Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!
C F wrote: You could, under programming section 1.3.4 in the http interface to configure the GW card enable DTMF Detection, that will enable Out of Band DTMF. In the TDE they renamed this to DTMF signalling. Believe me, I spent a great deal of time on this including Ethereal captures and nothing worked. Have you succeeded with this? If so, what DTMF protocols were passed? Regards, Richard ___ -- 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] Panasonic x Asterisk ... NO PROBLEM!
Jonn R Taylor wrote: Install a T1 between the Panasonic and Asterisk and program the T1 in the Panasonic as a other custom PBX. VOIP card would be the best. Jonn One thing to beware of with the Panasonic VoIP card, is that I have found no way of getting it to pass out of band DTMF, possibly because it handles this in a proprietary way. This has been my experience with a TDA100 and VoIP card. Regards, Richard ___ -- 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] How to turn on the H323 logging on Asterisk
Sema Arca wrote: Hi Richard, I could not succeed to make my ooh323 work somehow. I can see the peers and the users but although my exten definition states that the call should be forwarded to a GK, Asterisk does not send it out. I also have the same problem with registration. Do you think you can give me some ideas? Maybe send your conf as a reference? I am sorry I have no experience using ooh323 with a gatekeeper. My setup is as an endpoint between asterisk and a Panasonic TDM100 PBX. Regards, Richard ___ -- 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] How to turn on the H323 logging on Asterisk
Sema Arca wrote: Can you still send the config files? Maybe I can come up with an idea? :( extensions.conf entry exten = _1XX,1,Dial(OOH323/[EMAIL PROTECTED]) exten = _1XX,2,Congestion ooh323.conf [general] h323id=ObjSysAsterisk e164=100 callerid=asterisk context=default tos=lowdelay disallow=all allow=alaw dtmfmode=inband [Panasonic] type=friend context=default ip=192.168.0.2 port=1720 disallow=all allow=alaw Regards, Richard ___ -- 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] How to turn on the H323 logging on Asterisk
Tony Mountifield wrote: I don't know whether Objective Systems have abandoned chan_ooh323 and the ooh323c stack, but it would be great to see them moved from -addons into the main Asterisk tree. This was always the plan from the beginning. I have a post from one of the Objective Systems developers: We are working closely with digium towards making it an official channel driver once the testing phase is complete. I'm not sure what happened subsequently - perhaps licencing issues, but I agree, it is an easy to build implementation that I have had in light production use for 3 years. Regards, Richard ___ -- 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
[asterisk-users] SIP - ooh323 Bridging
Hi, I have the following setup, with asterisk on a dual homed box: PolyIP500(SIP)--192.168.4.0--Asterisk--192.168.0.0--Panasonic(H323) It is running a recent SVN version of Asterisk 1.2 and ooh323. The problem I have, is that despite having canreinvite=no in the sip.conf, asterisk still insists in attempting to native bridge the RTP streams: -- Executing Dial(OOH323/192.168.0.2-540d, SIP/polywn1) in new stack -- Called polywn1 -- SIP/polywn1-0817e828 is ringing -- SIP/polywn1-0817e828 answered OOH323/192.168.0.2-540d -- Attempting native bridge of OOH323/192.168.0.2-540d and SIP/polywn1-0817e828 This results in audio only running in the H323 to SIP direction - nothing the other way. There is NO NAT involved in the asterisk box. Investigation with Wireshark shows that as the call is setup, a couple of packets worth of RTP audio does flow in the SIP-H323 direction, until the native bridge occurs, at which point it fails. I realise that this may be something that is fixed in 1.4, but that is not an option at this stage. Can anyone offer a way of forcing asterisk to stay in the path and not be bridged out? Regards, Richard ___ --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] SIP - ooh323 Bridging
Dovid B wrote: I doubt this will help but try also nat=yes. Thanks for the reply but no, I have spent some time trying quite a number of variations of NAT related configuration changes in various places. I don't know why it does not honour the canreinvite=no entry. Regards, Richard ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
Steve Totaro wrote: I guess I am just lucky to have 24 hour manned data centers with staff that walk around looking for flashing LEDs. I am sure there is some error thrown in /var/log/messages about a failure that could be used to trigger a notification quite trivially. Both smartd and mdadm can be configured to send emails. Regards, Richard ___ --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] The High Performance Echo Canceller (HPEC)
Can someone comment why only Digium cards still under warranty are eligible to use this EC at no cost, versus older cards? Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic Hybrid Integration Advice Needed
C F wrote: Which panasonic system? I'm assuming you are talking about the TDA line. If so get a IP Gateway card on the TDA system, that card uses h323, then configure it with asterisk as h323, or my favorite, get a PRI card on the TDA sysem (unless it's a TDA50 then the option is not available), and a T1 card on asterisk, and create a dialplan on the Panasonic that goes out over the PRI card. Speaking as one who has been using the 16 channel IP-GW card with a TDA 100 for over a year, you might want to go with the PRI/T1 solution, although I have no experience with it. Although H323 integration with Asterisk has been flawless, I have had no luck with any form of out of band DTMF transport through the IP-GW. The documentation for the card states that it can do it, but does not state what protocol is used and packet captures show no sign of it. If anyone knows anything more, I would like to find out, as the local Panasonic agents have not been much help. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune unable to set impedence
Yuan LIU wrote: I just didn't want to accept fxotune.c's claim about working only with TDM. Several other users indicated that they were not able to tune X100P. There's also a README.debian note that specifically indicated exclusion of X100P. fxotune is written to change register values on a specific Silicon Labs chip, which is used in the TDM400 FXO modules. No X100P uses this chip, (and the chips they use do not have the feature used), so fxotune does nothing. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Motherboard 3.3V PCI for TE412P
Jesus Mogollon wrote: Hi all Does anyone know of any motherboards with PCI slots that can take the TE412P card? Is there such a MB for Athlon 64 or P4 procs? I have no experience of it, but you could look at the Asus M2N32 WS which has 2 x PCI-X (3.3V) slots. It is a socket AM2 (Athlon64) board. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Dave Fullerton wrote: I just verified it here as well. Running Asterisk 1.2.11 and two polycom I'll throw in a me too here, with the addition that it also occurs with canreinvite=no. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ooh323c - cdr
antonio wrote: I have a problem: when i make i call from a device h323 to sip, i have no cdr, and i don't see cdr variables for the channnel ooh323. Anyone can help me ?? Thanx On my system, this lives in /var/log/asterisk/cdr-csv/ast_h323.csv. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ooh323 svn updated
For those enquiring last week about ooh323 not compiling with the svn version of asterisk, the module loader changes have just been checked into the svn version of asterisk-addons and so should now work with svn asterisk. Have not yet tested this. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Addon-ooh323 install problem
Martin Joseph wrote: Do you just mean the tar balls of 1.2.9 and latest addon? Yes. I believe the svn addons package will be updated soon. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Addon-ooh323 install problem
Tetsuya Yamamoto wrote: I can't makel asterisk addon, asterisk-ooh323. I use Asterisk and addons svn version. The current svn version of asterisk has had the module loader code redesigned and to date, the svn addons have not been updated to match this change. You will need to use the latest production versions of both. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CentOS 4.x and ooh323
Bruce Reeves wrote: I'm trying to add ooh323c to my asterisk 1.2.7.1 http://1.2.7.1 install and did an svn update of asterisk-addons and followed the readme in asterisk-ooh323c and I get through the .configure with no errors. But make causes: rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0 -lpthread make: rpath: Command not found make: [libchan_h323.la] Error 127 (ignored) My previous mail mentioned that this had been posted recently on the list, however I was confusing it with the ooh323 list on Sourceforge. Unfortunately Sourceforge seems to be in the middle of a meltdown with CVS access to projects broken and for over a week, no list mail is being archived there, and I have deleted the mails I received relevent to this. So you may want to try on that list. This was the first post about it before archiving stopped. http://sourceforge.net/mailarchive/forum.php?thread_id=10291962forum_id=43045 Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CentOS 4.x and ooh323
Bruce Reeves wrote: I'm trying to add ooh323c to my asterisk 1.2.7.1 http://1.2.7.1 install and did an svn update of asterisk-addons and followed the readme in asterisk-ooh323c and I get through the .configure with no errors. But make causes: rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0 -lpthread make: rpath: Command not found make: [libchan_h323.la] Error 127 (ignored) I'm not real sure what to try to fix this. This came up over the last couple of weeks on the list and I think it is due to an autotools bug. Check back through the archive for a response by Anvin Patel. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 calls will not stay connected
Daren J. Howell DTCommunication wrote: I have restricted the asterisk server to G711 to match the choice on the PBX, and still same result. I have read that either endpoint have to be either a master or slave to communicate to each other. I see in the logs that both are shown to be a slave. The pbx side has to be set to slave. How can I lock the asterisk side to be a master? Or is this something to worry about? Hi Daren, I believe the endpoints negotiate the master slave thing, so I'm not sure this is the issue here. I had the exact same problem when I set up and it was caused by a codec mismatch, but I'm sure there are other factors that will give the same result. Sorry I can't offer any more. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 calls will not stay connected
Daren J. Howell DTCommunication wrote: Have Asterisk connected to a H323 compatible legacy PBX using QSIG protocol and IP trunks. I can call to Asterisk, and from Asterisk using X-Lite softphone but whenever either end picks up, the calls disconnects. Try restricting both ends to one codec; disallow=all allow=codec of choice at the asterisk end and whatever you need to do at the legacy end. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI voltage
Giordano Grandis wrote: Hi all, I have to bought a PCI with 4 PRI but on digium site I saw that there a re two different kind (3,3V and 5v). What’s the difference? 33MHz 32 bit PCI slots are 5V. PCI-X slots MAY support 5V and 3.3V depending on the age of the board. My understanding is that current PCI-X boards are 3.3V only now. The board you mention would appear to have PCI 5V slots. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compare to Skype
Eric ManxPower Wieling wrote: There are 2 issues here. 1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to transport audio for SIP (and other protocols). This means that ANY jitter on the SIP Phone - Asterisk link will cause audio problems. This is only an issue if your SIP phone has a poor/nonexistent jitter buffer. The ideal scenario from a latency point of view is for the end points to handle jitter buffering. I use Polycom 500's with G711 over a path where jitter can be quite severe on occasion and they handle it very well. Although I have not tried them, one would expect Cisco's to work well also. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ooh323 Gatekeeper Bug
Kenige Ho wrote: the ooh323 is from Asterisk-addon-1.2.1. Is there a bug on this version for the ooh323 and also how can i get the newer version of the ooh323(0.8.1) to compile with? Many thanks to you all. You will find 0.8.X in the asterisk-addons svn branch. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Performance differences 64-bit vs 32-bit
Morgan Gilroy wrote: As far as I know there will be no difference. 32bit runs natively on AMD64 chips. The only advantage of 64bit is the extra address space and huge integers :) But I could be wrong, iv not done any benchmarking myself just what i have read on the net. I have no idea if Asterisk gains a significant advantage on 64bits v 32, but thought I would respond to the commonly perpetuated statement that, The only advantage of 64bit is the extra address space and huge integers The x86 Instruction Set architecture allows for 8 general purpose registers. The x86_64 ISA allows for 16, somewhat more flexible ones. The potential here is for software that can effectively utilise the extra ones, will gain some benefit. See http://www.techreport.com/reviews/2005q1/64-bits/index.x?pg=2 for more information. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Dell blade servers
Mike Fedyk wrote: Matt Riddell wrote: I would instead recommend the SuperMicro 1U servers - we have had a really great run with these. Do you use Opteron or Intel? I would not suggest that Supermicro are in Intel's pocket, so they must have had their fingers in their ears going, Laa..Laa..Laa..Laa..., when the AMD guys came round with benchmarks of their current hardware... Supermicro do not do Opteron (or Athlon64) systems. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Dell blade servers
Dean Collins wrote: Lol, so Dell must be doing the same thing. Did you ever consider that Supermicro are an enterprise setup to make money, and that possibly their financial interests are served by sticking with Intel? Absolutely. However, it looks as though their lack of AMD product is finally hurting enough for them to do something. http://www.forbes.com/technology/feeds/afx/2005/11/20/afx2347168.html To Bob, My apologies. I had spent a bit of time recently looking for Opteron systems on their site without success. The fact that they do not seem to feel them worthy of mention on their home page I regard as an indication of their commitment. The page listing their AMD boards : http://www.supermicro.com/Aplus/motherboard/ is headed For OEM Customers, so I take it from that I cannot order one from my local supplier. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Jitterbuffer and trunking
Steve Kann wrote: Richard Scobie wrote: My SVN asterisk systems use the following topologies: 1) PolycomSIP - *1 -IAX- *2 - H323 Gateway 2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog 3) H323 Gateway - *2 -IAX- *3 - Zap TDM400 Analog There's a few points in here so far: 1) the new jitterbuffer and trunktimestamps are independent settings, and have independent effect. You get the same effect with trunktimestamps (correct pass-through of frame timestamps), whether you use the jb or not. Thanks. As they were introduced with the new jb, I was unsure if they were dependent on each other. 2) The IAX jitterbuffer is disabled _by default_ (unless you use forcejitterbuffer), when a call is bridged from an IAX channel to another VoIP channel. So, you don't need to forcibly disable the jb in your case, it should automatically be disabled: In your cases, it would only ever be enabled on box *3, when a call comes in from IAX, and goes to zap. I had noted this on *1 (the SIP end) and was not sure whether there was any extra fixed delay in the path with it automatically disabled, so to be sure, forcibly disabled it. However, on *2 (the H323 GW end), the IAX jb is not being automatically disabled. I am using OOH323, so perhaps this is something I need to bring to their attention? 3) Yes, the setting in zapata.conf is for 4 very small buffers, which are different than than the IAX jb. OK. Thanks again for clarification on all this. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo Canceller usage
Using a TDM400P with an FXO module and an FXS module, and a zapata.conf with echocancel=yes above both channel definitions, is echo cancelling applied individually to each module when a call is made out to the PSTN? Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo Canceller usage
Kevin P. Fleming wrote: Individually? Yes... but I don't know how else you are thinking it would be applied. Apologies for breaking the thread. Just trying to get an idea of how things work together. I had considered that in this scenario, the echo can on the FXS only has to deal with a tail length back to the FXO hybrid, which on adequate hardware would be so short that any echo would just be sidetone and so could be dispensed with for the sake of CPU usage. The echo can on the FXO would be the one doing the work, on the tail back to the far end hybrid. Or have I misunderstood how Zap EC works? Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Jitterbuffer and trunking
Is there a way to configure the IAX jitterbuffer to get the benefit of trunktimestamps, while not having any jitterbuffering (reducing delay)? My SVN asterisk systems use the following topologies: 1) PolycomSIP - *1 -IAX- *2 - H323 Gateway 2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog 3) H323 Gateway - *2 -IAX- *3 - Zap TDM400 Analog In all the above, the primary jitter path is the IAX one and the codec is Alaw all the way. In an effort to reduce path delay and multiple jitterbuffering I have configured the following: On the basis that the Polycom IP500 phones have a decent jitterbuffer built in, Asterisk 1 has jitterbuffer=no in iax.conf. Asterisk 2 has the same setting as the H323 GW has it's own jitterbuffer. Asterisk 3 has jitterbuffer=yes in iax.conf, to buffer the Zap interface and provide PLC. I notice that zapata.conf has an entry jitterbuffers=4 by default - is this a different one in which case should it be turned off or is it setting parameters for the IAX JB? Looking at README.jitterbuffer: If you don't use trunktimestamps, there's lots of ways the jitterbuffer can get confused because timestamps aren't necessarily sent through the trunk correctly. This presumably means that if I want to use IAX trunking effectively, I have to enable the IAX JB on all Asterisks. Thanks, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 stability problem.
Adam Rybak wrote: Hello, i have succesfully ipgraded my system to asterisk 1.2 with OOH323C channel driver, today i got hangup of my asterisk after this messages: Nov 25 21:03:22 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! This issue is currently being looked at by the ooh323 developers and a couple of patches were submitted to asterisk-addons cvs today. You could try checking this out from CVS, but as I believe they are using the cvs head version of Asterisk, you may need to using this also in order to help them debug. Regards, Richard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_iax2: ast_sched_runq
Today I updated a couple of TDM400 based asterisks to the latest CVS head and started seeing the following messages. The update prior to today was a couple of weeks ago. -- Starting simple switch on 'Zap/6-1' -- Executing Dial(Zap/6-1, IAX2/[EMAIL PROTECTED]/0) in new stack -- Called [EMAIL PROTECTED]/0 -- Call accepted by 192.168.4.223 (format alaw) -- Format for call is alaw -- IAX2/pbxwn-1 answered Zap/6-1 Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 140 scheduled tasks all at once Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 110 scheduled tasks all at once Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 106 scheduled tasks all at once Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 86 scheduled tasks all at once Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 96 scheduled tasks all at once Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 300 scheduled tasks all at once Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 400 scheduled tasks all at once Nov 10 15:44:34 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 200 scheduled tasks all at once Nov 10 15:44:35 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 1237 scheduled tasks all at once Nov 10 15:44:35 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 3722 scheduled tasks all at once Nov 10 15:44:35 WARNING[7972]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 242 scheduled tasks all at once -- Hungup 'IAX2/pbxwn-1' == Spawn extension (default, 0, 1) exited non-zero on 'Zap/6-1' -- Hungup 'Zap/6-1' Zap6-1 is an FXS port on a TDM400 card, calling another identical P4 2.4 box. It was the only call in progress on a stripped down system running a minimal asterisk setup on a custom 2.4 kernel. System load is nil. The log below is from the box at the other end, placing a call back to the box above. Zap/1-1 is an FXO port on a TDM400: -- Starting simple switch on 'Zap/1-1' -- Executing Answer(Zap/1-1, ) in new stack -- Executing Set(Zap/1-1, TIMEOUT(digit)=10) in new stack -- Digit timeout set to 10 -- Executing Set(Zap/1-1, TIMEOUT(response)=20) in new stack -- Response timeout set to 20 == CDR updated on Zap/1-1 -- Executing Dial(Zap/1-1, IAX2/[EMAIL PROTECTED]/503) in new stack -- Called [EMAIL PROTECTED]/503 -- Call accepted by 192.168.3.223 (format alaw) -- Format for call is alaw -- IAX2/pbxak-1 is ringing -- IAX2/pbxak-1 stopped sounds -- IAX2/pbxak-1 answered Zap/1-1 Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 23 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 28 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 35 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 59 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 28 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 27 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 257 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 176 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 199 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 24 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 26 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 173 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 278 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 37 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 73 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread: chan_iax2: ast_sched_runq ran 762 scheduled tasks all at once Nov 10 16:18:56 WARNING[9362]: chan_iax2.c:7965 network_thread:
Re: [Asterisk-Users] Wits end with echo
Jon Reynolds wrote: Hello, I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000 phones, I am having echo issues on the GXP-2000 side. Here is what I have tried so far: The server has everything in the bios turned off except what is needed, USB, LPT, Serial etc,etc. I have uncommented Echo Suppresion in zconfig.h and shutdown and turned back on the asterisk box. I have updated the phones to 1.0.12 firmware, I have echotraining=800, echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using Mark2 as the echo suppresion and still I have echo. Is this correct? I do not believe having these echo parameters in sip.conf will achieve anything. They should be at the top of zapata.conf. Regards, Richard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New TDM Revision in the wild: J
It looks like there is a PE-68624 chip near each RJ-45 connector now. Google says that it's a frequency control filter Looking at the data sheet for this chip, it is being used as an EMI filter, (preventing RF interference generated on the card being radiated back out the cable). It would also offer filtering the other way - attenuating incoming spikes and RF noise. Regards, Richard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk not detecting PSTN hang-up
[EMAIL PROTECTED] wrote: Put in your zapata.conf for the channel: busydetect=yes busypattern=1500,500 busycount=4 callprogress=no Steve, is this a better solution than the COMPARE_TONE_AND_SILENCE busydetect option that can be enabled in the Makefile? Regards, Richard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CPU spiking with TDM400 cards fixed
Of possible interest to people having various issues with TDM400 cards, is that a fix has just been submitted to CVS for the issue where CPU usage would regularly spike up to 100% with the wctdm driver loaded. Regards, Richard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer on Asterisk
Matt wrote: Am I correct in thinking that at this time the CVS-HEAD supports Jitter Buffer for SIP on Asterisk? No, but attached to issue 3854 you will find patches you may be able to apply to the current CVS-Head to acheive this. Regards, Richard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk
jennyw wrote: I've never heard about IO-APIC before, so I just did a Google search. The articles I found say that it's an Intel thing, and, since I have an AMD processor w/ ASUS motherboard, it's unlikely it'll work, right? Even so, it sounds interesting. But does it apply to single-processor systems, even if they're using they're using Intel processors and Intel chipsets? The articles say this is for SMP systems ... Hi Jenny, IOAPIC is not an Intel thing, nor exclusive to SMP. Here is a small section from dmesg on my nVidia nForce 4 chipset, single Athlon 64 system: ACPI: Using IOAPIC for interrupt routing PCI: Using ACPI for IRQ routing ACPI: PCI Interrupt Link [APCS] enabled at IRQ 23 ACPI: PCI interrupt :00:01.1[A] - GSI 23 (level, low) - IRQ 177 ACPI: PCI Interrupt Link [APCF] enabled at IRQ 22 ACPI: PCI interrupt :00:02.0[A] - GSI 22 (level, low) - IRQ 185 ACPI: PCI Interrupt Link [APCL] enabled at IRQ 21 ACPI: PCI interrupt :00:02.1[B] - GSI 21 (level, low) - IRQ 193 ACPI: PCI Interrupt Link [APCJ] enabled at IRQ 20 ACPI: PCI interrupt :00:04.0[A] - GSI 20 (level, low) - IRQ 201 Regards, Richard ___ 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] How to test H.323
Frank Tarczynski wrote: What is the easiest way to check if the H.323 code is working? I've edited the h323.conf and extensions.conf files but I'm sure that things aren't right. I've tried connecting to my asterisk box via netmeeting but I'm having much success. I don't know if my conf files are screwed-up or if ooh323c code isn't working. /var/log/asterisk/h323.log should give you some good information. Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 implementations
I am about to add h323 to my system and although I have found information on the Wiki, comparing the asterisk implementation to oh323, I have not found anything about the new ooh323, which is included in the addons. Can anyone please compare this to the other two? Thanks, Richard ___ 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] What is the Polycom 301, 501 601?
Matt Darnell wrote: These phones are mentioned in the Sip 1.5 manuals, anyone know what the differences are? Where are you getting SIP 1.5 from? When I log into the Polycom download area, all I can find is 1.4.1. Regards, Richard ___ 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] Polycom IP500 - Phone TIme
Paul Hales wrote: It now works - but only in the latest (1.5+) firmware releases. Where are the 1.5 releases? I see only 1.4.1 on all the Polycom sites. Regards, Richard ___ 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] Polycom IP500 - Phone TIme
Paul Hales wrote: And my dreamthat one day Polycom phones will support Australian Daylight savings... But it's only a dream. Unless I am missing something, you don't need to dream about it - set it in ipmid.cfg. Look at the Sip Admim PDF for an explanation of: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1 Regards, Richard ___ 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] Zaptel FXO crashing.
Jason Leach wrote: About every 24-48h the Zaptel FXO port crashes. If I pick up my phone and try to make a call on the FXS port I get a hissing and squealing sound. Seems to be right where Asterisk makes the bridge. Also Asterisk does not answer an inbound call on the FXO port; does not even display as ringing. To get the system working again. I must stop asterisk, restart zaptel and then restart asterisk. Next time an FXO stops responding, stop asterisk and do a register dump of the offending module. You may need to cd into the zaptel src directory - I'm not sure that fxstest is installed. ./fxstest /dev/zap/1 regdump will show you the contents of all the registers on Zap 1. If the majority of them show the value ff, contact Digium support. I had modules marked Rev C that did this replaced with X100B RevB ones and have not had any trouble since. Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wctdm module parameters (Was: Issues with ringing on FXS ports)
[EMAIL PROTECTED] wrote: Is there a list of these anywhere? This is now the third one I've heard of, with no documentation: lowpower (IIRC), robust and now boostringer. Do I have to go diving in the source, or is there a Wiki I can't find? I have only ever found the information in the driver source of on the CVS list as they have been added. There is a list of them at the end of wctdm.c. The non obvious ones I know about: opermode=COUNTRY Where COUNTRY is one from the list near the top of wctdm.c This will set the A.C. and D.C. line impedance on the FXO modules to suit the telecom standard used in that country. Default if not set, is FCC (US/Canada). fxshonormode=1 If used, it must be in conjuction with the above. This will set the A.C. and D.C. line impedance on the FXS modules to match COUNTRY. Default if not set, is FCC (US/Canada). lowpower=1 Reduces ringing volts on FXS to 50V peak. boostringer=1 Boosts ringing volts on FXS to 89V peak. Regards, Richard ___ 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] Issues with ringing on FXS ports
Ian Pattison wrote: ringing. If I connect the GE cordless models I've been asked to use (0.1 REN) they normally will not ring... they light up and indicate there's an incoming call and once or twice I've received 1/2 - 3/4 of a ring but never a complete ring or multiple rings. When connected directly to my incoming lines they ring normally. Normally I'd assume this was a power problem but at 0.1 REN?? Any other ideas I can try? Try adding the module parameter boostringer=1 when loading the wctdm driver. This raises the ringing volts to 89V peak. Regards, Richard ___ 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] Sangoma VS. Digium
David Brodbeck wrote: -Original Message- From: Scott Nelson [mailto:[EMAIL PROTECTED] Perhaps you have an earlier hardware revision than I do; I also have never rebooted the system. I have two TDM04Bs. If so, they must have sold me old stock. I bought the cards less than two months ago. This is about the timeframe new FXO modules were released. Next time an FXO stops responding, stop asterisk and do a register dump of the offending module fxstest /dev/zap/1 regdump will show you the contents of all the registers on Zap 1. If the majority of them show the value ff, contact Digium support. I had modules marked Rev C that did this replaced with X100B RevB ones and have not had any trouble since. Regards, Richard ___ 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] Major problems with TDM400 and specific telephones: suggestions?
Wilson Pickett wrote: Now here's a thread I've been waiting to see! I have had issues with what is considered to be a decent phone, the siemens DECT line. Fortunately, the problem is just callerID which although annoying, isn't mission critical, and we are in Europe. Still, the USA isn't the world, the more adaptable the technology solution, the better. I talked to Digium and got prompt action but it didn't change anything at all. What are the CID issues? I am using Siemens Gigaset DECT phones and CID (albeit internal - not from PSTN) is working. Regards, Richard ___ 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] Zap channels not hanging up...
Carlos Chavez wrote: I have 2 Asterisk servers that communicate with IAX2 between them and support multiple SIP clients each. Only one of them has Zap channels to the PSTN. I've been having problems because the Zap channels do not hang up when a sip client of the external server makes a call to the PSTN. SIP --- Asterisk IAX2 Asterisk --- Zap The local * server is using CVS-HEAD-03/08/05-16:08:10 and has 3 X100P cards. The remote server is using Stable 1.0.6. When I use a SIP phone on the local network the Zap channel hangs up properly, it only happens if the call comes from the remote server or it has happened a couple of times when I redirect my desk phone to my cell. Have a look at Bug 3813 and see if it fits with your experience. I suspect the echo cancellor and it would be interesting to see if the X100P has the same problem. I would encourage anyone who has been experiencing erratic or non - functioning busydetect to check it also. With the reliability I am now seeing with the latest FXO modules, I finally think I now have a production quality hardware setup. The inability to be able to fully adjust TX and RX gains on the FXO module to balance line loss to the PSTN is the only showstopper to me recommending the TDMXX solution for small setups. Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Excessive indications tone levels (longish)
Setup: POTS phone1 - Panasonic Analog PBX - Digium FXO - Asterisk1 - IAX2 - Asterisk2 - Digium FXS - POTS phone2 I am attempting to balance the Digium FXO (shown above), analog audio levels using ztmonitor -v, which the information I have found means getting the TX and RX indicators to hit half scale on normal speech. This results in an RX gain setting of 3.0 and TX gain of 0.0 and levels on both phones sound good to each party. If however a call is placed as per the flow above and POTS phone2 is not answered, when POTS phone1 is hung up, the FXO on Asterisk 1 is never hung up, due to a failure of busy detect. Looking at ztmonitor on the FXO, this is because the ringing tone on the TX side is going off the scale and after a few rings, appears on the RX side, overwhelming the congestion tone coming from the Panasonic when POTS phone 1 is hung up. I guess this is caused by the echo canceler being overloaded. If the RX gain is reduced to 0.0, busydetect works about 80% of the time. All of this is a long winded way of asking if there is somewhere in the code I can reduce the level of the indications tones (ringing, busy,congestion etc.)? Given that the TX gain is set to 0.0 and these tones register off the scale, it would seem they are too high. I am not a coder and some ploughing through indications.c shows mention of vol but I am not sure where this can be changed. Any help would be much appreciated. Regards, Richard ___ 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] EADS6550 and asterisk - echo on PSTN call
administrator tootai wrote: If you're telling that I have to pass parameters to module when loading, I checked with modinfo wctdm (at office I have head version) and options I have are those: [EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm debug int loopcurrent int robust int _opermode int opermode string timingonly int lowpower int boostringer int fxshonormode int battdebounce int battthresh int alawoverride int Pardon my ignorance but no one of them remaind me to impedance. And for what I saw earlier in the source file, those informations could be updated with the value of the zaptel.conf file. You need to add opermode=FRANCE If you are using Didium FXS modules as well, with non 600 ohm phones, i.e. European TBR21 standard, you should add fxshonormode=1 as well. This will set the FXS impedance to whatever is specified in the opermode = parameter. You can confirm this afterwards by checking dmesg for an entry showing: Module 0: Installed -- AUTO FXO (FRANCE mode) Regards, Richard ___ 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] TDM04B lock up
Goutam Shaw wrote: Hi I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B cards lock up at the same time and stop processing incoming and outgoing calls even though * shows that it is trying to communicate to ZAP channels (at least on the outgoing). The only cure is to reboot the system when it happens. It makes me very apprehensive of the system Has anyone seen this problem. Could this be something to do with the IRQ sharing. Here is the output of lspci -v. Get the cards on their own interrupts - use the BIOS, turn off unneeded onboard devices etc. If you still have problems and the FXO modules are marked Rev C on the non pin side, talk to Digium support. I had ongoing problems with FXO modules stopping responding and requiring reboots to restore. Regdumps of the offending module show ff loaded in almost all registers. After contacting Digium, I was told this was a hardware issue and after having them replaced with modules marked X100B RevB, I have so far had no problems. Regards, Richard ___ 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] Return of experience : Asterisk more stable with 2.6 or 2.4
Jeremy SALMON wrote: Hi, Just a question, For you, what is the more reliable kernel for an asterisk prod server... The following 2 recent quotes from kernel developers may be worth considering when making your decision: After 2.6.9-ac its clear that the long 2.6.9 process worked very badly. While 2.6.10 is looking much better its long period meant the allegedly official base kernel was a complete pile of insecure donkey turd for months. That doesn't hurt most vendor users but it does hurt those trying to do stuff on the base kernels very badly. -- Alan Cox Not all 2.6.x kernels will be good; but if we do releases every 1 or 2 weeks, some of them *will* be good. The problem with the -rc releases is that we try to predict in advance which releases in advance will be stable, and we don't seem to be able to do a good job of that. If we do a release every week, my guess is that at least 1 in 3 releases will turn out to be stable enough for most purposes. But we won't know until after 2 or 3 days which releases will be the good ones. -- Ted Ts'o Regards, Richard ___ 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] passing opermode to the wcfxs module
Kavit Munshi wrote: Hi, Has anyone in australia got asterisk running on FreeBSD? how would i pass the opermode=AUSTRALIA parameter to the wcfxs.ko module as kldload doesnt let you pass parameters to the module like modprobe in Linux. I tried to get the sysctl variable using sysctl -a it might use but nothing applicable was visible i need to set the impedences to 600 ohms, can someone please advice me on how to pass the parameters to the module? I'm sorry, I have no BSD experience, but if you really do require 600 ohm impedence, you should not use an opermode= as the default impedance is 600 ohms. opermode=AUSTRALIA will get you 220 ohms + 820 ohms || 120nF. Regards, Richard ___ 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] Qs about FXO/FXS cards
Andrei (MPI) wrote: Richard Scobie wrote: It is a simple one liner. ... Index: wctdm.c ... + reset_spi(wc,card); ... This is exact same patch that Digium support tried before sending me new fxo modules. That wctdm.c patch did not help in my case. Interesting, thanks Andrei. I have run for a month so far, without any trouble since including this. The issues I have seen, have occured at 2 week to 2 month intervals. The system has not had much use for the last 3 weeks, so I'll need to wait a few months to be sure. Regards, Richard ___ 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] Qs about FXO/FXS cards
Steven Critchfield wrote: Okay, link this to my rambling above and you would see that by thrashing the disk, you are actually keeping the spindle spooled up and not measuring the spool up draw. My guess is a spooled down machine getting a random incoming call that then must generate ring and spool up the HD(s) to start writing logs at the same time on a questionable PSU. ??! Who set's up servers with hard drives that are spun down? Powered down hard disks are for laptops and specialised setups where power consumption is critical. These are probably not where people install serious asterisk servers. Regards, Richard ___ 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] Qs about FXO/FXS cards
Victor Rini wrote: This has been an interesting discussion. I'll chime in with my experience here. I have two servers. One with the cheapest motherboard and athlon processor I could find on Newegg.com. The other is a 1999 era motherboard with a Via C3 processor, again a bargain basement special. The Athlon system has a decent power supply - 400+ watt, the Via has a very generic PS that came with the case - 300 watt tops. On both system I have TDM cards, the Athlon has a 4 port FXS and two x100p's, the Via has a 2 port FXS. Both systems are in production if you could call it that because they handle little traffic - home/hobby systems. I have had no problems at all with the tdm cards or Asterisk. I occassionally lose my network on the Athlon machine - I chalk that up to the fact that I'm currently sharing an IRQ with two ethernet cards and an X100P. I'm thinking of ditching the two x100p's in my Athlon machine for a TDM card with FXO modules to free up a slot and hopefully the burdened IRQ. Based on what I'm reading here I probably should think *really hard* about that. I'd stick with the X100Ps for now. Two of my systems where X100s have been replaced by TDM FXOs, have seen a drop in reliability - the well noted FXO fails to respond to either calls from the PSTN or take calls from internal phones. Driver reloads and sometimes machine reboots are required to restore operation. These are systems seeing small office loads - 30 to 70 calls a day. One of these, which only contained 2 X100s, ran absolutely trouble free for a year and has required 2 reboots since switching to a single 4 FXO TDM. It is interesting to see there have been no posts to this thread from small office load users saying, I am using TDM FXO's and have not had any problems at all. All my contact with others in similar situations to myself have elicited the same experiences. Based on the above and following, I'd say the TDM FXO's have issues: 1. The following line in wctdm.c : /* Try to track issues that plague slot one FXO's */ 2. Digium acknowledge there is a problem and I am currently testing a driver modification for them. Looking at the FXS modules, I too get Power Alarm messages on a semi-regular basis, although they seem harmless. It is hard to see how the power supply is involved - short of a PSU suffering very poor regulation. These messages are created by interrupts generated when power dissipation in various transistors external to the ProSLIC, exceed programmed limits. To quote from the data sheet: This feature protects the external transistors from fault conditions and, combined with the loop voltage and current monitors, allows diagnosis of the type of fault condition present on the line. Anyone interested in learning more, should read page 27 of the Si3210/11 data sheet, which is available from www.silabs.com, after a free registration. Regards, Richard ___ 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] Qs about FXO/FXS cards
Rich Adamson wrote:
Have you noticed that a TDM with fxo modules is more/less stable then
a TDM with only fxs modules?
Gut feeling (no reasonable analysis at all) from various postings tend
to suggest the TDM with fxo's is less stable. Would you agree or not?
Yes.
Also, could you share the driver mod?
It is a simple one liner.
Index: wctdm.c
===
RCS file: /usr/cvsroot/zaptel/wctdm.c,v
retrieving revision 1.90
diff -u -r1.90 wctdm.c
--- wctdm.c 9 Nov 2004 14:45:20 - 1.90
+++ wctdm.c 29 Nov 2004 15:35:46 -
@@ -1387,6 +1387,7 @@
#endif
signed char b;
int poopy = 0;
+ reset_spi(wc,card);
/* Try to track issues that plague slot one FXO's */
b = wctdm_getreg(wc, card, 5);
if ((b 0x2) || !(b 0x8)) {
Since adding it, I have not had any problems, but it will take another
couple of months before I know if it has had any effect.
Regards,
Richard
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Re: [Asterisk-Users] Looking for new hardware
Steven Critchfield wrote: I would suggest something in a serverworks board. So far we have had a PIII 850 on a serverworks chipset and SCSI drive running for a long time. Our main PSTN gateway has a 418 day uptime and asterisk has been running non-stop for nearly 20 weeks. We take nearly 500 calls a day right now on that machine. Are these calls using TDM400P FXO modules? Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcfxs causing constant CPU spikes
I'm really discouraged at Digium's disinterest in this problem. I understand they have limited resources with lots to do, but it only takes a minute to reply to my email to say either they are aware or not aware of the problem. Digium are aware and have acknowledged the issue. See my replies to your thread earlier this month: http://lists.digium.com/pipermail/asterisk-users/2004-December/075740.html http://lists.digium.com/pipermail/asterisk-users/2004-December/075996.html Regards, Richard ___ 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] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Rich Adamson wrote: The tdm card does have some unusual issues that appear to be driver oriented, but there are lots of folks using the card in production. Usually in situations where the client knows how to and tolerates having to reload drivers and/or reboot his PBX periodically... Regards, Richard ___ 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] Interrupt latency problems
Rich Adamson wrote: Is their an open Bug # that we can track against for those of us that watch the -cvs list and have a vested interest? I tried, see Bug 2901. Seems probable driver issues don't count as bugs. Reported it to support, who were aware of the issue, requested login to my machine and the last I heard from them as of last week was: We currently have all the data to begin fixing the problem at the software level we should have a patch for the zaptel driver soon. Thank you for all your help. Regards, Richard ___ 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] Interrupt latency problems
Steven Critchfield wrote: On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote: Steven Critchfield wrote: On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote: So, isn't the issue he/I are chasing after essentially 'why is cpu consumption jumping 30% (or 100%) every ten seconds when zaptel is running with no calls present? So where is that CPU time going? Is it in the system, or userspace? Have you tried changing to a non FC or RH kernel as suggested earlier? Yes, I've just completed the installation of 2.6.9, and the spikes have gone away. Thank you, Steven. Your welcome. I am glad it solved the problem. Now if only someone knew what it was about the stock RH or FC kernel that makes it happen you could get RH or FC to stop using that patch. That or maybe more people will be like me and always cast a weary eye upon a prepackaged kernel no matter what distro it came from. See http://lists.digium.com/pipermail/asterisk-dev/2004-November/007329.html This has been brought to Digiums attention and they are currently working on it. Regards, Richard ___ 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] Zap FXO channel locked up with steadystatic(white noise)
Damon Estep wrote: I'm having the same problem on my TDM40B (FXS). Unloading and loading the modules seems to fix it temporally. Digium is sending me a replacement. Hopefully that will fix it. I plan to call tech support and see what they have to say, hopefully it is just defective and not un-reliable. Have you heard other complaints of the same thing? I have had the same issues with the FXO modules on systems that have run fine for a year using X100Ps. Rebooting the box has been required to fix it. Please let us know what Digium tell you. Regards, Richard ___ 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] kernel: Power alarm on module 1, resetting!
Gabriel Gunderson wrote: I've installed a TDM04B and a TDM40B. I haven't plugged any lines into them yet but I'm starting to see this in my logs... [EMAIL PROTECTED] asterisk]# grep alarm /var/log/messages Sep 20 09:13:22 webster kernel: Power alarm on module 1, resetting! Sep 22 11:07:07 webster kernel: Power alarm on module 1, resetting! Sep 22 16:10:55 webster kernel: Power alarm on module 1, resetting! Sep 23 02:42:33 webster kernel: Power alarm on module 1, resetting! Sep 24 03:34:58 webster kernel: Power alarm on module 1, resetting! Should I be worried? Anyone know what this is? These are generated by an FXS module. I do not know why, as I and a number of others see them on a semi regular basis - in my case 6 or so times a month and very occasionally there will be a burst of four, the same message but each module in numerical order on a 4 module card. Given that a number of users see this same message on module 1, I do not think it is a faulty module. They do not seem to be related to phone usage, as they occur during times when the office is vacant. Ouch, part reset, quickly restoring reality is another message seen less frequently. Regards, Richard ___ 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] Static noise and server locked when using two 4FXO tdm400p pci cards
Luis Vazquez wrote: Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server using two(x24)channel banks (Access Bank 1) an three TDM400P pci cards with 4 FXO each. But when we use more that one TDM400P card, after some random number of calls, one of the cards starts to give a loud static noise when calling from inside in all their channels and if we keep trying to use the lines the server gets frozen. Restarting Asterisk don't solves the problem and the only way of recovering the channels is to reload the zaptel modules (if the system is not locked yet). Sometimes that does not work and a full reboot is required. We have seen some similar problems reports in the list, and some people telling they asked to digium support, but not a real solution. Does anybody knows if is this a major hardware problem with Digium TDM cards and zaptel driver or if there is some way of fixing this? I am not sure if it hardware or software related. I have a box which had 2 x X100P and 2 x TDM400 (8 x FXS), all on unique IRQs, which was relatively reliable until recently when the 2 x X100Ps were replaced by 1 x TDM400 (2 x FXO) and I have seen the static and freezes a number of times - reliability has dropped. The TDM FXO card has the jumper modification so I believe it is the latest available hardware. Regards, Richard ___ 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] Detecting DTMF tones
[EMAIL PROTECTED] wrote: On 15 Sep 2004 at 1:52, San Singhania wrote: Hello everyone, I am having big problems trying to detect dtmf tones while a IVR prompt is playing on zap channels. Sometimes the detection only starts 4-5 seconds into the prompts. Other times it works very well for the 1st few calls and then starts having problems. And most times it also does not detect all digits, eg when 123 is keyed in, it may detect 23 or 2 or 3, but never the complete string. Can someone help me with this? I am in Singapore so I dont know if its a localisation issue. Also I have relexdtmf on and my rxgain and txgain on zap channels are set to 0 so i dont think thats the problem. Help is most appreciated as I cannot go on with Asterisk if this function does not work :( Ok, I'm based here in New Zealand and found that the volumes for PSTN were quite a way off. I do not think this is a localisation issue - Mark has measured this (see Bug 2023). When I replaced X100Ps with TDM FXOs, busydetect stopped working until until the RX gain was increased 3dB. Which of the two was correct or maybe neither of them? Given that gain distribution in analog telephony is quite critical from a stability/echo performance point of view, it would be great if the cards came out of the box correctly calibrated for spec sheet levels. There are enough real world variables to deal with, without wondering if the TDM FXO TX gain is correct and the FXS RX gain is low or vice versa. I do realise that test sets, CO and asterisk milliwatt generators are needed to achieve a close to perfect result, but looping an FXS into an FXO should show unity before starting out. Regards, Richard ___ 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] TDM400P lockups (FXO)
David wrote: It sounds like my lockups may be related since my TDM422b card has the FXS FXS FXO FXO configuration and doesn't have an FXO in position 1 either. My card is identified in software as Rev E/F and has the wire jumper on the back. Further investigation shows that my TDM cards have the wire jumper and so are the current ones. Regards, Richard ___ 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] TDM400P lockups (FXO)
Maciej Kietlinski wrote: Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to do with *any* FXO on port 1... Please get back with the list with your findings. My experience led to a replacement from Digium, but the card is a TDM400P with 4 FXO...now that I think of it, during troubleshooting there was some correlation to the first port on the card (port 1)...not the first module - I swapped module positions to varying locations on the card without success, but then again they are all FXO...Maybe *is* possible that the TDM400P doesn't like an FXO module in port 1 as you are suggesting...Like I said, in the end I got a new revision board from digium, all 4 ports are still FXO and working great now... With my old revision TDM400P it was the same problem with FXO on port 1. Easiest way for me was to put FXO's on new revision card, and on old use FXS on port 1. I used info from post with: 'The card had been modified, evident from the jumper wire that been soldered between two points on the back of the card. I haven't had problems since installing the new card.' And before old card was used with FXS + 3 x FXO without problems, so it works in the same hw conf again. Now I heve no problems with TDMxxp I'll let you know how I get on. One of the cards that is giving trouble, has FXOs in positions 3 and 4. Can anyone tell me what these new revision cards are? My current ones are all Rev. E/F. Regards, Richard ___ 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] TDM400P lockups (FXO)
Michael George wrote: To follow up on this, I heard back from Digium and they asked the configuration of my TDM. It was: FXO,FXS,FXS,FXS. They said they have had report of this configuration being a problem and that I should change it to FXS,FXS,FXS,FXO. Before the change the system would either hang or lock up a port within 3 days. After the change I have over 10 days of uptime again. Thanks, Digium!! Interesting. I am waiting to hear from Digium with similar issues, on two cards, one with 4 x FXO and one with 2 x FXO. Regrads, Richard ___ 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] UK Disconnect supervision with TDM400P
Edward Eastman wrote: I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the remote party hangs up, I get approx 20secs of the bt line hungup tone before asterisk hangs up, which leads (if nothing else) to the well documented 20secs of beep on vm problem :) I have tried: busydetect=yes / busycount=7 / other busycounts / callprogess=yes but none of these make any difference. I have loadzone/defaultzone=uk and country=uk in indications.conf and fxs_ks signalling. Try increasing your RX gain in 1db steps, until it reliably hangs up. I had a box with X100Ps which busydetected perfectly with default gain settings. When they were replaced with TDM FXOs, busydetect stopped working and I needed 3db of RX gain added to get it working again. Regards, Richard ___ 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] system reboot often?
Leif Madsen wrote: Would you mind maybe expanding upon the hardware configuration you are using and why? I, and I'm sure others, are curious as to what you are using. I haven't had to roll out any systems yet that require multiple Digium cards, but I'm sure the information would be quite useful as I've seen few posts regarding this issue. Sure. They are nothing special - Asus P4B533 motherboard, P4 2.4, 256MB RAM, 40GB Western Digital SE PATA and 3c905c network card. As to why, the CPU was best value at the time and 533 FSB had just been introduced. The other components all seem to be solid performers in their classes. I will soon add a second drive - software RAID1. Personally I would not be deploying these for commercial use, as I feel the uptime is still not there. Regards, Richard ___ 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] system reboot often?
Michael George wrote: Well, we only want 3 TDM400s: 4 FXO and 8 FXS. That will fit in nearly any desktop PC. That's not the scale that should require multiple boxes. But the question is where does the IRQ sharing instability creep in? I would think that *someone* out there would have a * box with 2-4 Digium cards in it that might be willing to share their experience. If the Digium cards can only be reliably run in a machine with only 1 or 2 of them, then I need to know so we can plan appropriately. As Rich alluded to, it's a bit of a lottery. I have two identical P4 2.4 boxes with Intel 845 chipsets running updated, stripped down Redhat 7.3 and custom compiled kernels containing nothing more than is required for asterisk in a headless, ssh access only situation. All onboard sound, USB etc. is disabled in the BIOS. One box has 3 x TDM400P - 2 x FXO and 8 x FXS (latest rev) all on individual IRQs. Until a couple of months ago it had 2 x X100P and 8 x TDM400 FXS and required driver reloads about every couple of months over a 1 year period. I replaced the 2 X100s with the TDM FXOs for a few reasons including a hoped for improvement in reliability. In the 2 months since, I have had to reload the drivers once - the logs showed 4 error mesages, Ouch, part reset, restoring reality for each of the ports on one of the FXS cards. The second box has a single TDM with 4 x FXO which is IAX2 trunked to the first. It originally had 2 x X100Ps and gave no problems at all for a year. In the 2 months since replacing with the single TDM, the drivers have needed reloading once, with no sign of any errors in logs. Apart from the above, I have been very happy the system and have had no echo problems. Regards, Richard ___ 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] RAID affecting X100P performance...
Mike Benoit wrote: I have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear beeps and cutting out on a call using the X100P card. I ran the zttest program, and discovered HD activity would drop the accuracy down to between 2% and 50%. However I noticed if I disabled one drive in the RAID1 array, zttest would always report 99.98% or higher. So one drive running works fine, but as soon as I enable the second drive, all hell breaks loose. DMA and 32-bit mode are enabled on both drives as well. Have you tried hdparm -u1 on the drives as well? This prevents other IRQs being blocked during disk access. Regards, Richard ___ 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 - delay before dialing last DTMF digit?
William R Sowerbutts wrote: I have a TDM22B (TDM400 PCI + 2xFXO + 2xFXS). One FXO is connected to the PSTN. When Asterisk places a call, it dials using DTMF. If I listen in on the line during the dialing, there is a roughly one second pause between the penultimate and the final digit -- proportionally much longer than the pause between the preceeding digits. This accounts for about 25% of the time taken to dial. Anyone know why this happens? I'd like to remove it. As I understand it, the echotraining pulse is sent during during this pause, so removing echotraining from your conf. should fix it. Regards, Richard ___ 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] Stopping reinvite with IAX2?
Brian K. West wrote: per peer bkw - Original Message - From: Michael Graves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:25 PM Subject: Re: [Asterisk-Users] Stopping reinvite with IAX2? Is this set on a per peer basis, or in the general section? Michael Actually, as a result of bug 1579, it can also be applied to the general section, if using CVS. Regards, Richard ___ 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 4.10 dies
Dr. Rich Murphey wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arjan On Sun, 11 Jul 2004 at 15:39 -0500, Dr. Rich Murphey wrote: You might check login class in login.conf for the user that invokes asterisk. Setting cputime=unlimited may help. This will prevent the kernel from killing the process but I'm puzzled by the load Asterisk generates on a AMD XP+ 2000 cpu. While running the box goes to 40%, even though Asterisk is doing nothing (well at least: not handling calls, etc). That sounds like a bug. One should be able to attach to the process in gdb, stop the process and see where it's looping. Rich A slightly similar observation, which I assume is normal as the boxes work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load, about every 30 seconds, with no calls being handled. The boxes are only running asterisk, ntpd, sshd and the core 2.4 kernel services and the load can be observed in top by noting the system CPU usage figure in the upper part of the top display - no CPU usage is shown by any of the listed processes. As I say, it is probably normal, but I've wondered what causes it. Richard ___ 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 4.10 dies
Steven Critchfield wrote: On Mon, 2004-07-12 at 02:38, Richard Scobie wrote: A slightly similar observation, which I assume is normal as the boxes work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load, about every 30 seconds, with no calls being handled. You don't mention it, but it sounds like you are running either Fedora core or Red Hat. There are known problems regarding the threading in those distos. The is a known work around too. No, I have avoided the later RedHat distros for that reason. It is a stripped down RH 7.3 and updates with a custom compiled bare bones 2.4.21 kernel, (from memory - I'll be updating once .27 is out). As far as I know, threading is pretty standard there. If you haven't done so, take a moment to read or listen about quantum mechanics. The problems with wuantum mechanics is similar to what you describe. The tool you are using to make measurements affects what you are measuring. Something to think about. Top is a very crude way of measuring system load. It is nice and useful, but remember it's short comings. True, but it is out of character with what I see on similar boxes not running asterisk. What would one use to measure this less obtrusively? Regards, Richard ___ 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] Hangup's not detected correctly
Martin Pycko wrote: Well first of all if you're outside of US or callprogress-supported zones then you can use only busydetect. And that will only work if after the remote hangup your telco gives the fast-busy or any type of busy. You can tweak the duration of tone/pause and increase the count and it *will* work properly. regards Martin One thing to watch for here is RX gain if busydetect does not seem to be working after trying all the combinations. I had a 2 x X101P setup which busydetected perfectly - TX and RX gains were at the default levels. The X101Ps were replaced by a TDM card with 4 x FXO modules and with no config changes, busydetect stopped working. After incrementing in 1dB steps, an RX gain of 3.0 brought back reliable busydetect. I look forward to Rich Adamsons forthcoming writeup on setting up the gain distribution in an Asterisk system, to get everything working optimally. Regards, Richard ___ 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, Line Impedence and Echo
[EMAIL PROTECTED] wrote: Does this only work with the new fxo modules? Yes. Richard ___ 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] X101P on a UK BT line ---- txgain issue
Chris Stenton wrote: Is opermode set via asterisk or do you need to do modprobe wcfxs opermode=UK You need to do modprobe wcfxs opermode=UK This will only work if you have the TDM400 FXO modules. The X101P is a 600 ohm US/JATE card only. Richard ___ 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] X101P on a UK BT line ---- txgain issue
Chris Stenton wrote: Thanks for the info Rich looks like I'll have to wait for the new FXO module. The impedence in the UK is zcomplex(2) which looks a long way away from a straight 600 ohms. Here is the list of zcomplex impedences Zcomplex(1) = 150 nF // 750 ohms + 270 ohms ( European harmonized, France Telecom Telefonica ) Zcomplex(2) =230 nF // 1050 ohms + 320 ohms ( British Telecom plc ) Zcomplex(3) = 115 nF // 820 ohms + 220 ohms ( Deutsche Telekom AG ) Zcomplex(4) = 310 nF // 620 ohms + 370 ohms ( Telecom New Zealand ) Zcomplex(5) =47 nF // 510 ohms + 150 ohms ( Russian Telecom ) These complex impedances are all supported in the Silabs chips used in both the new TDM FXO module and the FXS module, but the driver currently sets them to 600 Ohms. I guess at some stage a patch will appear to perhaps set these depending on the default tonezone set in the config files. Regards, Richard ___ 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: TDM Card loses Dialtone and Battery
Adam Goryachev wrote: I might add that I has similar problems on a very frequant basis, finally I 'accidentally' found a version of asterisk + zaptel modules that was stable for more than 6 weeks. Eventually I asked for (and got) a replacement card from digium with the internal power connector. This worked fine with the same software versions, although it crashed once after about 3 weeks. I've just updated to current CVS of everything, and will see how it goes. I'm not doing anything major out of the ordinary, I have a single X101P, a single TDM400P and a 2 channel (single BRI) i4l ISDN card. I use IAX to connect to a *very* lightly used extension (ie, iax to second asterisk to sip ata186). Regards, Adam Since updating my 2 TDM400P's, with 4 channels each, to REV E/F, I have seen very intermittant loss of dialtone on some channels, one hard lookup that required a power reset and a Power alarm on module 1, resetting!, which oddly occurred in the small hours of the morning, when the phone was not being used, over a +_ 2 month period, where the first card has had regular daily office use and the second card virtually none. The machine is a 2.4GHz P4 on an ASUS Intel chipset MB, running a stripped down Redhat 7.3 with a custom kernel compiled completely with asterisk requirements only. Everything not required for asterisk (USB, serial, parallel, mouse) has been disabled in the BIOS and there is no X or framebuffer installed. Two X101Ps are also in use. /proc/interrupts CPU0 0: 456297846 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 267137019 XT-PIC wcfxo 7: 16745155 XT-PIC eth0 8: 1 XT-PIC rtc 9: 267049399 XT-PIC wcfxo 11: 267335750 XT-PIC wcfxs 12: 3313776049 XT-PIC wcfxs 14:1912794 XT-PIC ide0 NMI: 0 LOC: 456293550 ERR: 0 MIS: 0 I would be interested in hearing from anyone else running 2 x X101 and 2 x TDM400 4 channels, who is having no problems at all. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation
Peter Zeltins wrote: I do not have a hardphone to play around with, but the echo is there both with built-in audio card (SigmaTel) and Bluetooth headset. There are no mixer settings than I can adjust as well. I'll try disabling AGC and/or lowering mike sensitivity. Peter According to the excellent Cisco paper Echo Analysis for Voice over IP, which the Wiki links to at: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml Headsets are particularly notorious for poor echo performance.. This is due to lack of acoustic isolation. Perhaps you could test using headphones and a mic. After reading the above paper, I was able to tune my setup and make a significant improvement. Given the high number of questions about echo on the list, it would almost be worth including the link to it as a file, README.echo in the source. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring confusion
Thanks for all the help and I found the different cadences in chan_zap.c. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring confusion
I am somewhat unsure as to the definition of Distinctive Ring. What I am trying to achieve is to have Zap connected phones (TDM400P) ring with different cadences depending on whether the call is incoming on the PSTN context or an IAX2 context. Googling, I find this from Mark: I've added distinctive ring support to Asterisk now (also I've added answer confirmation which is an essoterric feature that nobody but oliver will likely ever use). Now you can do the following: exten = 1,1,Dial,Zap/28 ; Ring Zap/28 normally exten = 2,1,Dial,Zap/28r1 ; Ring Zap/28 with ring #1 exten = 3,1,Dial,Zap/28r2 ; Ring Zap/28 with ring #2 and when I do a show application Dial, I see : 'r' -- indicate ringing to the calling party, pass no audio until answered Which doesn't seem to match up with what I have in mind. Assuming the former usage, and that it does what I am trying to do, do I have to define ring #1 and ring #2 somewhere, or are they hardcoded in? Thanks, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX/IAX2 encryption?
Louis-David Mitterrand wrote: Snip The main problem with ipsec packets is the lack of TOS support: data and voice traffic are agregated in one stream which is opaque to external routers. This is not the case with FreeS/WAN, below is an excerpt from the website: Can I use Quality of Service routing with FreeS/WAN? From project technical lead Henry Spencer: Do QoS add to FreeS/WAN? For example integrating DiffServ and FreeS/WAN? With a current version of FreeS/WAN, you will have to add hidetos=no to the config-setup section of your configuration file. By default, the TOS field of tunnel packets is zeroed; with hidetos=no, it is copied from the packet inside. (This is a modest security hole, which is why it is no longer the default.) DiffServ does not interact well with tunneling in general. Ways of improving this are being studied. Copying the TOS (type of service) information from the encapsulated packet to the outer header reveals the TOS information to an eavesdropper. This does not tell him much, but it might be of use in traffic analysis. Since we do not have to give it to him, our default is not to. Even with the TOS hidden, you can still: * apply QOS rules to the tunneled (ESP) packets; for example, by giving ESP packets a certain priority. * apply QOS rules to the packets as they enter or exit the tunnel via an IPsec virtual interface (eg. ipsec0). See ipsec.conf(5) for more on the hidetos= parameter. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answer on second ring - need it on first.
Steven Critchfield wrote: On Sat, 2003-10-04 at 12:02, Lists wrote: But would you then not be able to use caller id from the telco? CallerID on an analog line is sent between the first and second ring normally. So if the requester wants callerid and first ring answer, he will have to move to PRI. Some choices have drawbacks, of course PRI is usually just cost. Fortunately I do not require Caller ID. Adding usecallerid=no to zapata.conf has resulted in answering on the first ring. Interestingly it has also fixed a 196kB memory leak that occurs for every call to the X100P. This is outlined in Bug 356. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answer on second ring - need it on first.
Martin Pycko wrote: take out usecallerid=yes in zapata.conf Martin Thanks Martin, but my zapata.conf is : [channels] echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=6 context=incoming signalling=fxs_ks group=1 channel = 1-2 Perhaps I need a usecallerid=no in there. I'll test this when I'm back at work. I have not changed this prior to rebuilding the source. Regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answer on second ring - need it on first.
After some months of Make updates, I have just deleted my Zaptel and Asterisk source directories and done cvs checkout 's of asterisk and zaptel, in order to clean up the trees. After re-installing, I am finding that when dialling into an X100P, that Answer is now answering on the second ring, where it always used to answer on the first before. In the console, Starting simple switch on 'Zap/1-1' appears halfway through the first ring and Executing Answer(Zap/1-1, ) in new stack appears at the end of the second. I cannot recall having changed anything previously, in order for it answer on the first, but I would really like the old behaviour back. Thanks, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Andy Powell wrote: Can't find the message in a search.. but below is a msg retreved from my archive.. this is what Mark sent a little while ago I have no idea if it actually does anything to the card, but on a modprobe I do get a msg saying it's using CTR21 Andy I'm in Paris right now and can't test this change, but I've been researching the DAA and there are a few international settings I can change, so I've changed the driver in CVS so that you can specify the operational mode. Try modprobe wcfxo opermode=1 if you're in most of Europe and that should switch to CTR21 mode which slightly modifies a few of the electrical characteristics of the DAA. As we add modes you'll be able to see them with modprobe wcfxo opermode=-1 and then doing a dmesg. Anyway all you folks that had some trouble like this try it out and let me know if it makes any difference. Mark I guess in his haste to help out the people who were having a problem, Mark looked at the wrong data sheet when he wrote this patch. I have corresponded with him and confirmed that this code requires the Global DAA chipset, which is not fitted to the current X100P cards. The mail I was referring to is: http://lists.digium.com/pipermail/asterisk-users/2003-August/017825.html Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Andy Powell wrote: FCC mode is for the US. CTR21 is for Europe - you even pasted the info in your message! Exactly, the question really is how do you change it? modprobe wcfxo opermode=1 HTH Andy This switch (opermode=1) is redundant with the current X100P cards, as it changes register contents that are specific to the Global version of the chipset on the card. The X100P currently out there uses the USA Japan chipset, and thus does not achieve the intended result. The register concerned deals with the impedance presented to the line connected to the card - 600 ohms (US) vs various complex impedances used in other countries. For internationalised FXO cards, see Mark's recent comments, in the thread Does Wildcard x100p support BT Caller ID in UK? Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy detect options
I have been running busydetect=yes, using BUSYDETECT_MARTIN and am having hangups during calls. The busycount=6 workaround seems to be doing the job, but I was wondering if there was any value in using BUSYDETECT_TONEONLY or BUSYDETECT_COMPARE_TONE_AND_SILENCE, as well as BUSYDETECT_MARTIN or are all the options meant to be used on their own? Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRI newbie queries.
Knowing very little about Basic Rate ISDN and having spent the last couple of hours educating myself, I thought I would seek some more informed comment. Please go easy if this is blindingly obvious :) I have a ZyXEL Prestige 100 ISDN Router, a stand alone relic from when we used to access the Net via ISDN. It has an ISDN BRI input, a 10BaseT ethernet connector, an RS232 connector for administration and a couple of RJ11s to connect analog phones. I can find no reference to this unit on I4Linux lists or the CAPI site. Am I write in assuming that I cannot access the the ISDN data from Asterisk, via the ethernet port? Assuming this to be the case, are there any cards to avoid from an Asterisk point of view when looking to purchase? Thanks, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Instant hangup on busy Zap channel.
Martin Pycko wrote: Do 'iax2 debug' to see more. Martin Thanks Martin, As it seems as if it may be a bug, I'll get IAX2 debug output from both *'s and put them in the bug tracker to save list clutter. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Instant hangup on busy Zap channel.
A call is placed via IAX2 from one asterisk to another, to a TDM400
channel whose extensions.conf entry is
exten = 502,1,Dial(${COLIN})
exten = 502,2,Congestion
If this channel is already busy when called, the call is instantly
hungup, without the caller hearing the congestion tone.
The log from the callers asterisk shows:
-- Executing Dial(Zap/1-1, IAX2/192.168.3.223/502|30) in new stack
-- Called 192.168.3.223/502
-- Call accepted by 192.168.3.223 (format 4)
-- Format for call is 4
-- IAX2[pbxak]/2 is circuit-busy
== Everyone is busy at this time
-- Hungup 'IAX2[pbxak]/2'
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (incoming, 502, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Is this expected behavior, or have I missed something in the configuration?
Thanks,
Richard
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[Asterisk-Users] IAX2 Warning
When starting *, I get the following when the chan_iax2.so loads: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 As I am not loading chan_iax.so and IAX is working, it is obviously not fatal, but after a reasonable amount of checking docs, archives etc. I am curious to know why it occurs and it is not using port 5036 as specified in iax.conf. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog 2x8
Is anyone on the list running an Asterisk system with 2 x X100P and 2 x TDM40 (4 port) cards? I am interested in your hardware setup. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ProSLIC error message
I have just updated to the current CVS from CVS of 12 June and I now receive the following error message when I start *. Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized ProSLIC on module 2 failed to powerup within 510 ms Unable to do INITIAL ProSLIC powerup on module 2 Module 2: Not installed Module 3: Initialized Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) Any help appreciated. Richard Scobie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
