Re: [Asterisk-Users] fax detection: false positive
Tilghman, What happens if someone needs the new signalling routines *and* working fax detection? I'm personally not in this boat, but it's only a matter of time before someone is. Is this a temporary fix? If not, this should be documented somewhere as it seems to be a problem for enough people. (Olle? :) Thanks, Pat - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 26, 2003 2:50 PM Subject: Re: [Asterisk-Users] fax detection: false positive On Friday 26 December 2003 13:42, john lawler wrote: Hi guys, I just moved from Asterisk release 0.5.0 to CVS 2003-12-22, and after overcoming a few changes in my configuration, I encountered one problem that I couldn't shake that was working fine in 0.5.0. It's the fax detection. I just have a simple extension setup like this: exten = fax,1,Dial(Zap/4,30,tr) exten = fax,2,Hangup in my main incoming context. This used to work fine, I don't think I ever had a false positive or negative, but now just about every call (possibly every call) that comes in when I've got that extension defined rolls to my fax machine on Zap/4 immediately. Uncomment OLD_DSP_ROUTINES near the top of dsp.c, recompile, install, and restart. The newer DSP routines are used to fix a type of signalling on EM lines. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call with bad/choppy sound
Ing. Angel Gomez Garcia wrote: Hi all. I have this configuration: Telco -(E1)-TE410P//Dual Xeon Server 2.4Ghz-(Ethernet)-Switch-GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS receives calls it sounds great. I have the exact same problem with the choppy sound when a call is originated from the GS phone to the PSTN (X100P).. Recieving calls if fine and calling other extensions is fine.. I have had this issue for a while now and have not been able to solve it.. I have tried beta firmware on the GS phone and I have kept to the latest asterisk CVS version but the problem remains.. Hopefully someone will have a solution.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql cdrs
Brian West wrote: cdr_odbc is for logging CDR data to a database. Its pretty much blind to the type of database you choose as long as it has an ODBC driver. We had it speaking to an AS/400 running DB2... we also have it working with MSSQL (not my goal but hey it works), mysql, pgsql and flatfiles. I have yet to hear it works with oracle (anyone out there test this?) bkw Brian, I see the cdr_odbc stuff is now in the CVS but I did not see a sample config file in the configs directory of the CVS.. Am I mising somthing? How is cdr_odbc configured? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] frame buffering
Tilghman Lesher wrote: On Saturday 27 December 2003 16:42, Steven Critchfield wrote: On Sat, 2003-12-27 at 16:28, Ing. Angel Gomez Garcia wrote: James Sharp wrote: Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. How do I verify that my console is running frame buffering ? and mos important, How do I disable it ? What should I do to run a plain VGA console ? Is there a penguin in the upper left when it boots, or some other graphic? If so your in a frame buffer. To disable requires recompiling the kernel and removing the option. Actually, it's even easier than that to disable: If you're using LILO: 1) Edit lilo.conf and remove any line that begins with vga= 2) run /sbin/lilo -v 3) reboot Thank you! http://www.voip-info.org/tiki-index.php?page=Asterisk+disable+frame+buffer Linked from the FAQ. http://www.voip-info.org/wiki-Asterisk+FAQ /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql cdrs
WipeOut wrote: I see the cdr_odbc stuff is now in the CVS but I did not see a sample config file in the configs directory of the CVS.. Am I mising somthing? How is cdr_odbc configured? http://www.voip-info.org/wiki-Asterisk+cdr+odbc /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple mpg123 processes when starting asterisk
When I start asterisk, it appears that multiple mpg123 processes start. Would this be normal operation? 2729 ?S 0:00 /usr/sbin/asterisk 2735 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 av-1.mp3 2736 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 av-1.mp3 2740 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -z av-1.mp3 2742 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -z av-1.mp3 Someone already mentioned two occurences of the above for each music on hold class defined (by you) in asterisk. In some cases, you may also find additional occurences of the above running if asterisk (and mpg123) did not shut down correctly. The problem is related to how asterisk kills mpg123 when asterisk is told to stop. Search the archives for SIG TERM is the last 60 days. (Or, simply stop asterisk and check to see if any remaining mpg123 processes are running. If so, kill them.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] frame buffering
For Olle-Wiki: Also in Grub you can pass parameters to kernel: 1) edit /boot/grub/menu.lst 2) find the command that loads kernel, e.g. something like this: kernel (hd0,1)/boot/vmlinuz root=/dev/hda2 vga=0x317:off splash=silent showopts 3) change the parameter vga=... to vga=normal 4) save and reboot --Markku -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Sunday, December 28, 2003 10:34 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] frame buffering Tilghman Lesher wrote: On Saturday 27 December 2003 16:42, Steven Critchfield wrote: On Sat, 2003-12-27 at 16:28, Ing. Angel Gomez Garcia wrote: James Sharp wrote: Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. How do I verify that my console is running frame buffering ? and mos important, How do I disable it ? What should I do to run a plain VGA console ? Is there a penguin in the upper left when it boots, or some other graphic? If so your in a frame buffer. To disable requires recompiling the kernel and removing the option. Actually, it's even easier than that to disable: If you're using LILO: 1) Edit lilo.conf and remove any line that begins with vga= 2) run /sbin/lilo -v 3) reboot Thank you! http://www.voip-info.org/tiki-index.php?page=Asterisk+disable+frame+buffer Linked from the FAQ. http://www.voip-info.org/wiki-Asterisk+FAQ /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual Athlon 2.4 MP *
I have a dual AMD running a hardware RAID card and a bunch of disks. Recently it wouldn't boot unless I removed the RAID card. Turns out the power supply had degraded to supply about 4.7V (instead of 5.0) and that was enough for it to fail during POST. Even replacing the PS with a new ATX did not solve the problem. I had to purchase a new EPS to solve it. This was a tough one. Mike Brian West wrote: Take a glance at bugs 714 thru 722 on bugs.digium.com I feel this is a local hardware issue. Has anyone else ran on a dual amd box ? Could his power supply be too weak? I don't know of anyone that has 9 totally diffrent and totally random crashes in asterisk in one day. Anyone care to input on this? Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SHARING DIGITAL CONTENT
I am in a development cycle for a telephony service based on Asterisk, and a question has occurred to me: What about sharing/transmitting digital content? Would it be possible, for example, to share a photo in a conference call between the newer digital cell phones (which have integrated cameras)? Is this just passing data through the channel, or are there significant technical obstacles? This capability will become useful in the not-so-distant future, and a PBX exhibiting this feature would obsolete many current systems. Charlie Hatchette [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] frame buffering
Markku Korpi wrote: For Olle-Wiki: Also in Grub you can pass parameters to kernel: 1) edit /boot/grub/menu.lst 2) find the command that loads kernel, e.g. something like this: kernel (hd0,1)/boot/vmlinuz root=/dev/hda2 vga=0x317:off splash=silent showopts 3) change the parameter vga=... to vga=normal 4) save and reboot Great! http://www.voip-info.org/tiki-index.php?page=Asterisk+disable+frame+buffer updated. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SHARING DIGITAL CONTENT
On Sun, 2003-12-28 at 11:55, Charles Hatchette wrote: I am in a development cycle for a telephony service based on Asterisk, and a question has occurred to me: What about sharing/transmitting digital content? Would it be possible, for example, to share a photo in a conference call between the newer digital cell phones (which have integrated cameras)? Is this just passing data through the channel, or are there significant technical obstacles? This capability will become useful in the not-so-distant future, and a PBX exhibiting this feature would obsolete many current systems. Cell phones share pictures via MMS. MMS is like SMS, and is not call based. It is just a messaging protocol. So the only thing you could do with the meetme app that might help send the pictures is collect the callerid from the users and provide that to a parallel app that dealt with the sending. If you use VoIP, H323 and IAX support pictures and video. While I have heard of a VoIP app for my cell phone, the data costs would be way to high to use it instead of a normal call. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SHARING DIGITAL CONTENT
If you want to send mms, one solution would be to link (*) with kannel (a sms, mms, wap platform, see www.kannel.org). With kannel, you have 2 solutions: -small use : connect a gsm modem with your computer -large use : connect to a sms center trough tcp/ip In both cases, you will have some transfert fees, in the first case with you mobile phone bill and in the second case you will receive a bill from the sms center. And finally, in both cases, you will be able to receive data (wap, sms, mms) on your platform and manage it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: dimanche 28 décembre 2003 19:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SHARING DIGITAL CONTENT On Sun, 2003-12-28 at 11:55, Charles Hatchette wrote: I am in a development cycle for a telephony service based on Asterisk, and a question has occurred to me: What about sharing/transmitting digital content? Would it be possible, for example, to share a photo in a conference call between the newer digital cell phones (which have integrated cameras)? Is this just passing data through the channel, or are there significant technical obstacles? This capability will become useful in the not-so-distant future, and a PBX exhibiting this feature would obsolete many current systems. Cell phones share pictures via MMS. MMS is like SMS, and is not call based. It is just a messaging protocol. So the only thing you could do with the meetme app that might help send the pictures is collect the callerid from the users and provide that to a parallel app that dealt with the sending. If you use VoIP, H323 and IAX support pictures and video. While I have heard of a VoIP app for my cell phone, the data costs would be way to high to use it instead of a normal call. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Compatible Hardware Components?
Hello, Can someone tell me which hardware components are compatible with the system? If possible, a complete spec of a computer setup would be preferable. I need to get two systems and would like to make sure the hardware will work properly. We would probably get a 2ghz single processor for the CPU. Thanks. Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex Codec - Error IAX2
Hi, I am trying to connect two * using Speex codec via IAX2. When it starts connection I get an error message : -- Format for call is SPEEX NOTICE[98311]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ALAW to SPEEX NOTICE[98311]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from SPEEX to ALAW Any clue? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Compatible Hardware Components?
I have had damn good luck with dell. Stay away from AMD MP boxes or stay away from AMD totally ( I'm not knocking AMD I just don't trust it in a server) bkw On Sun, 28 Dec 2003, Stephen Karrington wrote: Hello, Can someone tell me which hardware components are compatible with the system? If possible, a complete spec of a computer setup would be preferable. I need to get two systems and would like to make sure the hardware will work properly. We would probably get a 2ghz single processor for the CPU. Thanks. Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speex Codec - Error IAX2
Did you install speex? Its not there by default and you must build extra libs for the codec to work. www.speex.org bkw On Sun, 28 Dec 2003, Daniel Bichara wrote: Hi, I am trying to connect two * using Speex codec via IAX2. When it starts connection I get an error message : -- Format for call is SPEEX NOTICE[98311]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ALAW to SPEEX NOTICE[98311]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from SPEEX to ALAW Any clue? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Compatible Hardware Components?
Please let us know what you want to accomplish, and how large your telephone configuration is. Are you supporting 4 channels, or 400? etc etc. Then someone with that size configuration would be better able to assist you. Regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Karrington Sent: Sunday, December 28, 2003 7:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Compatible Hardware Components? Hello, Can someone tell me which hardware components are compatible with the system? If possible, a complete spec of a computer setup would be preferable. I need to get two systems and would like to make sure the hardware will work properly. We would probably get a 2ghz single processor for the CPU. Thanks. Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID trunks -- equipment requirement
Hi guys, I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1 correspondence. Since I'm in a less populated area of the country, SBC doesn't seem to have much in the way of fractional T1 products (on the scale that we need them) available, so I think my only option for DID is to use (analog) DID trunks for incoming calls and POTS lines for outbound calls. I'd look into ISDN, both PRI and BRI. If the costs are not too prohibitive, this would be the most flexable option. ISDN uses out of band signaling and has a number of features which complement a DID enviroment, such as DNIS (dialed number information service), where the number dialed is passed along with an incoming call. If your enviroment never uses all its outside lines at the same time, this can be cost effective, because you can have direct dial numbers for all the phones without a one to one correspondance of outside lines to extensions. PRI is usually too expensive, but sometimes BRI is affordable, however you should check on pricing for both to see if they may be cost effective. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Compatible Hardware Components?
On Sun, 2003-12-28 at 13:05, Stephen Karrington wrote: Hello, Can someone tell me which hardware components are compatible with the system? If possible, a complete spec of a computer setup would be preferable. I need to get two systems and would like to make sure the hardware will work properly. If you can't be troubled with reading and thinking, please hire a consultant that will do it for you. Better yet, consult the Digium website for a consultant that already knows this project. If you choose to spend time reading, then come back here with the new knowledge to ask a better, more informed, direct question. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Compatible Hardware Components?
We are starting out with two ISDN lines. That is 4 phone lines in total. We plan to expand up to 4 ISDN channels which is 8 phone lines in total. The first 2 ISDN lines will have approximately 5-8 extensions. Any suggestions on a softphone and physical phone would also be greatly appreciated. We are located in Eastern Europe, which means some hardware brands may not be too easy or economical to get. Thanks. S Please let us know what you want to accomplish, and how large your telephone configuration is. Are you supporting 4 channels, or 400? etc etc. Then someone with that size configuration would be better able to assist you. Regards Scott M. Stingel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outcall notification
Has anyone implemented an outcall notification when there is a voice message waiting? I would like to have the system notify me of awaiting voice messages by a telephone call rather than an email notification. I would imagine that a call could be dumped into the asterisk spool directory, but Im not sure how I would monitor for messages waiting. Has anyone implemented such a feature for asterisk? I did a google and wiki search with no information available. Thanks
Re: [Asterisk-Users] outcall notification
Maybe you just need to dump a file to the spool directory that has your phone number and an asterisk extension that goes to a voicemail check. You'd still need to patch app_voicemail to create the call file. Iain --On Sunday, December 28, 2003 4:07 pm -0500 Kevin [EMAIL PROTECTED] wrote: Has anyone implemented an outcall notification when there is a voice message waiting? I would like to have the system notify me of awaiting voice messages by a telephone call rather than an email notification. I would imagine that a call could be dumped into the asterisk spool directory, but I'm not sure how I would monitor for messages waiting. Has anyone implemented such a feature for asterisk? I did a google and wiki search with no information available. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo, ISDN And FXS
Hello Matteo, I use ISDN with AVM C2, also fritz pci with tdm fxs sip phone without any echo. what isdn channel driver are you using? I suggest using the avm with capi+chan_capi-0.3.0 and turn on echosquelch in capi.conf That is my configuration (0.3.0, echosquelch=1 in the general section of the capi.conf What version of * are you running, and did you do any changes to the Makefile? rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] specify maximum call duration
Hi, is it possible to specify a maximum call duration for a peer or an extention? David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium Wildcat E100 card mechanics issue
Hello, Anybody knows how to convert from RJ45 (E100) to BNC (g703) for E1 links? And/or where to buy it? Thanks.
Re: [Asterisk-Users] specify maximum call duration
show application AbsoluteTimeout On Sun, 28 Dec 2003, David Luyens wrote: Hi, is it possible to specify a maximum call duration for a peer or an extention? David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Wildcat E100 card mechanics issue
A search on Yahoo brought up quite a few RJ45-BNC cable sets -Original Message-From: Hector Q.-datafull [mailto:[EMAIL PROTECTED]Sent: Sunday, December 28, 2003 5:20 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Digium Wildcat E100 card mechanics issue Hello, Anybody knows how to convert from RJ45 (E100) to BNC (g703) for E1 links? And/or where to buy it? Thanks.
Re: [Asterisk-Users] specify maximum call duration
David Luyens wrote: Hi, is it possible to specify a maximum call duration for a peer or an extention? Before posting a question like this you should read all the docs and type Show Applications in the asterisk CLI. This way you would have noticed an application called 'AbsoulteTimeout' Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcat E100 card mechanics issue
On Sun, 2003-12-28 at 16:20, Hector Q.-datafull wrote: Hello, Anybody knows how to convert from RJ45 (E100) to BNC (g703) for E1 links? And/or where to buy it? Thanks. The archives document this well. http://www.marko.net/asterisk/archives/0208/0370.html -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium Wildcat E100 card mechanics issue
RJ-45 connectors use 120 ohm connections, and BNC are 75 ohm. You need to use converters, called Baluns, to change the impedance. A single balun will convert the separate BNC transmit and receive signals to the combined connections on the RJ-45. Regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hector Q.-datafull Sent: Sunday, December 28, 2003 10:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Digium Wildcat E100 card mechanics issue Hello, Anybody knows how to convert from RJ45 (E100) to BNC (g703) for E1 links? And/or where to buy it? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] specify maximum call duration
Hi! Hi, is it possible to specify a maximum call duration for a peer or an extention? Before posting a question like this you should read all the docs and type Show Applications in the asterisk CLI. This way you would have noticed an application called 'AbsoulteTimeout' Hehe... read the question again before jumping on a new user - the answer to the peer part of it is No instead AbsoluteTimeout. ;- Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo, ISDN And FXS
Hi! I have an installation utilizing * with an AVM C4 (ISDN card). Using softphones (SIP and IAX) I have sound problems, like echo. Did you test if a good soundcard and especially a really good headset make a difference? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] specify maximum call duration
On Sun, 2003-12-28 at 17:04, Philipp von Klitzing wrote: Hi! Hi, is it possible to specify a maximum call duration for a peer or an extention? Before posting a question like this you should read all the docs and type Show Applications in the asterisk CLI. This way you would have noticed an application called 'AbsoulteTimeout' Hehe... read the question again before jumping on a new user - the answer to the peer part of it is No instead AbsoluteTimeout. ;- It would be possible since you could select a callerid string for the peer in it's definition, and then do a match on it. It takes just a hair more thinking about the problem. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Compatible Hardware Components?
Stephen Karrington wrote: We are starting out with two ISDN lines. That is 4 phone lines in total. We plan to expand up to 4 ISDN channels which is 8 phone lines in total. The first 2 ISDN lines will have approximately 5-8 extensions. Any suggestions on a softphone and physical phone would also be greatly appreciated. We are located in Eastern Europe, which means some hardware brands may not be too easy or economical to get. Thanks. S Please let us know what you want to accomplish, and how large your telephone configuration is. Are you supporting 4 channels, or 400? etc etc. Then someone with that size configuration would be better able to assist you. Lots of reading here: http://www.voip-info.org/wiki-Asterisk+hardware+recommendations /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Error
Hello, On the Polycom IP 500 Phones, when I press the mic mute button, the mic on the speaker or headset goes muted. However when I press the mic mute button again, the call is terminated by asterisk. Asterisk shows a: WARNING[1236268096]: File channel.c, Line 1296 (do_senddigit): Unable to handle DTMF tone 'f' for 'SIP/-' I am using reinvite=no on the phones. After looking through the C file, it is clear where DTMF tones are declared, however, I want to make this one silent. Is there anyway I could alter this file, to simply ignore the tone and continue the call. Also, from a design perspective, should Asterisk natively be more lenient with tones that it doesn't understand, rather than just cutting the call off? Not sure if this would be considered a bug or whether there was some functionality with this that I just am missing? Thanks, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] specify maximum call duration
Hi! Hi, is it possible to specify a maximum call duration for a peer or an extention? Before posting a question like this you should read all the docs and type Show Applications in the asterisk CLI. This way you would have noticed an application called 'AbsoulteTimeout' Hehe... read the question again before jumping on a new user - the answer to the peer part of it is No instead AbsoluteTimeout. ;- It would be possible since you could select a callerid string for the peer in it's definition, and then do a match on it. It takes just a hair more thinking about the problem. For the sake of discussion: What if the peer is a * connected through IAX, and the user wants to dial an extension on that * box? As of now you need to specify AbsoluteTimeout for each extension or macro... this can be very tedious depening on the layout of your dialplan. I _do_ think it would make sense to be able to define such a timeout for a specific peer, or even as a global varaible valid for the entire * server. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID trunks -- equipment requirement
quote who=Josh Rollyson I'd look into ISDN, both PRI and BRI. If the costs are not too prohibitive, this would be the most flexable option. ISDN uses out of band signaling and has a number of features which complement a DID enviroment, such as DNIS (dialed number information service), where the number dialed is passed along with an incoming call. If your enviroment never uses all its outside lines at the same time, this can be cost effective, because you can have direct dial numbers for all the phones without a one to one correspondance of outside lines to extensions. PRI is usually too expensive, but sometimes BRI is affordable, however you should check on pricing for both to see if they may be cost effective. I wish SBC would offer DID via BRI. DID is only available via PRI or individule analog trunks. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Incoming callers aren't hearing ring
Yes. Add a Ringing command. exten = _5551212,1,Answer exten = _5551212,2,Ringing exten = _5551212,3,Dial(SIP/6710,12,tr) Ok, extensions.conf now contains: [incoming] include = sip-phones exten = _5551212,1,Answer exten = _5551212,2,Ringing exten = _5551212,2,Dial(SIP/6710,12,tr) ... etc. and still I am not getting a ring indicated to the caller. It used to work fine with the fxo cards. Is it possible that there is something wrong with the way asterisk and the switch at the CLEC are communicating ring to each other? I appreciate the suggestions thus far, any other ideas about what could be causing my problem? Thanks terry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with x101P
- Original Message - From: Burak Balasaygun [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 27, 2003 10:46 PM Subject: RE: [Asterisk-Users] Help with x101P snip I'm not sure what you mean by what type of switch you are connected to? The x101p is connected to the CO switch for my LEC. Occasionally I do NPA-NXX lookups for my local exchanges to see what other carriers have prefixes in my area. I used to use telcodata.us, but they seem to have gone offline. Usually, after you find the carrier's name, you can see info on the location and type of switch being used. I can't say with any assurity that the info is accurate, but it is there if you dig. Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with x101P
Occasionally I do NPA-NXX lookups for my local exchanges to see what other carriers have prefixes in my area. I used to use telcodata.us, but they seem to have gone offline. Usually, after you find the carrier's name, you can see info on the location and type of switch being used. I can't say with any assurity that the info is accurate, but it is there if you dig http://www.dslreports.com/coinfo Has the same kinda info. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] specify maximum call duration
On Sun, 2003-12-28 at 17:30, Philipp von Klitzing wrote:
Hi!
Hi, is it possible to specify a maximum call duration for a peer or an
extention?
Before posting a question like this you should read all the docs and
type Show Applications in the asterisk CLI.
This way you would have noticed an application called 'AbsoulteTimeout'
Hehe... read the question again before jumping on a new user - the answer
to the peer part of it is No instead AbsoluteTimeout. ;-
It would be possible since you could select a callerid string for the
peer in it's definition, and then do a match on it. It takes just a hair
more thinking about the problem.
For the sake of discussion: What if the peer is a * connected through
IAX, and the user wants to dial an extension on that * box? As of now you
need to specify AbsoluteTimeout for each extension or macro... this can
be very tedious depening on the layout of your dialplan.
To quote you, read the question again... Specifically it is max call
duration for a _PEER_ or _EXTENSION_. On a per extension basis, this is
no more tedious than the initial setup. On a per peer basis, well that
is just a matter of setting it on the initial allowed context they are
allowed in. It would have to be defined somewhere, and isn't any more
tedious than any other method of definition.
For your extended part of the original question.
[incoming]
exten = _.*,1,answer
;exten = _.*,2,agi(timeout-lookup.agi) ; alternative
exten = _.*/some match,2,Absolutetimeout(360)
exten = _.*,2,noop
exten = _.*,3,goto(realcontext,${EXTEN},1)
Does this look hard or tedious?
I _do_ think it would make sense to be able to define such a timeout for
a specific peer, or even as a global varaible valid for the entire *
server.
Maybe you don't yet grasp how easy things can already be done. Only
thing I think might be nice, but I don't really see a use for yet would
be a method to define extra variables on any of the VoIP user
definitions.
--
Steven Critchfield [EMAIL PROTECTED]
___
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RE: [Asterisk-Users] DID trunks -- equipment requirement
On Sun, 2003-12-28 at 17:52, Robert Hajime Lanning wrote: quote who=Josh Rollyson I'd look into ISDN, both PRI and BRI. If the costs are not too prohibitive, this would be the most flexable option. ISDN uses out of band signaling and has a number of features which complement a DID enviroment, such as DNIS (dialed number information service), where the number dialed is passed along with an incoming call. If your enviroment never uses all its outside lines at the same time, this can be cost effective, because you can have direct dial numbers for all the phones without a one to one correspondance of outside lines to extensions. PRI is usually too expensive, but sometimes BRI is affordable, however you should check on pricing for both to see if they may be cost effective. I wish SBC would offer DID via BRI. DID is only available via PRI or individule analog trunks. DID is the wrong name for ISDN. On BRI I think you want to ask for multiple MSN with rollover. This may be the time where you start asking around the CLECs to see what they call the features in your area. Oddly enough on the few circuits I have been part of the ordering process, it took a while before the sales man, the tech, and myself found a common language to define what was available and what we wanted. It may also be a good idea to contact your PSC if you are told you can't get what you want. If your PSC is worth anything, they will be able to point you in a decent direction, or put pressure on SBC to provide you what you want. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with x101P
Andrew Thompson wrote: - Original Message - From: Burak Balasaygun [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 27, 2003 10:46 PM Subject: RE: [Asterisk-Users] Help with x101P snip I'm not sure what you mean by what type of switch you are connected to? The x101p is connected to the CO switch for my LEC. Occasionally I do NPA-NXX lookups for my local exchanges to see what other carriers have prefixes in my area. I used to use telcodata.us, but they seem to have gone offline. Usually, after you find the carrier's name, you can see info on the location and type of switch being used. I can't say with any assurity that the info is accurate, but it is there if you dig. http://www.telcoexchange.com/resources/carriers/index.shtml will tell you the switch and other stuff when you put in a phone number. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM Card loses Dialtone and Battery
Hello all, I've posted on this problem before. Well here goes again. I have an intel mobo with a p4 2.4ghz proc, 1GB Ram. It has built-in ethernet and vga and 6 pci slots. I dreamed of making this my household communications server: internet router, firewall, vpn and asterisk. Everything works except the TDM fxs card. Well it works for a little while and it dies: no dialtone, no ring tone. All 6 slots are filled: two more Ethernet cards, two digium fxo cards, an sb live card and the tdm card. Everything that I don't use on the motherboard is turned off: serial and parallel ports, serial ata and motherboard sound. I've got all this stuff packed in a case with a 430 watt power supply. Interesting observation #1: When the tdm card dies, the fxo cards and asterisk still carry on. People can call and can leave messages, etc. I just can't hear the phone ring and I can't use the phone either. Interesting observation #2: I think I know how to make the tdm card die. I have a pc behind one of the Ethernet cards on the server. When I do a download off the net, the tdm card dies. Keep in mind when I'm doing a download two Ethernet interfaces are working, the one to which the pc is connected and the one connected to my cable modem. I've just tried another download - I'm almost 100 percent sure I can make the card die this way. Anyone been down this path before? I'd hate to buy a linksys box just to make the tdm card happy.
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
- Original Message - From: Victor Rini [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 28, 2003 8:48 PM Subject: [Asterisk-Users] TDM Card loses Dialtone and Battery snip I dreamed of making this my household communications server: internet router, firewall, vpn and asterisk. Everything works except the TDM fxs card. Well it works for a little while and it dies: no dialtone, no ring tone. Does asterisk start dumping grotesque error messages when the card dies? Checked /proc/interrupts? What about load, while you're downloading? (I realize the error is permanent, even though the load isn't.) All 6 slots are filled: two more Ethernet cards, two digium fxo cards, an sb live card and the tdm card. Everything that I don't use on the motherboard Side note, and probably not related, but what's the SB live card for? You don't actually use this computer, do you? It's a server, let it be one... Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
At 06:53 PM 12/28/2003, you wrote: Side note, and probably not related, but what's the SB live card for? You don't actually use this computer, do you? It's a server, let it be one... Asterisk requires a timing source to play music on hold and conference VoIP channels. The SB performs this function. However, I thought an fxo card was supposed to provide timing... --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
the zap cards, (fxo, fxs, etc) all provide timing, I dont know HOW asterisk is getting timing from the soundcard... if one doesnt have a zap card, one uses ztdummy, which gets timing from the usb... Ernest W. Lessenger wrote: At 06:53 PM 12/28/2003, you wrote: Side note, and probably not related, but what's the SB live card for? You don't actually use this computer, do you? It's a server, let it be one... Asterisk requires a timing source to play music on hold and conference VoIP channels. The SB performs this function. However, I thought an fxo card was supposed to provide timing... --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
On Sun, 2003-12-28 at 19:39, [EMAIL PROTECTED] wrote: All 6 slots are filled: two more Ethernet cards, two digium fxo cards, an sb live card and the tdm card. Everything that I don't use on the motherboard is turned off: serial and parallel ports, serial ata and motherboard sound. I've got all this stuff packed in a case with a 430 watt power supply. Interesting observation #1: When the tdm card dies, the fxo cards and asterisk still carry on. People can call and can leave messages, etc. I just can't hear the phone ring and I can't use the phone either. Interesting observation #2: I think I know how to make the tdm card die. I have a pc behind one of the Ethernet cards on the server. When I do a download off the net, the tdm card dies. Keep in mind when I'm doing a download two Ethernet interfaces are working, the one to which the pc is connected and the one connected to my cable modem. I've just tried another download - I'm almost 100 percent sure I can make the card die this way. Anyone been down this path before? I'd hate to buy a linksys box just to make the tdm card happy. It seems somewhat obvious-- I'll bet, if you look at (via 'cat /proc/interrupts') the interrupts, you'll see that the tdm shares an interrupt line with one of the ethernet cards. All it takes for something like this to happen, is for the ethernet card to take longer than 1/8000th second to handle it's interrupt. Solution? Work with the plug play BIOS settings, and maybe move boards around in the slots, so you can the tdm on an interrupt line all by itself. Maybe, make another machine the network gateway, use your asterisk machine for VOIP phones only. You'll have to experiment. murf signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
- Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 28, 2003 9:49 PM Subject: Re: [Asterisk-Users] TDM Card loses Dialtone and Battery Asterisk requires a timing source to play music on hold and conference VoIP channels. The SB performs this function. However, I thought an fxo card was supposed to provide timing... --Ernest I am aware of the timing source issues, but was not aware that the SB live could do this. The wiki does not currently mention it, google doesn't find anything, and I've not seen it mentioned in the few months that I've been subscribed to the list. Could someone post some info regarding this to the list, and/or the wiki so it can be documented? http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer I'm curious what module to load, and how it compares to ztdummy/zaprtc/a real zap interface. Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Hello again, Thanks for the timely responses. Andrew: Asterisk doesn't dump any messages except when a call comes in and asterisk tries to ring an extension - it leaves a device busy type of message. I checked /proc/interrupts. The fxs card is still there after it dies, but the interrupts counter does not change over time. When the fxs card is working it is usually constantly firing interrupts. I'll check load and report back. Thanks for the suggestion about the sound card. I really don't need it in the server. I'll take it out. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
On Sunday 28 December 2003 19:48, Victor Rini wrote: I've posted on this problem before. Well here goes again. I have an intel mobo with a p4 2.4ghz proc, 1GB Ram. It has built-in ethernet and vga and 6 pci slots. I dreamed of making this my household communications server: internet router, firewall, vpn and asterisk. Everything works except the TDM fxs card. Well it works for a little while and it dies: no dialtone, no ring tone. All 6 slots are filled: two more Ethernet cards, two digium fxo cards, an sb live card and the tdm card. Everything that I don't use on the motherboard is turned off: serial and parallel ports, serial ata and motherboard sound. I've got all this stuff packed in a case with a 430 watt power supply. Interesting observation #1: When the tdm card dies, the fxo cards and asterisk still carry on. People can call and can leave messages, etc. I just can't hear the phone ring and I can't use the phone either. Interesting observation #2: I think I know how to make the tdm card die. I have a pc behind one of the Ethernet cards on the server. When I do a download off the net, the tdm card dies. Keep in mind when I'm doing a download two Ethernet interfaces are working, the one to which the pc is connected and the one connected to my cable modem. I've just tried another download - I'm almost 100 percent sure I can make the card die this way. Anyone been down this path before? I'd hate to buy a linksys box just to make the tdm card happy. Which revision of the card is this? Is it the Revision C or before (with no molex connection) or the Revision E or later (with a molex connection)? I've had this problem before with the Revision C, but the Revision E/F cleared up the problem. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery
I think what Steve was getting at was interrupt sharing. Is the fxs card on the same interrupt as anything else? Sean -Original Message- From: Victor Rini [mailto:[EMAIL PROTECTED] Sent: Sunday, December 28, 2003 10:21 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery Hello again, Thanks for the timely responses. Andrew: Asterisk doesn't dump any messages except when a call comes in and asterisk tries to ring an extension - it leaves a device busy type of message. I checked /proc/interrupts. The fxs card is still there after it dies, but the interrupts counter does not change over time. When the fxs card is working it is usually constantly firing interrupts. I'll check load and report back. Thanks for the suggestion about the sound card. I really don't need it in the server. I'll take it out. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there something wrong with show manager commands?
Is it just my box, or is there something flaky in the implementation of show manager commands? Note: I'm using putty. About half way through this, I toggled my KVM over to the desktop and logged in to try and recreate it. The output was the same as the last two entries in this dump. bebop*CLI show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager bebop*CLHangup Hangup Channel bebop*CLStatus Status bebop*CLRedirect Redirect bebop*CLOriginate Originate Call bebop*CLMailboxStatus Check Mailbox bebop*CLCommand Execute Command bebop*CLExtensionState Check Extension Status bebop*CLAbsoluteTimeout Set Absolute Timeout bebop*CLMailboxCount Check Mailbox Message Count bebop*CLMonitor Monitor a channel bebop*CLStopMonitor Stop monitoring a channel bebop*CLChangeMonitor Change monitoring filename of a channel bebop*CLZapTransfer Transfer Zap Channel bebop*CLZapHangup Hangup Zap Channel bebop*CLZapDialOffhook Dial over Zap channel while offhook bebop*CLIAXpeers List IAX Peers bebop*CLIAXpeers List IAX Peers bebop*CLQueues Queues bebop*CLQueueStatus Queue Status bebop*CLI show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager bebop*CLHangup Hangup Channel bebop*CLStatus Status bebop*CLRedirect Redirect bebop*CLOriginate Originate Call bebop*CLMailboxStatus Check Mailbox bebop*CLCommand Execute Command bebop*CLExtensionState Check Extension Status bebop*CLAbsoluteTimeout Set Absolute Timeout bebop*CLMailboxCount Check Mailbox Message Count bebop*CLMonitor Monitor a channel bebop*CLStopMonitor Stop monitoring a channel bebop*CLChangeMonitor Change monitoring filename of a channel bebop*CLZapTransfer Transfer Zap Channel bebop*CLZapHangup Hangup Zap Channel bebop*CLZapDialOffhook Dial over Zap channel while offhook bebop*CLIAXpeers List IAX Peers bebop*CLIAXpeers List IAX Peers bebop*CLQueues Queues bebop*CLQueueStatus Queue Status -- Remote UNIX connectionnds -- Remote UNIX connection disconnected bebop*CLPing Ping Logoff Logoff Manager Hangup Hangup Channel Status Status Redirect Redirect Originate Originate Call MailboxStatus Check Mailbox Command Execute Command ExtensionState Check Extension Status AbsoluteTimeout Set Absolute Timeout MailboxCount Check Mailbox Message Count Monitor Monitor a channel StopMonitor Stop monitoring a channel ChangeMonitor Change monitoring filename of a channel ZapTransfer Transfer Zap Channel ZapHangup Hangup Zap Channel ZapDialOffhook Dial over Zap channel while offhook IAXpeers List IAX Peers IAXpeers List IAX Peers Queues Queues QueueStatus Queue Status bebop*CLI bebop*CLI bebop*CLI bebop*CLI show manager commands bebop*CLPing Ping Logoff Logoff Manager Hangup Hangup Channel Status Status Redirect Redirect Originate Originate Call MailboxStatus Check Mailbox Command Execute Command ExtensionState Check Extension Status AbsoluteTimeout Set Absolute Timeout MailboxCount Check Mailbox Message Count Monitor Monitor a channel StopMonitor Stop monitoring a channel ChangeMonitor Change monitoring filename of a channel ZapTransfer Transfer Zap Channel ZapHangup Hangup Zap Channel ZapDialOffhook Dial over Zap channel while offhook IAXpeers List IAX Peers IAXpeers List IAX Peers Queues Queues QueueStatus Queue Status bebop*CLI Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Steve, I have the tdm card on it's own IRQ. That's one of the first things I tried. Both of my fxo cards are on the same IRQ and they seem to hold together. It's interesting that you bring up the timing issue. Why would the tdm card be so sensitive? I can understand a drop in voice quality but dying? Another thought. Downloads are usually big tcp packets? Maybe 1500 bytes a packet? Processing them probably takes more time. I've run 300kbit streaming video through the server which I believe are smaller packets and the tdm card seems to hold up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery
- Original Message - From: Victor Rini [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 28, 2003 10:20 PM Subject: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery Hello again, Thanks for the timely responses. Andrew: Asterisk doesn't dump any messages except when a call comes in and asterisk tries to ring an extension - it leaves a device busy type of message. I don't have internal fx devices, so I'm scrambling to find a command to help see what * thinks is up/down. Can you restart asterisk with asterisk -vvvc, convince the card to crash, and see if it generates anything then? (If you've done this already, just say so.) I checked /proc/interrupts. The fxs card is still there after it dies, but the interrupts counter does not change over time. When the fxs card is working it is usually constantly firing interrupts. Sorry, I should have asked explicitly for what I wanted to see. Can you paste the contents of /proc/interrupts in your reply? I'll check load and report back. Thanks for the suggestion about the sound card. I really don't need it in the server. I'll take it out. The notable thing here is, your interrupts may move around when you take this card out. Your other * problems may get better or worse when this happens. There ought to be a better way to manage interrupts, but apart from taking boards out and trying them in varying orders, I don't know what it is. If you do take the sound card out, can you(just for fun) post the /proc/interrupts before and after? Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Tilghman, I have a feeling we're getting somewhere. I ordered three cards the very day they went on sale through the digium website. Yes, it's revision C. I guess I'll talk to digium about this. Thanks, Victor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM Card loses Dialtone and Battery
On Sun, 2003-12-28 at 21:44, Andrew Thompson wrote: - Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 28, 2003 9:49 PM Subject: Re: [Asterisk-Users] TDM Card loses Dialtone and Battery Asterisk requires a timing source to play music on hold and conference VoIP channels. The SB performs this function. However, I thought an fxo card was supposed to provide timing... --Ernest I am aware of the timing source issues, but was not aware that the SB live could do this. The wiki does not currently mention it, google doesn't find anything, and I've not seen it mentioned in the few months that I've been subscribed to the list. Could someone post some info regarding this to the list, and/or the wiki so it can be documented? http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer I'm curious what module to load, and how it compares to ztdummy/zaprtc/a real zap interface. It doesn't, thats why it isn't documented. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery
On Sunday 28 December 2003 21:53, Victor Rini wrote: I have a feeling we're getting somewhere. I ordered three cards the very day they went on sale through the digium website. Yes, it's revision C. I guess I'll talk to digium about this. In case you're wondering, the problem is the amount of power the TDM cards pull off the PCI bus. When you have another device sucking power, it can momentarily drop the power enough on the TDM card to reset it. The molex connector allows the card to pull power directly from the power supply instead of through the PCI bus, which of course solves that problem. I'm told the TDM Rev. C card is within the PCI spec for power drain, but the stresses it puts on the PCI bus will show how many motherboards are in fact close, but a little deficient The sole reason the TDM card requires so much power is to generate ring voltage for connected telephones. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Andrew: I tried the asterisk -vvvc suggestion and I didn't get any messages when the card died. Here's /proc/interrupts before I take out the sound card: CPU0 0: 102777IO-APIC-edge timer 1:471IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 9159IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17:1995769 IO-APIC-level wcfxo, wcfxo 18: 341396 IO-APIC-level wcfxs 19: 0 IO-APIC-level EMU10K1 20: 3390 IO-APIC-level eth1 21: 8652 IO-APIC-level eth0 22:788 IO-APIC-level eth2 NMI: 0 LOC: 102728 ERR: 0 MIS: 0 and after: CPU0 0: 14903IO-APIC-edge timer 1: 2IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 7469IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17: 111534 IO-APIC-level wcfxo 18: 111626 IO-APIC-level wcfxo 19: 104013 IO-APIC-level wcfxs 20:680 IO-APIC-level eth1 21:509 IO-APIC-level eth0 22: 41 IO-APIC-level eth2 NMI: 0 LOC: 14855 ERR: 0 MIS: 0 About load: almost impossible to tell. I was sshed into the server and running top - top was showing the system 100% idle. Then I hit a download link and bang, the card died. This is all pretty academic at this point - I think Tilghman found the problem for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery
now we're getting somewhere! anything above interrupt 15 will be interrupt sharing. bad! If you can get the cards assigned to 10 or 11, you should be in better shape. Sean -Original Message- From: Victor Rini [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 12:12 AM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery Andrew: I tried the asterisk -vvvc suggestion and I didn't get any messages when the card died. Here's /proc/interrupts before I take out the sound card: CPU0 0: 102777IO-APIC-edge timer 1:471IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 9159IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17:1995769 IO-APIC-level wcfxo, wcfxo 18: 341396 IO-APIC-level wcfxs 19: 0 IO-APIC-level EMU10K1 20: 3390 IO-APIC-level eth1 21: 8652 IO-APIC-level eth0 22:788 IO-APIC-level eth2 NMI: 0 LOC: 102728 ERR: 0 MIS: 0 and after: CPU0 0: 14903IO-APIC-edge timer 1: 2IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 7469IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17: 111534 IO-APIC-level wcfxo 18: 111626 IO-APIC-level wcfxo 19: 104013 IO-APIC-level wcfxs 20:680 IO-APIC-level eth1 21:509 IO-APIC-level eth0 22: 41 IO-APIC-level eth2 NMI: 0 LOC: 14855 ERR: 0 MIS: 0 About load: almost impossible to tell. I was sshed into the server and running top - top was showing the system 100% idle. Then I hit a download link and bang, the card died. This is all pretty academic at this point - I think Tilghman found the problem for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: TDM Card loses Dialtone and Battery
Sean, Yes, that IRQ assignment seemed strange to me too. I don't understand why the kernel wanted to assign IRQS this way. I guess it's something to do with this APIC technology. Can anyone fill me in here? By the way, thanks to everyone who has contributed to this thread. It's really helped a lot. Victor CPU0 0: 102777IO-APIC-edge timer 1:471IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 9159IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17:1995769 IO-APIC-level wcfxo, wcfxo 18: 341396 IO-APIC-level wcfxs 19: 0 IO-APIC-level EMU10K1 20: 3390 IO-APIC-level eth1 21: 8652 IO-APIC-level eth0 22:788 IO-APIC-level eth2 NMI: 0 LOC: 102728 ERR: 0 MIS: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery
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 [EMAIL PROTECTED] wrote: now we're getting somewhere! anything above interrupt 15 will be interrupt sharing. bad! If you can get the cards assigned to 10 or 11, you should be in better shape. Sean -Original Message- From: Victor Rini [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 12:12 AM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] RE: TDM Card loses Dialtone and Battery Andrew: I tried the asterisk -vvvc suggestion and I didn't get any messages when the card died. Here's /proc/interrupts before I take out the sound card: CPU0 0: 102777IO-APIC-edge timer 1:471IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 4IO-APIC-edge rtc 14: 9159IO-APIC-edge ide0 15: 6IO-APIC-edge ide1 17:1995769 IO-APIC-level wcfxo, wcfxo 18: 341396 IO-APIC-level wcfxs 19: 0 IO-APIC-level EMU10K1 20: 3390 IO-APIC-level eth1 21: 8652 IO-APIC-level eth0 22:788 IO-APIC-level eth2 NMI: 0 LOC: 102728 ERR: 0 MIS: 0 -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
