Re: [Asterisk-Users] Wireless phones connected to VOIP DECT base station
[EMAIL PROTECTED] wrote: On Sat, 30 Oct 2004, Remco Barende wrote: Digital Enhanced Cordless Telephones. The system is more or less similar to GSM (the mobile phone) that all the speech is transmitted digitally. Also you can have (similar to GSM) unnoticeable switching from one base station to another. Most wireless phones in Europe are DECT nowadays, unless you're looking for really cheap ones. In short DECT is a kick-ass cordless phone system. Up to 8 phones, intercom calls between them etc etc etc. Wish someone would offer a PCI board with base-station firmware and docs so I could do chan_dect.so There are PCI DECT Boards but I don't know if they will do base-station stuff. *http://tinyurl.com/3hnq5 Maybe somebody else has more insight into this if those PCI and PCMCIA Dect boards can be turned into base-stations. -- Thomas * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ?
vrushank wrote: ! p.s. maybe set your time/date correct ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending Caller ID info in MD/USA
William C. Lohr Jr. wrote: Marty, My business would own the actual T1, but I may provide an outbound call service for a client and would want their name sent as well as their toll free number and not the local number for the outbound line. Calls being sent over PRI with Digium card. Sounds like the PSTN just doesn't forward the name for whatever reason. Thanks, Bill Correct me if I am wrong but I think you would need access to a SS7 switch to send the number that you want it to come from. -- Ciao, Thomas \__o___\_o_-. |\--/ | / Thomas Gallaway\ http://atom.port11.net | \/--\_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending Caller ID info in MD/USA
William C. Lohr Jr. wrote: Marty, My business would own the actual T1, but I may provide an outbound call service for a client and would want their name sent as well as their toll free number and not the local number for the outbound line. Calls being sent over PRI with Digium card. Sounds like the PSTN just doesn't forward the name for whatever reason. Thanks, Bill Sorry to add another thing. The reason why they do not want everybody to be able to send this information is obvious. You don't want your average joe to be able to spoof their caller id or name to something like Bank Of America and social engineer information out of people. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FC2 compile of zaptel
Jerry Geis wrote: I just grabbed my fresh Fedora Core 2 final release. Untared zaptel-0.9.1 dir make linux26 and I get errors on the compile. Anyone else tried this yet and been sucessful? I did have to ln -s /usr/src/linux-2.6.5-1.358 /usr/src/linux-2.6 But I still get errors after that... about asm/linkage.h asm/types.h See post above by Joshua M Thompson. Same issue? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Sip (FWD, SipGate and such) problem
Karl Brose wrote: I think when you have this setup you need to keep the media path going through Asterisk at all times. Your SIP is binding to both ports, internal and external, but that doesn't correctly set it up for either scenario, localnet calls and external calls. It won't keep the addresses straight for the RTP channels. Try setting CANREINVITE=NO for peers (FWD,..) and for your local SIP phones. When a channel is created in asterisk the media path is going through Asterisk, but during a call the endpoints can issue reinvites which switches the media path directly between the endpoints. You need to prevent that. Other solutions are to run IAX to/from FWD and SIP locally, or SIP to the external peers and IAX to a local IAX phone (or another protocol). Or you should be able to create your own NAT using the iptables and bind asterisk only on one port either outside or inside and set the right corresponding parameters. The RTP will still bind on all ports currently, but that will be fixed in a matter of days. Also, sipgate.net should be sipgate.de (works ok though since they don't care) fromdomain is meant to be realm not a hostname. Thomas Gallaway wrote: Hi all I use sipgate and FWD but seem not to get it going. I do not have NAT on the asterisk box (static ip). The asterisk box has 2 network interfaces. One internal and one external. Now when I make an call to a FWD or SipGate number all I get is -- Executing NoOp(SIP/113-6d2e, ) in new stack -- Executing Goto(SIP/113-6d2e, intern-post|714551|1) in new stack -- Goto (intern-post,714551,1) -- Executing SetCallerID(SIP/113-6d2e, 270002) in new stack -- Executing SetCIDName(SIP/113-6d2e, Thomas Gallaway) in new stack -- Executing Dial(SIP/113-6d2e, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd270002-6ee7 answered SIP/113-6d2e -- Attempting native bridge of SIP/113-6d2e and SIP/fwd270002-6ee7 == Spawn extension (intern-post, 714551, 3) exited non-zero on 'SIP/113-6d2e' But either I get 1/2 second of audio or no audio. No matter how long I wait there is just no audio or just a short snippet of audio at the beginning. Here is parts of my sip.conf; [general] port = 5060 ; Port to bind to localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask externip = 206.40.161.235 context = intern; Default for incoming calls maxexpirey=3600 defaultexpirey=300 disallow=all; Disallow all codecsa allow=gsm allow=alaw allow=ulaw tos=reliability register = xxx:[EMAIL PROTECTED]/150 register = xxx:[EMAIL PROTECTED]/151 [sipgate1] type=friend username=xxx secret=xxx host=sipgate.de fromuser=xxx fromdomain=sipgate.net nat=no context=incoming-sipgate canreinvite=yes [fwd270002] allow=ulaw type=friend context=incoming-fwd secret=xxx username=xxx host=fwd.pulver.com Any ideas? When I put nat=yes I actually will get 1 second of audio, then it dies. I have been googling for a while now and not seem to find any sollution to this. -- Thomas I will try that. I had to remove all the IAX / SIP changes I did as even on the local network it started to give me a one way communication thing. I was able to hear other people but they could not hear me. Will give this another try later in the week. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 2 and Kernel 2.6
WipeOut wrote: Joshua M. Thompson wrote: On Thu, 2004-05-20 at 05:12, WipeOut wrote: When trying to build zaptel it required me to link /usr/scr/linux-2.6 to the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :) After than I tried again but the page rolls with errors and finally ends with.. make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Anyone got ant ideas? You'll need to configure the source tree before zaptel will compile. The config files are in /usr/src/linux-2.6/configs...copy the one that matches what you're running to /usr/src/linux-2.6/.config and then run make oldconfig. Zaptel should compile after that. Thanks for the try but its didn't work.. Got exactly the same result.. Anything else I can try? It really sounds like your do not have the kernel-headers installed. I never tried a 2.6 kernel but on 2.4 I got similar errors until I installed the kernel-headers. How did you get the kernel header files for FC2? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream v1.0.4.68 firmware
Mikael Andersson wrote: Thomas Gallaway -- wrote on den 17 maj 2004 16:57: Hmmm well I need to kinda figure out how to get the custom ringtones to ring on the phone... :-) ___ Asterisk-Users or how to change them /M Yeah I can change them in the firmware, but I wonder if there is an option in asterisk to pass to have it do a certain ring if the call is internal, or external. The format the rings are at are after what I found out uLaw compressed 8bit 8000hz mono samples. But they also have a header infront of the file. I will play arround with it later. Maybe there is a way to chop off the header of the ones that come with it and put it infront of a regular file. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? I just slapped them all onto my TFTP server and they all load fine. Then on the bottom I can choose between the 3 ringtones and tell it to ring a certain ringtone if coming from a certain caller id. I just do not like the ringtones (piano). It'd be great if there was a way to upload own ringtones but I can not seem to be able to find out how to edit the files. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Jeremy McNamara wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Didn't you hear you've gota purchase their $100,000,000 provisioning tool to enable ringtones. My ringtones just work on all the grandstream's :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Brian Capouch wrote: Thomas Gallaway wrote: My ringtones just work on all the grandstream's :-) Do the URLS for the ringtones at the top show up as something other than all zeroes? I've fiddled with this until blue in the face, and the ring sounds just like the ring it had before. This is with 1.0.4.68, but it was no different with the earlier supposedly ringtone enabled version. *Product Model: * BT100 *Software Version: * Program--1.0.4.68Bootloader--1.0.0.16 HTML--1.0.0.31VOC--1.0.0.5 *Custom Ring Tone: * ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Stephen R. Besch wrote: Thomas Galloway wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? I just slapped them all onto my TFTP server and they all load fine. Then on the bottom I can choose between the 3 ringtones and tell it to ring a certain ringtone if coming from a certain caller id. I just do not like the ringtones (piano). It'd be great if there was a way to upload own ringtones but I can not seem to be able to find out how to edit the files. That has not been my experience. I do indeed get the option to associate each ring tone with a given caller ID (which in my estimation is a really stupid implementation anyway - the real value would be in associating each line on the 2 line GS with a different ring tone. The caller ID already tells you who is calling). However, I can put anything I want into the text boxes and nothing happens - I always get the system ring tone. And, what are those stupid little radio boxes for. No matter which one I check, when the screen refreshes it defaults back to the System Ring Tone. Here's what they look like (the o's are supposed to be radio boxes): o System Ring Tone o Custom Ring tone 1, used if incoming caller ID is (Text Box) o Custom Ring tone 2, etc. For me when I select Custom Ring tone 1 I get the first ring tone (what is some stupid piano playing and is pretty much useless) As for ring tone 2 is some piano too. (even more useless now) and ring tone 3 guess piano. (That killed the sense of that function for me) My boss was asking to have have a different ringtone so he can figure out if it's his phone ringing or the one in the office next door. Well I guess with piano's they are the same again. Doooh What's this supposed to mean? It implies that if I select one of the custom ring tones, then the phone will ring on the matching CID, otherwise, it won't ring at all! This feature really needs work. I hope it doesn't wind up like the useless Daylight Savings Time option, which you may have noticed does not pay any attention to the date so you have to log into each and every phone and change the option anyway (please, correct me if this has been fixed). Why bother? I can just as easily change the time zone and get the same effect. GS is obviously targeting their phones for the consumer market where a customer has 1, maybe 2 phones and this kind of thing is irrelevant. The whole concept of their overpriced provisioning system is rather a funny joke in this context, and a rather pretentious one at that. Might be able to hack up some script that changes the function in the tftp file that get's uploaded when the phone is turned on. What would mean everytime there is a timezone change have to run the script and reboot the phones. Ah well we just have 6 of those phones here and so far they are kinda okay (besides mine crashing all the time especially after calls). -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Stephen R. Besch wrote: Thomas Gallaway wrote: Brian Capouch wrote: Thomas Gallaway wrote: My ringtones just work on all the grandstream's :-) Do the URLS for the ringtones at the top show up as something other than all zeroes? I've fiddled with this until blue in the face, and the ring sounds just like the ring it had before. This is with 1.0.4.68, but it was no different with the earlier supposedly ringtone enabled version. *Product Model: * BT100 *Software Version: * Program--1.0.4.68 Bootloader--1.0.0.16 HTML--1.0.0.31VOC--1.0.0.5 *Custom Ring Tone: * ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) -- Thomas Well in my case, the ring versions are all 0.0.0.0 no matter what I do. Could you also post the exact spelling of the binarys on the tftp, including capitalization, access rights and access mode. Mine are ring1.bin, ring2.bin, ring3.bin, all lower case, owned by root, in the asterisk group and with r/w mode=rw_rw_r__ (664), which is the same as all the other items on the server, which do in fact get loaded. Also, I've checked the binaries and they do in fact have version 1.0.0.0 embedded in the file (look at hex offset 6, which is where the version signature of all the GS bianries is located). I also can connect to the tftp server from another machine and successfully get ring1.bin. Perhaps the binaries that came with my copy of the firmware are corrupted. Maybe you could zip up the tones you are using and post them to me so I could see if these fix the problem. I right now run solarwinds tftp server on a winblooze 2000 server. Maybe that's the problem. I had some issues with tftpd on linux. Well actually I just had not the time to mess arround with them hehe. http://atom.port11.net/data/110468.zip (this is the archive I am using) -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream v1.0.4.68 firmware
Duane wrote: Grandstream v1.0.4.68 firmware http://www.hellofone.com/downloads.html Seems to have loaded ok on my BT100.. Hmmm well I need to kinda figure out how to get the custom ringtones to ring on the phone... :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] total newbie sanity check
Mike Stupak wrote: Im a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w Im planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate ([EMAIL PROTECTED] mailto:[EMAIL PROTECTED]). Heres what Im planning: === Parts List === 1 Digium Wildcard TDM400P w/ 4-port FXO bundle Im planning on using this to connect to a few CO POTS lines. That should be fine if you have 4 incoming pots lines A mid-range computer (600MHz or so w/ 512 MB RAM) Should do fine too! We run a similar setup with a P3-1Ghz and it's most of the time in idle. Some form of Linux (fedora?) I run fedora FC1 too. Runs quite well. Just remember to install the kernel and kernel headers. This is required for asterisk and all zaptel stuff to compile right. The tool you want to use is yum. yum install kernel-header and so on. There is an yum.conf file. Yum is also good if you want to up date your system. Asterisk Yup. Install asterisk from CVS and NOT from rpm. The rpm version that I installed back then was causing me nothing but trouble. 10/100 Ethernet card (in the computer) 10/100 Ethernet Switch (8 port or so) A few SIP capable phones === End Parts List === And now a few questions: 1) Is this a feasible system? Am I missing any important hardware? Yes 2) What is a good Linux to use? Im reasonably proficient w/ Linux. I think most people will suggest you to use debian. Fedora works quite well here at my office. 3) Do I need to tell the phone company anything special or do I just have them connect up standard phone lines? 4) Can the phone company usually roll the calls onto a spare incoming CO line? (e.g. if the first line is busy - route it to the second line, if the 2^nd is busy route to the 3^rd , etc.) Is there a special name for feature this that the phone company will recognize? We do it that way. We have 4 incoming pots lines with 4 numbers. When number 1 is busy it will forward to the next line. If that one is busy it will forward to the next. The only thing I have run into as a problem sometimes people dial back the number they got called from. If that is the last number we have and it is busy even if line 1/2/3 are free line 4 will not forward back to 1. For some reason our phone provider will not do that. 5) Id like to support a special feature Id like to have 2 different incoming phone numbers (o n all lines) and have asterisk multiplex to the right voice menu system based on the incoming phone number. Is this possible? Does it require special features from the phone company? I guess that is not that easy. If you can just group 2 of the 4 pots lines together I guess you can create a dialplan like that. But then you could only have 2 incoming lines per number but 4 outgoing lines. Thats all for now (though Im sure this wont be my last post. Thanks again for any help. - Mike Stupak -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Scalable IVR
Digvijay Singh wrote: Hi, I am presently thinking of making just a demo application because i need to assemble it real quickhooking up an analog phone line for an input channel...with a dial out feature as well.. I think I would need an XP100 wildcard as FXO .not sure of what to use for FXS... Incidentally steven informed me that anything with a PCI bus even may be a P133 could do it...I am presently in india and trying hard to find a vendor... getting the stuff shipped might take upto 2 weeks ...with usual custom delays .. Suggestions are heartily appreciated .. thanks digvijay - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 14, 2004 11:28 PM Subject: RE: [Asterisk-Users] Scalable IVR Hi Digvijay- I have done something similar to what you're looking to do, so maybe I can help you. I currently have a system that takes up to 600 simultaneous IVR calls, supplied by a large private DMS-100 PBX over 20 E1 spans. I think my load is a little higher than yours however, because the calls typically are very short (5 seconds). Here's a summary of what I've found - please contact me directly for more detail: I suggest: * Use 1 processor (example: 2.8Ghz P4) for every 4 E1's (example: one TE405P card) when you have that much call setup traffic. * Try to do as much as possible using the dialplan (extensions.conf). AGI scripts are very powerful but cost you in performance when you're running large numbers of lines. * For the processor to use, see the Wiki for suggestions. I've used P4 and Xeon based Tyan and Intel motherboards with success. * Where you get the hardware depends of course on where you are located. Good luck! Scott Stingel 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 digvijay singh Sent: Friday, May 14, 2004 10:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Scalable IVR Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future may have huge loads .. from 1000 to 1 simultaneous peak time callers with 1 minute duration calls ( just to mention how scalable we would ideally desire it to be ) At present we would need a maximum of 10 simultaneous users peak load capability. From what i know so far asterisk is the most cost effective and sound option. Now to my question.. 1-) What would be the hardware requirements in these different cases a-) for a single analog phone line for demo purposes intially to b-) a peak time 10 simultaneous calls facility and later c-) going upto 10,000 simultaneous calls scalability 2-) Where can i procure the hardware from I was thinking of taking 30 access channels of 64 Kbps and 1 signalling channel of 64 Kbps (30B + D). for an ISDN compatible EPABX Kindly let me know of your opinion Thanks digvijay Now if you need to get an demo application going and need it going quick why dont you just install asterisk on some box (maybe a P2-300+ suggestable cause compiling will take ages on a P1-133) and use all software until you get the hardware. Use a software phone (see voip-info.org) so you can get your demo coded. Then once you got the rest of the hardware you can hang it onto the PSTN. This will get you going for now. I guess you could just use some sort of SIP provider to give you an dial in phone line too. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kernel Freezes with T100P
Zach Chambers wrote: Hey Folks, I have an extremely reproducable problem with the T100P card. I'm running one card in an Athlon XP 1800 machine running Fedora Core 1. The kernel is a stock 2.4.26 kernel pulled from kernel.org. I'm going to be using the T1 interface for data only. If I bring the interface up, then down, then up again, the machine will freeze within 30 seconds or so. There are are no debugging messages even if debugging is turned on. Occasionally it takes another up/down cycle to cause the machine to freeze, but that is rare. I've seen other incidents of machine's freezing in the archives, but no real answers to the problem. Any idea where to start with this? Thanks, -Zach. Ever tried to use the latest fedora kernel. Also what type of chipset do you have? Might be an chipset incompatiblity. Did you ever try swapping arround the pci card into a nother PCI slot or playing arround with the IRQ's? -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kernel Freezes with T100P
Zach Chambers wrote: Ever tried to use the latest fedora kernel. Also what type of chipset do you have? Might be an chipset incompatiblity. Did you ever try swapping arround the pci card into a nother PCI slot or playing arround with the IRQ's? -- Thomas Thomas, I could not get the zaptel drivers to compile using any of the fedora kernels which is what drove me to the stock kernel to begin with. I'll check the archives again on that issue. The chipset is a VIA chipset. I don't know much about it other than that yet. I'll try PCI slot move as well. Thanks, -Zach. I actually just installed the zaptel driver for fedora from cvs. Christian helped me getting the latest CVS version of asterisk, libpri and zaptel. Then what I did is grabbed the kernel-source for fedora. I guess in your case you want to do an kernel update. yum update yum install kernel-source It should install the 2188 kernel and 2188 kernel source. Then just go to /usr/src/zaptel make clean make make install and same with libpri and asterisk. Just make sure you removed the wcfxo kernel mod be4 you install zaptel. And then reboot the box as when I modprobe wcfxp it gave me an actuall kernel panic :-) After reboot everything worked just fine. It even seems like my dropped call issue is gone. If you need a copy of the yum.conf file let me know. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropped calles (with mp3)
Hi I am still struggling with those dropped calls and am about to give up. I have created an MP3 of the exact scenario that I can reproduce when the call gets dropped. Maybe this might have an clue for somebody. http://atom.port11.net/media/disconnectasterisk.mp3 Also here is my zaptel.conf: [channels] group=1 language=en context=default signalling=fxs_ks busydetect=yes echocancel=yes echotraining=yes rxgain=0.0 txgain=0.0 callprogress=no echocancelwhenbridged=yes relaxdtmf=yes immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel=1-4 Thanks for your help. If there are any changes I should do just let me know. I can just call t-mobile customer support and see if I get disconnected. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I love you!
[EMAIL PROTECTED] wrote: lovely, :-) Is it just me or where there allready 3 virus sent to this list today? Maybe time for denim to disallow attachments? :-) -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P keeping PSTN line Offhook
Shahid wrote: Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 I have the exact same thing happening. I was able to track it down to the Music On Hold. Scenario: - My boss calls my cellphone - I do not pick up - My cellphone's voicemail picks up - He hangs up - Asterisk plays On Hold music to voicemail until voicemail on cellphone times out - Zap channel is stuck -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random disconnect of calls
Matt Riddell wrote: Are you using the more than one manager application? I.E. op_panel etc... Reason I ask, is that I had two copies of op_panel running and lost a couple of calls, but with just one running things have ben fine... Matt Riddell - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, April 13, 2004 4:54 AM Subject: Re: [Asterisk-Users] Random disconnect of calls | Chris A. Icide wrote: | | This was happening to me as well. | | What finally fixed it was disabling echo cancelling on the X100P cards | in the zapata.conf files. | | However this resulted in a horrid echo on my cisco phone when I was | using a line attached via one of my x100P cards. | | So I went back and re-enabled echo cancelling and set echotraining=yes. | | I've had much better luck with, however I still get a dropped call now | and then. | | -Chris | | On 08:35 AM 4/12/2004, [EMAIL PROTECTED] wrote: | Hi | | I am experiencing some weird behaviour. Calls get disconnected random. | There is no error in the log files. | | Sometimes I can talk over 30minutes+ and it is fine. Just earlier I was | only able to talk 2 minutes per session and get disconnected. All I hear | when this happens is a fast busy. | My set up is this: 8 * Grandstream Budge Tone 101. 4 * X100P cards. | Compaq 1Ghz ML Server. | I am running Asterisk 0.7.2 installed from RPM's on Fedora. | | The CPU load of the machine is fine | | What I noticed and I do not know if this is related to the problem but | the messages file of asterisk has the following entrys: | Apr 12 11:11:50 WARNING[-1210991696]: Maximum retries exceeded on call | [EMAIL PROTECTED] for seqno 102 (Request) | Apr 12 11:14:20 WARNING[-1210991696]: Maximum retries exceeded on call | [EMAIL PROTECTED] for seqno 102 (Request) | | I am not using NAT asterisk is on an internal ip. | | Thanks for you help. | | | Yes turning off echo cancelling would be fatal. We have some serious | echo going on here that I can not seem to track done. I am assuming it | is just this old building. Maybe we can go ISDN or so but the dropped | calls are rather bad. I am just running 1 instance of the op_panel. But today I noticed that 2 calls got ended after 2 minutes 34 seconds. I will disable the management thing just for testes. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random disconnect of calls
Eric Wieling wrote: [EMAIL PROTECTED] wrote: Yes turning off echo cancelling would be fatal. We have some serious echo going on here that I can not seem to track done. I am assuming it is just this old building. Maybe we can go ISDN or so but the dropped calls are rather bad. Rnadom disconnect with X100P is usually caused by busydetect= or callprogress= set to yes Both are set to no. Checked that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random disconnect of calls
Matt Riddell wrote: - Original Message - From: Thomas Gallaway -big snip - | I am just running 1 instance of the op_panel. But today I noticed that 2 | calls got ended after | 2 minutes 34 seconds. I will disable the management thing just for testes. | ___ I can't imagine the management interface having much influence over your testes. Matt Riddell Yeah people should not type with fever. Somebody better take away my keyboard. Damn rainy weather. Of course I did mean disable the management module just for testing. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P card issues - noise, volume, etc
San Singhania wrote: Hello, I have just managed to get my 1st * server up and running and have a lot of issues with theX100P analog card. Would really appreciate anyone trying to help me on the following : 1. The receive and transmit is too soft. So i increased the txgain and rxgain. The volume is fine after this, but there is a lot of 'wind' noise on the line. I have my echo cancellation on, aggregive suppression on but still no use. However, the moment I set the txgain and rxgain back to 0, everything works fine but both parties compain that they cannot hear each other. Any solution for this? Also, if I use an ISDN interface, will it solve my problems? I understand that since it is digital, there are no longer issues with hangup detection, noise, echo, etc. Is that true? Also, has anyone had any experience with conferening (i.e bridging) 2 FXO cards together? Thanks San I think the problem is when you turn on rx/txgain the echo cancel will stop working? Check your /var/log/messages file that is how I found out about it. Mar 26 12:26:34 voip kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO PCI Card
Paul Tyreman wrote: Does anyone know if you can put two of the X100P cards in to the same machine and have access to two landlines ? I just need to know if it's worth buying two or not ! Thanks, Paul. I run 4 X100P's in our asterisk box. Just make sure you give each card it's own IRQ. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who has access numbers in the UK and Germany?
Stephen Karrington wrote: I can't read German. Can you outline the cost for me? Thanks. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ===8==Original message text=== sipgate.de has DIDs in Germany and the UK. -Alfred -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Karrington Sent: Friday, April 09, 2004 4:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Who has access numbers in the UK and Germany? Hello, I need a few access numbers in the UK and Germany. Does anyone have this available right now? I need the incoming calls to be directed through IP to one of my asterisk servers in Europe. Please contact me off the list if you want. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us What information are you looking for? They fee's are 1.79 Cent/Minute (euro that is) plus taxes. No monthly fee, no minimum amount of calls required. Their second option is for 8.90Euro you can get 1000 minutes/month. You will receive a free phone number. SIP to SIP is free. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restart Asterisk
Jain, Sonal wrote: Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. Thanks, Entering reload in the console should do if you edit the extensions.conf and some other files. There are some files if you edit them you need to shut down and restart asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't hear asterisk sound files on snom200
jc wrote: I have a simple * setup with a couple of SNOM200 installed. I can make IP calls and internal calls fine. But, I cant hear any of the asterisk sound files on playback. Any ideas are there any errors on the CLI? I had that problem too but it did actually throw out errors on the CLI. Had to restart asterisk to get it working. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 10 day old email, virus already received
[EMAIL PROTECTED] wrote: Hi, I´m a member from this list and I do not have a virus because I check my email from my server linux. cheers. vozip Mensaje citado por randulo [EMAIL PROTECTED]: For info, I receive the mailing list on a brand new account that is not used for anything else. Just received, a virus (*apparently*) From: [EMAIL PROTECTED] I suppose there may be 8,000 people getting it but just in case. ___ As everybody should know by now all those new viruses/worms forge from headers. So the virus might seem to come from a certain email adress but it's not. The only way to figure out where it came from is by checking the header file. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High latency from Europe, 500-800ms.
[EMAIL PROTECTED] wrote: We're using a 7940 from Europe, connecting to a US Asterisk server, and it works great. We setup a local Asterisk server in Europe, had the 7940 connect to it, and used IAX2/GSM to connect to the US. It is choppy using all CODECS, and I am curious if there are any recommendations on getting this to work well? I'd rather not have the phones connect directly to the US. Thanks. Bill Sounds like you are having network issues. What kind of connectivity is on both ends? Do a traceroute from site A to site B and see where the latency is coming from. I usually have 128ms pings to europe (germany). -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High latency from Europe, 500-800ms.
Angel Gabriel wrote: I'm no expert on * , I don;t even think i class as a newbie yet, but my understanding of a sat. link to the net is that the link is one way, as in downstream, and you still need an upstream conenction to your service provider. If this is the case, then remember, your internet service is only as fast as the slowest link in the chain. Not all sat connections are one way. But the issue with sat connections is *drumroll* latency! As the signal is beeing relayed over the sattelite this will cause latency. Also if the sat service is not providing enough downstream it's bad too. I would definately look into getting your network straighend out first. There are many factors. Is your connection shared? What speeds? Let say it like that if you have people on your local lan using bandwith or running peer 2 peer filesharing stuff this will take away your upstream speed. Do some tests. Go to http://www.dslreports.com and do some speedtests to see how much speed you really got towards the USA. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registration from xxx failed for 'xxx'
Hi Here is my problem. I have 2 phones (Grandstream Budge Tone-100) loosing the sip registration every 4 hours. I can not find out why. It seems like the registration fails, then a few minutes after registers sucessfull. Mar 19 14:06:14 NOTICE[147466]: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.1.114' -- Executing NoOp(SIP/114-5d35, ) in new stack -- Registered SIP '114' at 192.168.1.114 port 5060 expires 180 sip show peers show's me this: 116/116 192.168.1.116 (D) 255.255.255.255 5060 Unmonitored 114/114 (Unspecified) (D) 255.255.255.255 0Unmonitored 113/113 192.168.1.113 (D) 255.255.255.255 5060 Unmonitored Thanks for any input on this. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration from xxx failed for 'xxx'
Eric Wieling wrote: If the phones are behind NAT you can try lowering the registration interval to something LESS than 4 hours, like 60 seconds or 120 seconds. Usually this problem is because the NAT firewall sees no activity on the registration port and so deletes the NAT translation for that port. On Fri, 2004-03-19 at 16:26, Thomas Gallaway wrote: Hi Here is my problem. I have 2 phones (Grandstream Budge Tone-100) loosing the sip registration every 4 hours. I can not find out why. It seems like the registration fails, then a few minutes after registers sucessfull. Mar 19 14:06:14 NOTICE[147466]: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.1.114' -- Executing NoOp(SIP/114-5d35, ) in new stack -- Registered SIP '114' at 192.168.1.114 port 5060 expires 180 sip show peers show's me this: 116/116 192.168.1.116 (D) 255.255.255.255 5060 Unmonitored 114/114 (Unspecified) (D) 255.255.255.255 0Unmonitored 113/113 192.168.1.113 (D) 255.255.255.255 5060 Unmonitored Thanks for any input on this. -- Thomas Not using nat. That is the weird thing. They are all directly connected on the same switch as asterisk. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP Read error: Resource temporarily unavailable (DTMF Issues)
Hi I am working on this since a while now and seem to be stuck. Here is my issue: I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN lines. It all works fine just the DTMF is not working. I am not beind a NAT so the phones can talk directly to the asterisk server. When I want to check voicemail for example the voicemail box is 113 and password is : -- Playing 'vm-password' (language 'en') Mar 16 10:24:16 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:24:16 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:24:17 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:24:18 WARNING[770065]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Incorrect password '1' for user '1113' (context = any) -- Playing 'vm-incorrect' (language 'en') -- Playing 'vm-password' (language 'en') Mar 16 10:20:42 WARNING[753681]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:20:43 WARNING[753681]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:20:44 WARNING[753681]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Incorrect password '11' for user '1133' (context = any) -- Playing 'vm-incorrect' (language 'en') -- -- Playing 'vm-login' (language 'en') Mar 16 10:32:00 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:32:00 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Mar 16 10:32:00 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Playing 'vm-password' (language 'en') Mar 16 10:32:04 WARNING[786449]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable -- Incorrect password '' for user '1133' (context = any) -- Playing 'vm-incorrect' (language 'en') and so on. Maybe 1 out of 20 tries I can get the password right. The same happens with Meetme. If I tell meetme to use a PIN it wont detect it right. My sip.conf looks like this for one phone: [113] username=113 type=friend host=dynamic disallow=all allow=alaw allow=ulaw context=intern secret= mailbox=113 dtmfmode=rfc2833 nat=0 My budgetone is set to send DTMF via via RTP (RFC2833) with a payload type of 101. (tried 100 and 102) The first 2 codecs are set to PCMU and PCMA (tried to switch those arround too). Any help very appreciated! Thanks, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: RTP Read error: Resource temporarily unavailable (DTMF Issues)
Tony Mountifield wrote: In article [EMAIL PROTECTED], Thomas Gallaway [EMAIL PROTECTED] wrote: Hi I am working on this since a while now and seem to be stuck. Here is my issue: I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN lines. It all works fine just the DTMF is not working. I am not beind a NAT so the phones can talk directly to the asterisk server. [...] My sip.conf looks like this for one phone: [113] username=113 type=friend host=dynamic disallow=all allow=alaw allow=ulaw context=intern secret= mailbox=113 dtmfmode=rfc2833 nat=0 My budgetone is set to send DTMF via via RTP (RFC2833) with a payload type of 101. (tried 100 and 102) The first 2 codecs are set to PCMU and PCMA (tried to switch those arround too). Put dtmfmode=info in your sip.conf, and set the phone to use SIP INFO. Then it will work. Thanks for the fast response! This solved the problem! Many thanks! -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
