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

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, po

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 di

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. > >

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

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

[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

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 smar

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 upd

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)

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

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

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

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 _

[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 ___

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 upd

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 lates

Re: [Asterisk-Users] CentOS 4.x and ooh323

Bruce Reeves wrote: I'm trying to add ooh323c to my asterisk 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-

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.

Re: [Asterisk-Users] CentOS 4.x and ooh323

Bruce Reeves wrote: I'm trying to add ooh323c to my asterisk 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-

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

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 tha

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 i

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

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 As

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 fin

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.

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

[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

[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

[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 Analo

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

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

[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 -- C

[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 woul

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 _

[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 Col

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 _

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-process

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 fi

[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 ___

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 maili

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.

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.daylightSaving

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 di

[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 s

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

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 direc

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

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

[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

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

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 reb

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

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

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

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

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

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 specia

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

[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 yo

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, Ri

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

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

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

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

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 wi

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 fe

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 w

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 troublesho

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

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

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

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 wi

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 -- proporti

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 th

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 menti

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 h

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 Ac

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

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.co

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 ___ As

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 harm

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. T

Re: AW: [Asterisk-Users] CLIP in Germany

Johannes von Drachenfels wrote: Hi, i have a Siemens Gigaset phone with clip-capabilities on the tdm400 with extension 610 installed. Extension 600 is another phone. But i still cannot see the number of the caller at the display of the Siemens phone... here is my config: ; Main extension exten =

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

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 excell

[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 dist

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 ro

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

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 te

[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 r

Re: [Asterisk-Users] te410p with serial console fails with error:TE410P: Double/missed interrupt detected

Mark Spencer wrote: I am having trouble getting the te410p to work with a serial console. Has anyone else had this problem? Am I doing somthing wrong? Interupt problem? Is this a bug? The serial console likely is blocking interrupts for too long when it engages. Zaptel requires interrupts be

Re: [Asterisk-Users] FXO mode

Dave Cotton wrote: So the question is does changing to CTR21 do anything at all with the existing cards? No as far as enabling CTR21 features, but the bits addressed in that register are labeled "Reserved", so to be correct the card should be left in the default FCC mode. If it does nothing t

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

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

[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 a

[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 N

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

[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

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