Inbound faxs from PSTN into * and on to the ATA work fine but outbound faxs receive congestion from *. I've got packet dumps from both sides and everything appears normal but after about 3 seconds the * servers sends the AS5300 a CANCEL and sends the ATA a '503 Service Unavailable' (with CSeq: 2 INVITE). The ATA responds with a SIP 2 ACK but does not stop sending RTP packets but the * server has taken down its RTP state so responds with ICMP port unreachables. I've disable all fax tone detection on the ATA and AS5300 but still can't seem to get this to work. If anyone has any advise or recommended ATA configs it would be much appreciated. Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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] Basic Asterisk capabilities question
Whaaa?? So, to allow 24+ lines of dial in access, how would I configure it? Would I need to purchase or lease a voice-over-ip box to connect our T1 or phone lines into? And then from there send the VOIP to the linux/Asterisk box for recording? (forgive me, Im new to telephony, but I need to make this work) :-) Ive started looking for some good AGI examples... It looks promising. Thanks Gary F. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of C. Maj Sent: Wednesday, January 14, 2004 4:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Basic Asterisk capabilities question On Wed, 14 Jan 2004, Gary Franczyk waxed: I'd like to configure a voice recording system using Asterisk and a Tormenta2 Quad T1 card. A co-worker was able to create this system a while back with Bayonne and a Dialogic card, but I would like to do the same thing with much cheaper hardware. I do not believe Bayonne supports the Tormenta2. (I think it is also known as the Wildcard TE410P) Yes, but the TE410P is not required. Only if you want T1 connections. You could just use a regular linux box and VoIP, for example. When the user dials in: - He enters his user id - he enters his password - he begins his recording. - while recording, he has the option to pause, continue, rewind by 4 seconds and review his recording, - press a button to start another recording. - press a button to recieve a recording ID number In addition to this, I would like the user to be able to call in again later and listen to his previous recordings by entering his recording id numbers. So, Is this possible and resonable to accomplish with the Asterisk system? This system is possible with Asterisk, using the Asterisk Gateway Interface (AGI). You could code it in your favorite language, too. Or is Asterisk more of an out-of-the-box PBX rather than a voice application system? It's not a turnkey system as downloaded from CVS. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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] wav49 voicemail problem with Windows Media Player
Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08:32. Is anyone else having this problem? For a quick check click on the bug link above and then try to play the attached wav file with your Windows Media Player. It would be great if you could also verify if wav49 files recorded on your Asterisk machine give the error. 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] Choosing a VoIP Protocol.
Hello everybody, I have the next scenario, working now. ISDN Network AGW IP 1(*)IP NetworkGW IP 2(*)ISDN Network B Now we are looking the avalibility of sending call's from 1 to 2, using an VoIP Protocol supported by *. But we have the next questions, before began to do it. 1. There are any VoIP protocol that allow us to send the source number from ISDN Network A and the destination number (in ISDN Network B), during the session establishment of the call. 2. It's possible that * take the destination number and dial it automatically without any user accion to the ISDN Network B. Note, that we have E400P installed in IP GW2. Thanks a lot for you help. Bye bye ___ 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] wav49 voicemail problem with Windows Media Player
I'm having the same problem. Warwick - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 5:39 PM Subject: [Asterisk-Users] wav49 voicemail problem with Windows Media Player Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08:32. Is anyone else having this problem? For a quick check click on the bug link above and then try to play the attached wav file with your Windows Media Player. It would be great if you could also verify if wav49 files recorded on your Asterisk machine give the error. 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 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
The restarts only occur on idle channels. However, what is interesting is that according to this log, the *switch* requests a restart on channel 20 (which if my calculations are correct, is the channel that has the call on it). You can see this because we log the messages on getting RESTART_ACK as successfully restarted and the ones on the RESTART request as simply restarted. You can also confirm this by doing a pri debug span 3. You can also increase this time by adjusting RESET_INTERVAL at the top of channels/chan_zap.c to make the reset of idle channels less frequent, but generally speaking the reset of idle channels is a Good Thing to make sure both sides are always in sync, state wise, on a channel, and clearly the other side is requesting the RESTART on the channel with an active call. Mark On Thu, 15 Jan 2004, Anton Tinchev wrote: 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 ___ 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] zaptel compile erro!(asterisk last version0.7.1)
you don't have libm (m for math) library ? Martin On Thu, 15 Jan 2004, [gb2312] Âí÷ë wrote: erro cocent:cc -shared -Wl,-soname,libtonezone.so.1 -lm -o libtonezone.so.1.0 zonedata.lo tonezone.lo /sbin/ldconfig -n . ln -sf libtonezone.so.1 libtonezone.so cc -o ztcfg ztcfg.o -lm -L. -ltonezone ./libtonezone.so: undefined reference to `cos' ./libtonezone.so: undefined reference to `sin' ./libtonezone.so: undefined reference to `pow' collect2: ld returned 1 exit status make: *** [ztcfg] Error 1 ___ 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] ultra-cheap asterisk box
hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy ___ 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] Disturbing trend of * production boxes that shouldn't be
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Franczyk Sent: Thursday, January 15, 2004 10:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question Whaaa?? So, to allow 24+ lines of dial in access, how would I configure it? Would I need to purchase or lease a voice-over-ip box to connect our T1 or phone lines into? And then from there send the VOIP to the linux/Asterisk box for recording? (forgive me, Im new to telephony, but I need to make this work) :-) This is a disturbing trendpeople who don't know much about Asterisk and/or Linux and/or telephony who Need to make these things work or need to know how to update [their] production box installation...sorry I don't know Linux at all. Asterisk is great, but to maintain it and especially to repair things when they've gone wrong, you need to know what you're doing. Its your job (the people I'm talking about know who they are), so do as you wish, but I sure would install something I know little or nothing about and call it production. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] wav49 voicemail problem with Windows Media Player
I am having problems too Just shy of the 5-second mark in the test vm. WMP 9.00.00.3075 Windows 2000 SP4 -Original Message- From: Warwick Ward-Cox [mailto:[EMAIL PROTECTED] Sent: Thursday, January 15, 2004 10:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wav49 voicemail problem with Windows Media Player I'm having the same problem. Warwick - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 5:39 PM Subject: [Asterisk-Users] wav49 voicemail problem with Windows Media Player Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08:32. Is anyone else having this problem? For a quick check click on the bug link above and then try to play the attached wav file with your Windows Media Player. It would be great if you could also verify if wav49 files recorded on your Asterisk machine give the error. 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 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] ISDN CAPI and anonymous callers
Hi!
I am trying to use * to handle anonymous ISDN callers.
exten = 5150/0,1,Congestion
should work but doesn't. Apparently because the ISDN CAPI doesn't
use 0 for callers who don't send their number.
Is there a way to make * identify ISDN callers who use CLIR?
Here is what I do:
exten = 22,1,Answer
exten = 22,2,SetCallerID(ISDN 0${CALLERIDNUM})
exten = 22,3,GotoIf($[${CALLERIDNUM} = 0]?6:4)
exten = 22,4,LookupCIDName
exten = 22,5,Goto(7)
exten = 22,6,SetCallerID(ISDN hidden 0)
exten = 22,7. ...continue your code...
Cheers, Philipp
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RE: [Asterisk-Users] wav49 voicemail problem with Windows Media Player
I can't reproduce this either, but I do have the gsm codec installed (though WMP won't play a .gsm file). I play the wav49 files in Winamp with no issue. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warwick Ward-Cox Sent: Thursday, January 15, 2004 10:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wav49 voicemail problem with Windows Media Player I'm having the same problem. Warwick - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 5:39 PM Subject: [Asterisk-Users] wav49 voicemail problem with Windows Media Player Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08:32. Is anyone else having this problem? For a quick check click on the bug link above and then try to play the attached wav file with your Windows Media Player. It would be great if you could also verify if wav49 files recorded on your Asterisk machine give the error. 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 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] People detected as fax machines
A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include = mailboxes include = day|07:55-23:00 include = night exten = t,1,Voicemail2,100 exten = t,2,Hangup [day] ; set music on hold for parked calls exten = s,1,setmusiconhold,default exten = s,2,responsetimeout,20 ; ring SIP for 20 seconds exten = s,3,Dial,sip/ciscosip/cisco1sip/cisco2sip/cisco3|20|tT ;if nobody answers tell them how to use the voicemail system. ; exten = s,4,Background,vmprompt exten = s,5,Voicemail2,100 exten = s,6,Hangup ___ 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] 100% of cpu in an out of the box *
Me too :( 100% CPU. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of F.G.Testa Sent: 14 January 2004 20:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 100% of cpu in an out of the box * Hi all! I'm newbie, so here goes my situation: I have succefully compiled the cvs version as shown in asterisk website in some linux distros: Debian (2.4.22), Conectiva, Fedora Core 1 and in all of them, * starts and consumes all the cpu (on top). Does anybody know this issue? Thanks! Testa ___ 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] asterisk.org webpage
Hi! for new users, finding asterisk info is unneccesary troublesome. the asterisk.org page has very little information about the product and using google for 'asterisk' is like using google for 'linux'. you get all too many hits that has nothing to do with the product. perhaps the asterisk.org page at least should point to voip-info.org? or perhaps it's time someone rewrote the page? Here's my little attempt to adjust the README file: http://bugs.digium.com/bug_view_page.php?bug_id=846 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] ultra-cheap asterisk box
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Thursday, January 15, 2004 11:08 AM To: Asterisk Users Subject: [Asterisk-Users] ultra-cheap asterisk box hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 ( 216) plus a small 50 drive and cables, so say 300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? I've got one system with 10 IP phones + SIP term + 2 FXS + 4 FXO running on a P700 with 256 MB RAM. It works just fine, and the CPU is rarely over 40%. Sounds like that box will work from a capability standpoint. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] Major format changes
On Thu, 2004-01-15 at 10:41, Robert Murray wrote: Hi Mark Would it be possible to include a way of streaming audio from memory? For example registering a file type which read from a fifo in memory? I need this for app_theta. (Cepstral TTS) I could copy the code from file.c, but I think it would be better if the same code could be used to avoid duplication. Whats wrong with just creating your frames and handing them off to be dealt with? Maybe you should check out the stuff in app_festival. On Sat, Jun 28, 2003 at 05:48:50PM -0500, Mark Spencer wrote: I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ 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] ultra-cheap asterisk box
I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. Ideally Asterisk will be ported one day to Linux/ARM or some other very low cost platform. for VOIP you do not need the PCI slots. In theory Asterisk could run on a Lynksys router box with re-flashed EEPROM. After all Lynksys' latest wireless router runs Linux inside Low cost to me means low total cost of ownership To get this I don't think buying the lowest priced parts is the way to go. I want quality mainboard, and a quality power supply and, this is importernt: A low internal case temperature. for this reason I'll spend the extra $50 to go with Antec cases and ASUS mainboards over the generic ones. What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] Users in sweden
Hi, Any * users in sweden, particularly in the Malmo or Lund areas? Mail me off-list, i have some questions :) Faiz ___ 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] * For Call Center
On Thu, 15 Jan 2004, Steve waxed: On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote: sounds like one of those pesky auto dialers the simpsons make fun of. It sure does... The AT-5000 was Prof. Frink's first patent, and it was designed to alert children of snow days and such. I think Homer bought it at one of those pesky police auctions, you know, the ones where the liberty and freedom loving US government says your property is guilty of a crime and theirs to sell... But don't forget that Prof. Frink went on to invent such wonders as the Flying Motorcycle, a Matter Transporter, and the Frinkahedron: http://www.internerd.com/frink.retired/frinkv.3/inventions/ --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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] re hardware requirement - asterisk
--- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have just checked the Openbsd box on the if interface. fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 For fxp0, the internal interface although the nic can do full-duplex it seems to me that it is only running simplex!! Why do you think it is running simplex. I read the above and see where it says (100baseTX full-duplex) I don't think 10BaseT can run full duplex. I could be wrong but I don't think so. But why does it matter? A single VOIP connection will not even use 1% of a simplex 10BaseT. Simplex 100BaseT should be able to handle dozens and dozens of calls Same for xl0, the external interface. It is running 10BaseT but again it is simplex. Does that affect my voip performance? Is it true that every step of the way the network has to be full-duplex? David Kwok = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] People detected as fax machines
On Thursday, January 15, 2004 10:42 AM, Iain Stevenson [SMTP:[EMAIL PROTECTED] wrote: ... Is there any way to stop * even considering an incoming call on a line as a fax call? Sure, just don't have exten = fax. in the same context (or included context). Iain -- Don Pobanz ___ 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] People detected as fax machines
I think this is a MAROR bug in the new dsp.c routines, recompile using the old dsp stuff by changing the makefile and set OLD_DSP_ROUTINES - Original Message - From: Iain Stevenson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 8:41 AM Subject: [Asterisk-Users] People detected as fax machines A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include = mailboxes include = day|07:55-23:00 include = night exten = t,1,Voicemail2,100 exten = t,2,Hangup [day] ; set music on hold for parked calls exten = s,1,setmusiconhold,default exten = s,2,responsetimeout,20 ; ring SIP for 20 seconds exten = s,3,Dial,sip/ciscosip/cisco1sip/cisco2sip/cisco3|20|tT ;if nobody answers tell them how to use the voicemail system. ; exten = s,4,Background,vmprompt exten = s,5,Voicemail2,100 exten = s,6,Hangup ___ 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] * For Call Center
On Thu, 15 Jan 2004, mattf waxed: 8's There is a group of Asterisk users that decided to modify the code of Asterisk to try to make it a predictive dialer, called shady_dial I believe, but I haven't heard anything about it lately. http://shadydial.sourceforge.net/ Lots of recent updates made in CVS, and it works with the latest and greatest * CVS, too. No screen pops yet, but that is the next step. Call results are simply logged in the phone, which is pretty sloppy since it resides in the agent hangup function. Francois Lambert posted some time ago on -dev that his company had worked on a predictive dialer with answering machine detection. Said they hacked * code a little, too, and since it's GPL I would be interested in seeing it. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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] People detected as fax machines
If you don't have a fax connected to * then create and exten: exten = fax,1,Goto(day,s,1) I had the same today... :/ Andy *** REPLY SEPARATOR *** On 15/01/2004 at 16:41 Iain Stevenson wrote: A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include = mailboxes include = day|07:55-23:00 include = night exten = t,1,Voicemail2,100 exten = t,2,Hangup [day] ; set music on hold for parked calls exten = s,1,setmusiconhold,default exten = s,2,responsetimeout,20 ; ring SIP for 20 seconds exten = s,3,Dial,sip/ciscosip/cisco1sip/cisco2sip/cisco3|20|tT ;if nobody answers tell them how to use the voicemail system. ; exten = s,4,Background,vmprompt exten = s,5,Voicemail2,100 exten = s,6,Hangup ___ 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] ultra-cheap asterisk box
It's not what I would want to depend on day in and day out. I know that you can buy Dell PowerEdge SC400 servers for $299 with HDD, memory, and either a celeron or p4, depending on what day of the week it is. I'd put my name on the Dell based solution before the white box solution for the same money. --Mike hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy ___ 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] Cisco FXO as PSTN gateway (updated request for assistance)
I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see the rejection message: y.y.y.y x.x.x.x INVITE sip:[EMAIL PROTECTED]:5060 x.x.x.x y.y.y.y Status: 100 Trying x.x.x.x y.y.y.y Status: 503 Service Unavailable y.y.y.y x.x.x.x Request: ACK sip:[EMAIL PROTECTED]:5060 (resent as retested with 0.7.1 the addition of autocreatepeer=yes) Thanks a lot, Fran. ___ 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] Credit Card Terminal
Hello, I have a Hypercom T7P swipe card terminal sitting on a dedicated G711ulaw port. The Hypercom operates at either 1200 or 2400bps. I get about a 50% success rate when I try to authorize cards. On this same G711ulaw port, I have a fax machine with a 100% success rate operating at 9600bps. Any suggestions on how to change *, ATA186, or SIPURA SPA-2000 to enhance the card terminals ability to process would be appreciated. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com ___ 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] Disturbing trend of * production boxes that shouldn't be
how do you spell Teleecooomm again? [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Franczyk Sent: Thursday, January 15, 2004 10:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question Whaaa?? So, to allow 24+ lines of dial in access, how would I configure it? Would I need to purchase or lease a voice-over-ip box to connect our T1 or phone lines into? And then from there send the VOIP to the linux/Asterisk box for recording? (forgive me, Im new to telephony, but I need to make this work) :-) This is a disturbing trendpeople who don't know much about Asterisk and/or Linux and/or telephony who Need to make these things work or need to know how to update [their] production box installation...sorry I don't know Linux at all. Asterisk is great, but to maintain it and especially to repair things when they've gone wrong, you need to know what you're doing. Its your job (the people I'm talking about know who they are), so do as you wish, but I sure would install something I know little or nothing about and call it production. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] People detected as fax machines
- Original Message - From: Iain Stevenson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 11:41 AM Subject: [Asterisk-Users] People detected as fax machines A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Search the archives... google for: site:lists.digium.com fax detection You're looking for something about OLD_DSP_ROUTINES It's been discussed several times now. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ 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] asterisk.org webpage
I get 'Access Denied'... Can it be downloaded zip or tar ball? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Thursday, January 15, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk.org webpage Hi! for new users, finding asterisk info is unneccesary troublesome. the asterisk.org page has very little information about the product and using google for 'asterisk' is like using google for 'linux'. you get all too many hits that has nothing to do with the product. perhaps the asterisk.org page at least should point to voip-info.org? or perhaps it's time someone rewrote the page? Here's my little attempt to adjust the README file: http://bugs.digium.com/bug_view_page.php?bug_id=846 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 ___ 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] ultra-cheap asterisk box
Are you wanting to make a pre-built * box, with hardware to connect a single dial line and one traditional phone, or.. ? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 11:08 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ultra-cheap asterisk box -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Thursday, January 15, 2004 11:08 AM To: Asterisk Users Subject: [Asterisk-Users] ultra-cheap asterisk box hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 ( 216) plus a small 50 drive and cables, so say 300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? I've got one system with 10 IP phones + SIP term + 2 FXS + 4 FXO running on a P700 with 256 MB RAM. It works just fine, and the CPU is rarely over 40%. Sounds like that box will work from a capability standpoint. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] * For Call Center
Actually he found it in the dumpster after the police threw it out following a bust! Does anyone want to send a dollar to Mr. Happy?! -Original Message- From: C. Maj [mailto:[EMAIL PROTECTED] Sent: Thursday, January 15, 2004 12:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * For Call Center On Thu, 15 Jan 2004, Steve waxed: On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote: sounds like one of those pesky auto dialers the simpsons make fun of. It sure does... The AT-5000 was Prof. Frink's first patent, and it was designed to alert children of snow days and such. I think Homer bought it at one of those pesky police auctions, you know, the ones where the liberty and freedom loving US government says your property is guilty of a crime and theirs to sell... But don't forget that Prof. Frink went on to invent such wonders as the Flying Motorcycle, a Matter Transporter, and the Frinkahedron: http://www.internerd.com/frink.retired/frinkv.3/inventions/ --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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] Free Message Signaling
Hi To ALL i have made an application for billing a traffic but i have strange problem with free message from Telco provider because when dial the number and Telco reply The customer have change number... i dont receive a connect so i cant listen nothing... Yes is right from PRI dont receive a connect signal but how can listen the mex? Thanks in advance Dimi PS: Happy New Year ___ 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] ultra-cheap asterisk box
On Thu, 2004-01-15 at 14:31, Chris Albertson wrote: I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. I'm also looking at this. I was thinking on a system without a hard drive, booting from a pendrive or flashdive. I want to avoid moving parts, they always break or get dirty and are noisy. If there are other people working on this, we might join efforts and work together and came up with a small linux version with asterisk included, that can boot from a pendrive or a cdrom. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ 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] re hardware requirement - asterisk
well, it does say SIMPLEX in the fxp0 flags section. I don't honestly know if this means it's negotiated half duplex, or something beyond that 10baseT is capable of running full duplex, although this requires a NIC capable of is, as well as a switch that can do FD. And regarding the 1% comment, the benefit with full duplex comes in to play with collisions, not so much traffic amounts. -Original Message- From: Chris Albertson [mailto:[EMAIL PROTECTED] Sent: Thursday, January 15, 2004 12:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] re hardware requirement - asterisk --- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have just checked the Openbsd box on the if interface. fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 For fxp0, the internal interface although the nic can do full-duplex it seems to me that it is only running simplex!! Why do you think it is running simplex. I read the above and see where it says (100baseTX full-duplex) I don't think 10BaseT can run full duplex. I could be wrong but I don't think so. But why does it matter? A single VOIP connection will not even use 1% of a simplex 10BaseT. Simplex 100BaseT should be able to handle dozens and dozens of calls Same for xl0, the external interface. It is running 10BaseT but again it is simplex. Does that affect my voip performance? Is it true that every step of the way the network has to be full-duplex? David Kwok = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] Asterisk drops calls - E100P
Don Pobanz wrote: On Wednesday, January 14, 2004 5:48 AM, Daniel Bichara [SMTP:[EMAIL PROTECTED]] wrote: Hi, Once a day, * drops all calls (E100P board). Yesterday, I updated * version to CVS but I got the problem again today. Monitoring log files, I found this messages just before: Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 25 failed: Unknown error 500 Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Short write: -1/5 (Unknown error 500) Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 25 failed: Unknown error 500 Few minutes after this, everything becomes fine. Any clue? just a guess here... The simple answer is have you verified that loop timing is set up in zaptel.conf. If not in loop timing a slip could cause the drop. Loop timing on span 2 as primary timing would be: span=2,1,0,esf,b8zs Does this happen at the same time every day? If so it does not sound like a timing issue. If at random times, it could be. Hi Don, You are right! Yesterday, my telco called me about slip. I changed timing and now it is ok. Daniel Daniel Don Pobanz ___ 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] re hardware requirement - asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson Sent: Thursday, January 15, 2004 12:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] re hardware requirement - asterisk I don't think 10BaseT can run full duplex. I could be wrong but I don't think so. Where'd you get that idea from? A 10-Base-T connection to a switch port most definitely will (and should) fun full duplex. But why does it matter? A single VOIP connection will not even use 1% of a simplex 10BaseT. Simplex 100BaseT should be able to handle dozens and dozens of calls Properly configured, yes. I don't know the details of your issue, but I've seen more shoddily auto-detected connections that I care to remember (3Com cards on Auto - Cisco Catalyst on Auto anyone?). Lock the speed/duplex on the switch and the server, and check for collisions, etc. on the port. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] cdr processing
Hi friends, Could some one recommand a good cdr processing software out there for post paid billing (invoicing, web-based payment processing) etc., Thanks a bunch. SW ___ 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] capacity testing
Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip-h323 gateway to tie SIP H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN-TNT-Asterisk-SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP-ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this: Run asterisk with -vvvcg. Do your test (core file generated). Run gdb /usr/sbin/asterisk core_filename From within gdb run bt and send me the output of it. if it is of use, here it is (from asterisk v.0.5.0) - (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at channel.c:1385 #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at res_parking.c:224 #5 0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at app_dial.c:668 #6 0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, data=0x6ef216e8, newstack=1) at pbx.c:396 #7 0x08068c61 in pbx_extension_helper (c=0x5de5ed98, context=0x5de5eeec longdistance, exten=0x8214488 H323:8257, priority=2, callerid=0x5de10048 \Jesse Peterson\ 2474766, action=1104606132) at pbx.c:1150 #8 0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634 #9 0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855 #10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0 - If anyone has tried something like this or has any comments, I'd be interested in hearing from them. jesse ___ 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] Possible Bug: Crash when Parking Calls
Hi, I'm relativle new to *, so I may be wrong. I build up * from cvs today (show version: CVS-01/15/04-16:27:36). In an test I use 2 SIP phones (linphone) to connect to eachother. The phones are called via the extensions 100 (user 'kwe') and 200 (user 'phone'). I can call from one to another and I can park a call and take it again: 1) call from 100 to 200 2) press # and dial 700 on phone '200' 3) hear the voice and get the call back (dial 701 on phone '200') the steps can be repeated. But * crahes if I do the following during the same call: 4) pree # and dial 700 on phone '100' I can reproduce this. Every time I park a call first on one end and lateron on the other end, * crahes. On console I see the following: ---snip--- -- Called phone -- SIP/phone-c9da is ringing -- SIP/phone-c9da answered SIP/kwe-9857 -- Attempting native bridge of SIP/kwe-9857 and SIP/phone-c9da -- Started music on hold, class 'default', on SIP/kwe-9857 -- Playing 'pbx-transfer' (language 'de') -- Stopped music on hold on SIP/kwe-9857 -- Started music on hold, class 'default', on SIP/kwe-9857 == Parked SIP/kwe-9857 on 701 -- Playing 'digits/7' (language 'de') -- Playing 'digits/0' (language 'de') -- Playing 'digits/1' (language 'de') == Spawn extension (default, 200, 1) exited KEEPALIVE on 'SIP/kwe-9857' -- Executing ParkedCall(SIP/phone-b583, 701) in new stack -- Stopped music on hold on SIP/kwe-9857 -- Channel SIP/phone-b583 connected to parked call 701 -- Attempting native bridge of SIP/kwe-9857 and SIP/phone-b583 -- Started music on hold, class 'default', on SIP/phone-b583 -- Playing 'pbx-transfer' (language 'de') -- Stopped music on hold on SIP/phone-b583 -- Started music on hold, class 'default', on SIP/phone-b583 == Parked SIP/phone-b583 on 701 -- Playing 'digits/7' (language 'de') -- Playing 'digits/0' (language 'de') -- Playing 'digits/1' (language 'de') == Spawn extension (default, 701, 1) exited non-zero on 'SIP/phone-b583' -- Stopped music on hold on SIP/phone-b583 pbx*CLI Disconnected from Asterisk server Executing last minute cleanups ---snip--- I would be glad, if someone could give me some information: a) Yes I can reproduce this b) Yes, it's a known bug c) No, I can not reproduce this In case of c): could You please give me some advise, how to locate the error? Thanks in advance Karsten ___ 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] Cooperate with SIP ITSP
- Original Message -
From: Zhang Peihao [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 8:51 PM
Subject: [Asterisk-Users] Cooperate with SIP ITSP
Hi All,
When I want use Asterisk as a PBX to cooperate SIP ITSP,
I can not set the caller ID, so SIP ITSP do not accept
the call.
In Asterisk, I set a account in sip.conf to register on
ITSP SIP Server:
register = [EMAIL PROTECTED]/6292
And I added a user 6292 in Asterisk just like the account
on ITSP SIP Server:
[6291]
type=friend
username=6291
callerid=6291
host=dynamic
context=default
defaultip=172.16.195.92
dtmf=info
canreinvite=no
And in extensions.conf, I set the incoming call's process
flow as below:
exten = 6291,1,Dial(SIP/6203SIP/6202,20,tr)
I set the outgoing call's process flow as below:
exten = _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60)
When I use 6291 pressing 06000 to make a call to ITSP user
6000, the call connected, but when I use other accounts on
Asterisk make calls to ITSP, The ITSP server rejected them,
because the caller IDs were not recognized. So, can I set
the caller account when I want make a call to ITSP?
Zhang Peihao
[EMAIL PROTECTED]
2004-01-15
Perhaps you're looking for SetCallerID?
-= Info about application 'SetCallerID' =-
[Synopsis]:
Set CallerID
[Description]:
SetCallerID(clid[|a]): Set Caller*ID on a call to a new
value. Sets ANI as well if a flag is used. Always returns 0
-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.
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Re: [Asterisk-Users] Major format changes
app_festival currently seems to chop the start of sound it plays back - probably something to do with rtp and maybe the same problem that was present in voicemail prompt plauback. Iain --On Thursday, January 15, 2004 11:16 am -0600 Steven Critchfield [EMAIL PROTECTED] wrote: On Thu, 2004-01-15 at 10:41, Robert Murray wrote: Hi Mark Would it be possible to include a way of streaming audio from memory? For example registering a file type which read from a fifo in memory? I need this for app_theta. (Cepstral TTS) I could copy the code from file.c, but I think it would be better if the same code could be used to avoid duplication. Whats wrong with just creating your frames and handing them off to be dealt with? Maybe you should check out the stuff in app_festival. On Sat, Jun 28, 2003 at 05:48:50PM -0500, Mark Spencer wrote: I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ 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] ultra-cheap (and easy) asterisk box
I think that it will be greate to include * inside of a router like ix66 from intertex... 1 GB usb removable flash to record voice mail.and prompts in the computer..2 fxo...real internal sip server ...internal dns server..good user interface.. all nat / firewall nightmare ended, no computers to worry about. Just dreaming with my little office pbx for about $200 regards Miklos - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 3:31 PM Subject: Re: [Asterisk-Users] ultra-cheap asterisk box I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. Ideally Asterisk will be ported one day to Linux/ARM or some other very low cost platform. for VOIP you do not need the PCI slots. In theory Asterisk could run on a Lynksys router box with re-flashed EEPROM. After all Lynksys' latest wireless router runs Linux inside Low cost to me means low total cost of ownership To get this I don't think buying the lowest priced parts is the way to go. I want quality mainboard, and a quality power supply and, this is importernt: A low internal case temperature. for this reason I'll spend the extra $50 to go with Antec cases and ASUS mainboards over the generic ones. What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] asterisk.org webpage
On 15/01/04 13:12, Roy Sigurd Karlsbakk wrote: hi all for new users, finding asterisk info is unneccesary troublesome. the asterisk.org page has very little information about the product and using google for 'asterisk' is like using google for 'linux'. you get all too many hits that has nothing to do with the product. perhaps the asterisk.org page at least should point to voip-info.org? or perhaps it's time someone rewrote the page? I can't say that I've found this a problem - it *does* point to voip-info.org http://asterisk.org/index.php?menu=support Perhaps support should be broken down into documentation and support to make this more obvious? Alastair ___ 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] t1xxp Unable to request IRQ
Hi All, I have a e100p that is not receiving any interrupts. My /proc/interrupts look like CPU0 0: 87288 XT-PIC timer 1:104 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 10: 814092 XT-PIC eth0, wcfxo 11: 0 XT-PIC t1xxp 12: 32 XT-PIC PS/2 Mouse 14: 4553 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 My dmesg gives the following output t1xxp: Unable to request IRQ 0 Any hint? TIA, PHM ___ 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] re hardware requirement - asterisk
fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 For fxp0, the internal interface although the nic can do full-duplex it seems to me that it is only running simplex!! Why do you think it is running simplex. I read the above and see where it says (100baseTX full-duplex) I don't think 10BaseT can run full duplex. I could be wrong but I don't think so. But why does it matter? A single VOIP connection will not even use 1% of a simplex 10BaseT. Simplex 100BaseT should be able to handle dozens and dozens of calls Just for fun, I moved our * box to a 10meg al-cheapo hub to force 10-half, placed a sip-to-sip call (via two C7960's) and noticed audio was very much half duplex. Very irritating to say the least (worse then most digital cell-to-cell calls). Then without changing anything other then moving the * interface to an upstream switch running 100 full (and verifying settings), the half-duplex-sounding audio effects completely disappeared (as expected). While both tests were being conducted, I ran a Sniffer analyzer to monitor packets and validate results. 10-half vs 10-full does have a substantial impact on quality. Moving from 10-full to 100-full would have no impact unless I could have loaded it with more rtp sessions then what I currently have the ability to do. And, FWIW, an interface set to half-duplex on one end with full-duplex on the other end was by far worse then when both ends of the cat 5 matched. All tests were conducted by forcing rtp traffic thru * (didn't allow the rtp to flow between the two sip phones). BTW, 10-Full setting is truly available on a large number of NICs, but not all. Obviously, the older stuff didn't support it, nor do the older Cisco 10 meg interfaces, etc. ___ 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] Disturbing trend of * production boxes that shouldn't be
It's spelled MCI, WorldCom, Sprint, T-Mobile... All the same except for the billing and the twists and turns of the contract. Whatever happened to POTS (i.e., Bell System.) An Old ATT 4A/ETS and ESS/S7 Craft == -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Burr Sent: Thursday, January 15, 2004 12:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Disturbing trend of * production boxes that shouldn't be how do you spell Teleecooomm again? [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Franczyk Sent: Thursday, January 15, 2004 10:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question Whaaa?? So, to allow 24+ lines of dial in access, how would I configure it? Would I need to purchase or lease a voice-over-ip box to connect our T1 or phone lines into? And then from there send the VOIP to the linux/Asterisk box for recording? (forgive me, Im new to telephony, but I need to make this work) :-) This is a disturbing trendpeople who don't know much about Asterisk and/or Linux and/or telephony who Need to make these things work or need to know how to update [their] production box installation...sorry I don't know Linux at all. Asterisk is great, but to maintain it and especially to repair things when they've gone wrong, you need to know what you're doing. Its your job (the people I'm talking about know who they are), so do as you wish, but I sure would install something I know little or nothing about and call it production. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] QoS anyone?
Has anyone played around with QoS or TOS relative to * and sip phones? I was just doing a little real-time research and noticed our C7960's mark IP packets with low delay and high throughput (presumably due to tos_media: 5 in the SIPDefault config file), and rtp packets flowing from asterisk back to the sip phone are not marked at all. Is there a * config parameter to enable such a function? Rich ___ 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] [OT] Commercial conferencing solution?
This is not directly * related, but could be. My company is using a VoIP conferencing solution that is suffering from developer neglect. I've considered trying to leverage *, and our internal developers can build the management interfaces. If that plan is not accepted by management, I need to find a commercial solution that works. Requirements: H.323 SIP G7.11/G7.29 Multiple unique conferences (conference ids passwords) enforcable limits per conference (max participants/time limits) Single access number Web managed to assign conferenc ids and passwords The only short term issue I see with * for this is we are standardized on platform where Digium cards are not an option, and ztdummy and zaprtc cannot be loaded (2.6 kernels). Any thoughts? Thanks, Dan ___ 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] asterisk.org webpage
Oooops... A little jump ahead. It asked for sign on etc... Got it now, mucho thanks and understanding my slow brain... :-} -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Mynatt Sent: Thursday, January 15, 2004 12:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk.org webpage I get 'Access Denied'... Can it be downloaded zip or tar ball? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Thursday, January 15, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk.org webpage Hi! for new users, finding asterisk info is unneccesary troublesome. the asterisk.org page has very little information about the product and using google for 'asterisk' is like using google for 'linux'. you get all too many hits that has nothing to do with the product. perhaps the asterisk.org page at least should point to voip-info.org? or perhaps it's time someone rewrote the page? Here's my little attempt to adjust the README file: http://bugs.digium.com/bug_view_page.php?bug_id=846 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 ___ 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] SIP Phones - Power over ethernet?
Are there any cheap SIP phones (like the Grandstream for example) that support power over ethernet? What is necessary to support SIP phones in a Cisco Call Manager environment? ___ 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] Hardware for Asterisk
I am real close to finalizing my hardware selection for my Asterisk test machine. I am going to use the following hardware: Dell 400SC w\Red Hat 9.0 1 - 4 Port TDM40B Card (FXS) 3 - Wildcard X100P Cards (FXO) Are there any known conflicts using this setup in this machine? I will be occupying all the PCI slots for this configuration. Also, is it worth the trouble to tie Asterisk into our present system which is a Panasonic D816 Hybrid System, or should I just dump our current Panasonic system all together? Thanks, Charles Alvis Internet Technology Group, Inc. Redmond, WA --- [This E-mail scanned for viruses by Virus Hunter at itechgroup.com] ___ 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] 100% of cpu in an out of the box *
are you running safe_asterisk ? If so try to modify safe_asterisk ... CONSOLE=yes to CONSOLE=no or if not list all the asteirsk threads 'ps -axum | grep asterisk' find the thread that takes the most CPU and connect with gdb gdb /usr/sbin/asterisk pid and do 'bt' and post the last few lines back ... Martin On Thu, 15 Jan 2004, Craig Waddington wrote: Me too :( 100% CPU. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of F.G.Testa Sent: 14 January 2004 20:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 100% of cpu in an out of the box * Hi all! I'm newbie, so here goes my situation: I have succefully compiled the cvs version as shown in asterisk website in some linux distros: Debian (2.4.22), Conectiva, Fedora Core 1 and in all of them, * starts and consumes all the cpu (on top). Does anybody know this issue? Thanks! Testa ___ 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] Credit Card Terminal
Sipura recommended disabling the echo cancellation on the SPA-2000 for modem pass-through. It does help although still not 100% success rate. Stephen -Original Message- From: Christopher J. Wolff [mailto:[EMAIL PROTECTED] Sent: Thursday, January 15, 2004 12:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Credit Card Terminal Hello, I have a Hypercom T7P swipe card terminal sitting on a dedicated G711ulaw port. The Hypercom operates at either 1200 or 2400bps. I get about a 50% success rate when I try to authorize cards. On this same G711ulaw port, I have a fax machine with a 100% success rate operating at 9600bps. Any suggestions on how to change *, ATA186, or SIPURA SPA-2000 to enhance the card terminals ability to process would be appreciated. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com ___ 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] Voicemail Sequence Bug?
I have a user, running CVS a/o 11/23/03, who has complained about phantom messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/asterisk/voicemail/default//Old directory, and to my surprise the messages are indeed misordered. I assume this is a bug, can't find anything on it on the bugtracker, but wonder if anyone else knows anything about this before I report it. . . Thanks. None of the files in question is empty, btw. B. msg.WAV msg0001.WAV msg0002.WAV msg0006.WAV msg0010.WAV msg.gsm msg0001.gsm msg0002.gsm msg0006.gsm msg0010.gsm msg.txt msg0001.txt msg0002.txt msg0006.txt msg0010.txt msg.wav msg0001.wav msg0002.wav msg0006.wav msg0010.wav ___ 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] Major format changes
That's why I stoped using app_festival and instead use the Festival text2wav program to generate a .WAV file and use app_playback to stream the audio to the user. On Thu, 2004-01-15 at 13:41, Iain Stevenson wrote: app_festival currently seems to chop the start of sound it plays back - probably something to do with rtp and maybe the same problem that was present in voicemail prompt plauback. Iain --On Thursday, January 15, 2004 11:16 am -0600 Steven Critchfield [EMAIL PROTECTED] wrote: On Thu, 2004-01-15 at 10:41, Robert Murray wrote: Hi Mark Would it be possible to include a way of streaming audio from memory? For example registering a file type which read from a fifo in memory? I need this for app_theta. (Cepstral TTS) I could copy the code from file.c, but I think it would be better if the same code could be used to avoid duplication. Whats wrong with just creating your frames and handing them off to be dealt with? Maybe you should check out the stuff in app_festival. On Sat, Jun 28, 2003 at 05:48:50PM -0500, Mark Spencer wrote: I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ 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 -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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] capacity testing
On 15/01/04 19:39, Jesse Peterson wrote: #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 Do you experience the same problems when you use the other (bundled) h323 driver? (asterisk/channels/h323/README for instructions) Alastair ___ 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] capacity testing
Hi, I am a newbie in Asterisk as well, intending to use it in a similar way as you are, communicating with AS5300 as well as other gateways including MAXTNT. I have had similar, but yet different experiences than yours. 1. Asterisk does crash with the number of calls, but in my case, about or less than 20 calls, then I would get either a Segmentation Error and then crashed OR it would just crash saying Disconnected from Asterisk server all of a sudden. 2. I am using Pentium Xeon chip and hence more powerful than yours with 512M RAM, my CPU usage has always been low, however, I have not had a chance to look at the CPU usage just before crashing, but all the time that I was looking, it has been low. Rather the MEMORY has always remained high at 450M usage even with no calls. This is a different experience as compared to yours. 3. I have also noticed that with more calls, and after a certain random period of time, any H323 calls going into the Asterisk would fail, my AS5300 and MAXT TNT would get their calls all rejected from Asterisk. However, Asterisk was still running at the time and I could actually call in and out the zap interface and outbound H323 from Asterisk was not a problem. It seems that something got hung with H323, causing inbound H323 calls into Asterisk to all fail. In this situation, I would have to stop the Asterisk and rerun it to fix the problem. 4. I have not tried the 0.7.0 version, but with existing version, I am not getting reliable and stable system, nothing close to Cisco and Lucent which are rock solid. However, I really love the power and the features of Asterisk, and I remain in good faith to see improvements. Any associate out there who can shed some lights into this? I am rather curious as to why I seem to be using up all memory although I am not running any unnecessary processes, or should I actually disable all modules, other than really necessary ones to support VOIP? Thanks ! Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jesse Peterson Sent: Thursday, January 15, 2004 2:40 PM To: Asterisk-Users (E-mail) Subject: [Asterisk-Users] capacity testing Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip-h323 gateway to tie SIP H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN-TNT-Asterisk-SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP-ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this: Run asterisk with -vvvcg. Do your test (core file generated). Run gdb /usr/sbin/asterisk core_filename From within gdb run bt and send me the output of it. if it is of use, here it is (from asterisk v.0.5.0) - (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at
[Asterisk-Users] re: hardware requirement -asterisk
Referring to my previous post about degradation of voice quality when having more than 2 connection. The actual route is: pc xlite - local asterisk box - iaxtel - local asterisk I have tried out a different situation: pc xlite - local asterisk box - iaxtel and the second connection pc xlite - local asterisk box - iaxtel - local asterisk The same degradation happens as soon as the second connection is connected. I am suspecting the ADSL connection. The internet part is ADSL with 512k down and 128k UP. The nic is a 3c905c 100baseTX and connected to a NEC ADSL modem. # ifconfig xl0 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 But ifconfig seems to suggest that it is running in simplex mode. Is the degradation a result of the ADSL connection? David Kwok smime.p7s Description: S/MIME Cryptographic Signature
