Re: [asterisk-users] OPENR2 in Thailand
I got Asterisk to work with R2 in Thailand about 3 years ago. This was back before OpenR2, and I had to modify the C libraries directly. Of course, outbound dialing is DTMF, not R2, however inbound is pretty standard. This was on TA (now True) circuits. I guess I can't tell you much about how to set it up today as my experience is so out of date, but I can offer you encouragement that it will work for you if you put in the effort. There is nothing fundamental which will cause it to fail. Which carrier are you planning on connecting to? Chris Moises Silva wrote: Hello Peter, You can ask this better in the asterisk-r2 mailing list. I don't know of anyone that has used OpenR2 in Thailand, but I am interested in adding support for that variant. Contact me at this same e-mail address or via Google talk (my e-mail address works for MSN as well ) to discuss further details. Moisés Silva On Mon, Oct 20, 2008 at 12:58 PM, Peter Lindquist [EMAIL PROTECTED] wrote: Dear All, I'm looking for someone who has implemented OpenR2 in Thailand successfully. Any settings, advice, caveats etc. are welcome. Best regards, Peter Lindqvist www.voxion.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPENR2 in Thailand
Hi Peter, Basically, I had to rewrite the R2 state machine to use pulsed outbound signalling instead of compelled, and hack in support of the DTMF tone groups instead of R2 frequencies. It was messy, but possible. The inbound side was much cleaner, and you may be correct when you say you could just use China R2 for that. Is this application going to be limited to inbound, or do you need outbound dialing? The originating switch may also make a difference. Just be prepared for that. TOT is switching some of their stuff over to Huawei, which in my experience causes issues (however, no idea about their R2 support in general). We were connecting to a Siemens EWSD out of Prakanong, and it proved pretty reliable. Not much variation at all. This application was used for a time to do call ins for the 07 show, so we were doing very heavy signalling for about 1 hour every week. TA had configured a 2:1 ratio for us on MF senders on the incoming side. Average call duration was only 30 seconds, and 120 lines were packed for 1 hour straight. I assume you've already asked the Turtlephone Organization of Thailand about ISDN? Have you asked more than once? Often times they'll just say no we don't support it because they're lazy or don't know, not because it isn't possible. If you haven't actually had a sit down with an engineer in their offices, I wouldn't necessarily believe them if the salesrep tells you they can't do it. A bottle of Johnny Walker as a gift can often make these discussions go easier. Chris Peter Lindquist wrote: We are using TOT (Telephone Organization of Thailand). They are very messy on their side so we are sorting out some unrelated problems with them right now - very slow response. I believe you are correct when you say that outbound dialing is DTMF, I have heard this before too. Out of curiosity what did you have to change in the C libraries? Peter Chris Ziomkowski wrote: I got Asterisk to work with R2 in Thailand about 3 years ago. This was back before OpenR2, and I had to modify the C libraries directly. Of course, outbound dialing is DTMF, not R2, however inbound is pretty standard. This was on TA (now True) circuits. I guess I can't tell you much about how to set it up today as my experience is so out of date, but I can offer you encouragement that it will work for you if you put in the effort. There is nothing fundamental which will cause it to fail. Which carrier are you planning on connecting to? Chris Moises Silva wrote: Hello Peter, You can ask this better in the asterisk-r2 mailing list. I don't know of anyone that has used OpenR2 in Thailand, but I am interested in adding support for that variant. Contact me at this same e-mail address or via Google talk (my e-mail address works for MSN as well ) to discuss further details. Moisés Silva On Mon, Oct 20, 2008 at 12:58 PM, Peter Lindquist [EMAIL PROTECTED] wrote: Dear All, I'm looking for someone who has implemented OpenR2 in Thailand successfully. Any settings, advice, caveats etc. are welcome. Best regards, Peter Lindqvist www.voxion.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Nextone using H323
The meaning of softswitch in the Nextone is that it will try and use whatever protocol the corresponding Ingress/Egress side is set for. That means, if you place/receive a call from/to Asterisk, and route to an endpoint doing SIP, the Nextone will expect Asterisk to speak SIP. If you place/receive a call to/from an endpoint using H.323, it will expect Asterisk to speak H.323. When you use softswitch, the Nextone is much more forgiving about what it passes through also, expecting that your softswitch on the far side will take care of the issues. If you only want to use H.323 with Asterisk, you should configure it as an H.323 gateway. Why are you trying to set softswitch? That is how all of our systems are configured with Asterisk and ooh323. Works very well and very stable. Chris Everton Goularth wrote: Hi people, Someone have already used asterisk with Nextone? I`m trying to use it, but there are some problems.. One of these are when we set up a connection between Nextone and Asterisk using H323, we use our asterisk server as a Softswitch in the Nextone configuration, so it doesn`t work. But, when we just change (in Nextone configuration) from Softswitch to Gateway, it work. Where is the difference? I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf: - [general] ;Default - 1720 ;port=1720 bindaddr= IP_ADDRESS ;This parameter indicates whether channel driver should register with ;gatekeeper as a gateway or an endpoint. ;Default - no ;gateway=yes ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes ;faststart=no ;h245tunneling=no ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=GW2 e164=1521# ;CallerID to use for calls ;Default - Same as h323id callerid=MediaXChange 1.0 ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d ;gatekeeper = 189.44.163.125 logfile=/var/log/asterisk/h323_log context=default disallow=all allow=g729 allow=g723 dtmfmode=rfc2833 - Anyone know what can I do? or what am I doing wrong??? Thank`s a lot for the opportunity. Everton Goularth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to modify incoming DNIS?
Hello everyone,
I have a question that seems simple, but I am stumped. Basically, I have
several incoming SIP trunk from gateways connected to physical E1 lines
around the world. Everything works, but the DNIS that comes in are non
standard and sometimes conflict. It varies not only by country, but also
by carrier within a country and even by switch within a carrier. I want
the complete E.164 number, but I rarely get this.
What I would like to do is modify the incoming number as soon as it
enters Asterisk. What happens now is that DNIS 12345678 from SIP trunk A
maps to the same extension as 12345678 from SIP trunk B, even though one
number is actually from Europe and the other from Asia. This causes
many, many conflicts. I want to remap the incoming DNIS to 4412345678
and 6012345678.
Unfortunately, the simple solution doesn't seem to work. The following
fails
(CALLERID(num) is updated correctly, but EXTEN never gets changed.)
[from-sip-trunk-A]
exten = _1234,1,Set(EXTEN=44${EXTEN})
exten = _1234,2,Set(CALLERID(num)=44${CALLERID(num)})
exten = _1234,3,Goto(s,1)
exten = s,1,Goto(incoming-call,${EXTEN},1)
[from-sip-trunk-B]
exten = _1234,1,Set(EXTEN=60${EXTEN})
exten = _1234,2,Set(CALLERID(num)=60${CALLERID(num)})
exten = _1234,2,Goto(s,1)
exten = s,1,Goto(incoming-call,${EXTEN},1)
No error messages are generated, but the EXTEN variable is never
updated! Right after changing it, I print it out with a NoOP and the
value is not updated.
The incoming-call context makes extensive use of macros and other
applications which assume EXTEN will be set correctly. How can I force
EXTEN to be set to the value I want instead of the value that arrives on
the trunk?
Please help. This sounds so simple but I am now very frustrated. The
idea of rewriting every application in the system because EXTEN can't be
changed is very unappealing. I'm already 4 days late getting this up and
running.
Thank you for any assistance,
Chris
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[Asterisk-Users] G.723.1, pass thru and DTMF. Possible?
I am investigating the use of Asterisk for a new project and am confused about all the literature available on G.723.1 in pass thru mode. Specifically, I need to be able to take 2 H.323 channels, each running a G.723.1 codec, and bridge them together. However, before I do that, I need to play a message and then listen to one of the channels to determine how to route the call. For example, it may play a menu asking the user to select one of technical support, sales, or accounting. Or it might ask them to press 1 for Mandarin and route the call to Singapore, 2 for Khmer and and route to Phenom Phen, etc. Since I can program the gateways which will be interfacing to the PSTN to send the DTMF tones via H.245 out of band, there should be no technical reason why this won't work...in other words, I never have to actually decode the G.723.1 stream. The messages can be stored in G.723,1 format already, so I never have to encode either. However, I have not been able to discern whether Asterisk will work in this mode or not. Can someone who has actually implemented an Asterisk system using G.723.1 in pass thru enlighten me? The documentation on this is very confusing. Will it use the out of band H.245 messages to detect a DTMF tone? Since there is no technical reason it should not work, if the answer is it doesn't work can someone give me a hint on what must be changed to make it functional? I would certainly be willing to put in a little effort to make this functional if it isn't already and someone can give me some instructions on where to begin. Thanks in advance for any assistance, Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Fw: Asterisk R2 Signaling
I did quite a bit of work on this library about a year ago. In the end, I abandoned it because the project went belly up. I can tell you this much MFC-R2 for Argentina is the default configuration and looked fairly well supported in the source code. I, of course. couldn't check it because I'm not in Argentina, but my guess is it would probably work with only minor if any modifications. The big drawback however is that it only works for incoming calls. If you want to do outbound, forget it. I had to write the outbound dialing routines from scratch. So, if you're looking for inbound IVR type stuff and you are in Argentina, I would recommend getting the MFC library from Mark. It is available but not advertised in the CVS tree. He put it in there for me last year. You can probably make it work with not a lot of effort, but it does take some coaxing to get it to compile. I wouldn't try it without having someone around who is a pretty good developer to help you through the rough spots. Good luck and let us know how this turns out for you. Chris Ziomkowski At 06:28 PM 9/17/2004 +0200, you wrote: +++ Tenorio, Leandro [15/09/04 10:58 -0300]: I've seen a lot of times, people that try to get R2 MFC to *, most of them trying to use Dialogic Boards (BTW They 're Very expensive), none of them where succesfully, If you want to use PCI Cards on your server, why don?t u ask to your carrier to provide you E1/PRI? or better put a Gateway with E1s and SIP. I'm in Argentina also and I get PRI E1s from some of the Carriers I'm from India and I'm trying to look for info abt E1 R2 and asterisk too. What do u mean by put a gateway with E1's and SIP ? I dont know if any providers provide E1/PRI who they need any extra hardware for that ? -- regards Vikram (http://www.vicramresearch.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
Re: [Asterisk-Users] Does IAX pass ISDN result codes?
At 08:34 PM 11/4/2003 -0600, you wrote: Quoting Chris Ziomkowski [EMAIL PROTECTED]: I can't try this setup yet (still don't have the hardware), and have been trying to answer this question merely from the source code. So far, I have not been able to convince myself. Does anyone have definitive information on this? Passing q931 disconnect causes doesnt work on iax. rgrds m. Lovely. Yet another thing I'll have to hack up before I can get this project off the ground. It's a good thing asterisk is open source. Thanks for the information Martin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does IAX pass ISDN result codes?
Have a quick question. In the following setup: PC w/ E100P running EuroISDN = IAX = PC w/ E100P running EuroISDN Calls come in on the first PC and are terminated out to the PSTN on the second. Do the ISDN cause codes get transferred correctly in this situation? I can't try this setup yet (still don't have the hardware), and have been trying to answer this question merely from the source code. So far, I have not been able to convince myself. Does anyone have definitive information on this? Thanks for any assistance, Chris Ziomkowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Absolute Minimum Installation Packages
At 10:24 AM 10/31/2003 -0600, you wrote: I can understand the size concerns for putting it in an appliance or what-not. However, my opinion is that, due to the low cost of hard disk space, it is cheaper for the company to go out and buy another hard disk to replace the extra 500 MB they wasted on a sub-optimal installation than to pay me to try to get the installation as small as possible. What are the benefits to a really tiny installation, aside from possible appliance applications? Moreover, won't you still need a sizable hard disk for voice prompts, voicemail messages, sound file to direct people to dial the correct extension, etc? Again, I may be WAY off track, but one of the things I really like about * is that I can update it easily. Wouldn't you lose some of the beauty by putting it in an appliance? Moreover, I HATE Nortel because they have a user-unfriendly interface, proprietary controls, non-standard connections, and the like. It seems to me that by appliance-izing we would be inviting the same abuses that the current systems enjoy. I could see it becoming an issue of open-source software on extremely proprietary hardware, meaning the user can modify their system if they can figure out how to get in it. Of course, all of this is in the assumption that the end-user wants to own their PBX. I know I do. I think that we should be focusing on a useful administrative interface, database-based extension definitions, and other features that will advance the power, flexibility, and usability of * instead of shrinking the distro as much as possible. What am I missing? I see many people much smarter than I am excited about this, so I am sure I simply failed to consider how it will revolutionize everything. Awaiting your enlightenment (preferably sans-flame), David Gomillion 2 things. 1) Your time might be alot cheaper than memory to a company who wanted to sell a product on this. $10 times 10,000 units is $100K. Just how valuable do you think your time is? It all depends on volume. 2) The obvious reason not to use a hard disk is reliability. If you are looking at mission critical applications, the hard disk in a PC is the most likely thing to fail. So you can either invest in an expensive RAID system to mitigate this problem, or else go to silicon storage devices. If you could get asterisk+linux down to 150 MB or so, then there is a cost effective way of compressing this system onto a 512 Mbit flash. Still can be a PC, just booting from flash instead of failure prone disk drives. If you want voice prompts, database, etc. do it on a PC externally via network storage. I myself am very interested to hear how small one could make this. I think there is a entire segment of unexplored markets for an embedded asterisk system. Chris Ziomkowski -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Friday, October 31, 2003 8:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Absolute Minimum Installation Packages Hi. Il ven, 2003-10-31 alle 02:31, JR Richardson ha scritto: I'm trying to get the total Linux/* installation size as small as possible. I'm wondering if anyone has looked at the installed packages list from the Redhat installation [rpm -qa] and has parsed out all packages not needed for * to run. I follow the custom install guide from Andy Powell but the installation yields 948+ Meg with 340 installed packages. I'm sure most of those packages can be eliminated. I have a very little RedHat 9.0 installation that's about 504 MB, with asterisk+sounds+some voicemail installed. Also devel tools installed. Also apache+mysql+db, since we have many things of asterisk moved onto a mysql db... stripping away devel tools, I can manage a to have 450Mb RedHat 9.0. surely big, but very small to be a RH ;) -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection
Actually, Back in '99, Dialogic used a very simple algorithm, and it was surprisingly accurate. You simply watch and see how long the initial greeting is. If it is short (say, only a few seconds), then it is generally a live person. However, if the initial greeting lasts for much longer (say 20 seconds) then you have contacted an answering machine. That is one of the big reasons CPA on Dialogic used to give so many headaches on drop and insert applications. It would sometimes wait 10 seconds before returning answer supervision to the application and the talk path would be cut through (Had to wait to determine whether it was a human or an answering machine). In this time, if a human answered, he would sometimes hangup because he wouldn't hear any response from the remote side. Properly tuned, just watching how many seconds of energy you get in the initial greeting before silence sets in will give you 90% accuracy in determining answering machine or live person. There are always exceptions however. As a first guess though, you can assume anything less than 5-10 seconds is human, anything greater is a machine. Lots of ways to get it wrong though. Not recognizing a SIT tone and returning answering machine for circuit failure, not recognizing when ringing has ended and misinterpreting the hellohello as still being ringing cadence (Dialogic did this about 3% of the time). But in theory it should be trivial to implement in Asterisk. Might want to write a new energy detector algorithm in dsp.c though based on a wideband/low Q resonator approach (move the pole way in towards the origin) as opposed to narrow band goertzels (pole on the unit circle). More robust for this type of work. Chris At 08:24 PM 10/29/2003 +0800, you wrote: Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. How very inefficient. Looking for peaks in the autocorrelation function requires much less compute. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Right. Dialogic and others make a big fuss of the super detection algorithms, and quote 90+% accuracy. In the real world they are utterly useless. Call answering just doesn't fall into a sufficient redular patterm. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Machine Detection
At 09:17 AM 10/29/2003 -0500, you wrote: Might want to write a new energy detector algorithm in dsp.c though based on a wideband/low Q resonator approach (move the pole way in towards the origin) as opposed to narrow band goertzels (pole on the unit circle). More robust for this type of work. Where does one go to learn this terminology and the math to implement it? Caltech. Take PP's class. Just don't fail. Seriously, any introductory book on filter design. When you design a filter, you write out the transfer function. zeroes in the numerator are called zeroes. Zeroes in the denominator are called poles. A filter will pass frequencies near its poles. It will suppress frequencies near its zeroes. As you move poles out towards the unit circle, the filter becomes less stable. You can imagine a hitting a filter with a burst at a single frequency. A filter with poles near the origin will die off very quickly. As you move towards the unit circle, the filter will ring longer and longer. On the unit circle there is no dampening at all. Outside of the unit circle, the filter becomes divergent. The order of the filter is simply the number of zeroes. Goertzel (named after the guy who invented them) filters have their poles on the unit circle. Asterisk uses these to recognize DTMF and other tones. They are easy to calculate mathematically, but they have terrible properties if you need a wide frequency response over a sustained period. A resonator is essentially the same filter but moved inside the unit circle so that it will relax over time. It takes a few more operations to calculate, but you have alot more control over the final shape. If you are interested, I would suggest getting a book on the subject. Just go to Amazon and find one you think meets your requirements. There are several. Any book on Digital Signal Processing will have at least a few chapters on filter design. Pay specific attention to the chapters on IIR filters. They will teach you about poles, zeroes and stability. I would also recommend this tutorial on the web if you're interested: http://www.dewtronics.com/tutorial.html Chris -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection
At 11:52 PM 10/29/2003 +0800, you wrote: Ray Burkholder wrote: Might want to write a new energy detector algorithm in dsp.c though based on a wideband/low Q resonator approach (move the pole way in towards the origin) as opposed to narrow band goertzels (pole on the unit circle). More robust for this type of work. Where does one go to learn this terminology and the math to implement it? Apparantly not to this source. :-) A Goertzel filter finds one output bin of a DFT. Since the width of the bins in a DFT is directly related to the number of samples you include in a processed block, the width of the Goertzel is too. A Geortzel is as wide or as narrow as you want it to be. Oh, and k does not need to be an integer, unless you are trying to evaluate phase. That is a common misconception. There is a sliding window version of a Goertzel filter, but this has the same characteristics as the standard version, as it is just a trick for calculating a continuous stream of Goertzels efficiently. Now, now, Steve. Let's not get personal. You are correct. I am also correct. It depends on how you look at it. A Goerzel is only as wide as you want it to be if you have short periods and/or infinite representation. Goertzels are on the unit circle, and as such on the edge of stability. They never dampen their responses, so they will grow infinitely large if you keep ringing them. Try using them on a 16 bit fixed point DSP and you'll understand how important this point really is. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oh323 segmentation fault in asterisk...
This looks like the same case as the thread: Problem ATA-711-723-Oh323-Asterisk that was posted back in August. Although, given the nature of trampled memory problems it could be almost anything. The Oh323 library dies in a call to realloc. This is almost certainly indicative of someone stomping on memory somewhere previously, or else a memory bug in the pwlib library itself. (Which I find hard to believe given the several years its been around.) My suspicion is an out of bounds memory access previously. Here is the trace: (gdb) where #0 0x4207b2ef in chunk_realloc () from /lib/i686/libc.so.6 #1 0x4207b1f8 in realloc () from /lib/i686/libc.so.6 #2 0x489d2176 in PAbstractArray::SetSize () from /usr/local/lib/libpt_linux_x86_r.so.1.5.2 #3 0x489d1a3d in PContainer::SetMinSize () from /usr/local/lib/libpt_linux_x86_r.so.1.5.2 #4 0x483eedd7 in RTP_DataFrame::SetPayloadSize () from /usr/local/lib/libh323_linux_x86_r.so.1.12.2 #5 0x483d73c9 in H323_RTPChannel::Transmit () from /usr/local/lib/libh323_linux_x86_r.so.1.12.2 #6 0x483d497c in H323LogicalChannelThread::Main () from /usr/local/lib/libh323_linux_x86_r.so.1.12.2 #7 0x489ca634 in PThread::PX_ThreadStart () from /usr/local/lib/libpt_linux_x86_r.so.1.5.2 #8 0x4003bfef in pthread_start_thread () from /lib/i686/libpthread.so.0 #9 0x4003c0df in pthread_start_thread_event () from /lib/i686/libpthread.so.0 This happens the instant the call connects to Asterisk, exactly as the previous poster back in August described. Tried various frame sizes, nothing seems to help. G.711 is the only codec in common between the 2 systems. (Quintum A400 Tenor and Asterisk). Does anyone know if this previous issue was ever resolved, and if so, how? I have to make this work, and tracking down a bad memory access in unfamiliar code is a task so unappealing that I'd place it right behind pulling off my fingernails and pouring salt over the open wounds. (Not to mention I am not a C++ person at all, so digging through the Oh323 code will prove a formidable challenge.) Are other people having success with Oh323 and G.711? If so, can anyone send me the oh323.conf that they are using? Perhaps it is a combination of flags that causes the issue. Otherwise, can anyone suggest a place to start/offer assistance? Thanks, Chris Ziomkowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best performing CPU for G.729 codec?
Hi everyone, I need to build a machine capable of running at least 30 G.729 channels with lots of room to spare because it will be doing some other CPU intensive tasks also. I've seen Mark's post about being able to run 60 channels on a dual 1.8 Xeon, but that unfortunately raises more questions than it answers. You see, I may have to build many of these systems, and I need to find a good price/performance ratio, I can't simply go out and buy quad opteron systems for fun. I was trying to find a processor for under $200 to handle this. Given that, my question is: what constrains the G.729 codec? Is it mostly due to branches (implying a dual CPU or P4 HT architecture will be required) or is it mostly due to floating point (in which case a lower speed AMD might be able to accomplish the same task cheaper and better.) Anyone have any ideas on this? Looking for feedback. BTW, can someone answer how the G.729 licensing works? If I change the CPU and motherboard (probably necessitating a kernel change), will my licenses still be valid? What exactly are they keyed to? Thanks for input on this topic. Chris Ziomkowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best performing CPU for G.729 codec?
At 09:19 AM 10/22/2003 -0700, you wrote: I hate to say it, but jumpping off into a 100 channel PBX is not the way to go with Asterisk. Build a 1x1 PBX first on an old Pentium 500. get this to work then try adding SIP phones then add some other features. After you've spent some time you will not need to ask the above. In fact I'd say if you need to ask Guys, Thanks for all the helpful advice, but I think we're getting off topic. I have specific engineering requirements I have to meet. I can not change my problem space. I need a 30 user G.729 (not 100's) plus custom apps, single box solution. No buts, and I need to make it as cheap as possible. This is for Thailand...you can spend $100 on an overpowered computer or you can feed your family for a month. (No joke!) Price mattersalot. I don't want it to scale. My problem space is fixed. I don't need to build a general purpose PBX. What I really want to know is, what are the constraints of the G.729 codec? Does anyone have any real world experience implementing this, and is the resource utilization a result of mathematical floating point calculations, or a result of branches and sequential processing instructions? Can anyone give me some solid advice here on how resources are being used? What you've taught me so far is: A) you believe that MP or HT will help the performance. Problem is, these are expensive solutions. Can you quantify this answer at all? How much will it help, and would it help more than additional floating point performance? B) The G.729 codec is linked to the hardware, so I can't even experiment without having to pay for it each time. Question...is there anyway to transfer the G.729 licenses from one system to another when I don't want them anymore? I need to answer this questions, and paying $300 everytime I need to check out a new architecture (and worse than that waiting for 24 hours) simply isn't practical. Does anyone have contact info for an engineer at Voiceage who can help answer my questions? Is there a multi channel developers license available that would transfer across different hardware architectures? Thanks for your feedback, Chris Ziomkowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
