Re: [Asterisk-Users] My Boss wants background music!!!!
On Fri, 17 Dec 2004, Wilson Pickett wrote: I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do What a waste of resources though, like installing video games on the asterisk server... Ther must be a powerline intercom that would handle this (adding a speaker per music distribution point.) The requirement of the original poster was to mute the music at the desk when a call is in progress. It would be really nice if there was a hardphone capable of accepting a multicast high-quality stream when no call was in progress. Peter ___ 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] My Boss wants background music!!!!
Xml services for cisco 7960, setup a broadcast stream. Check the wiki. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Friday, December 17, 2004 3:28 AM To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] My Boss wants background music On Fri, 17 Dec 2004, Wilson Pickett wrote: I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do What a waste of resources though, like installing video games on the asterisk server... Ther must be a powerline intercom that would handle this (adding a speaker per music distribution point.) The requirement of the original poster was to mute the music at the desk when a call is in progress. It would be really nice if there was a hardphone capable of accepting a multicast high-quality stream when no call was in progress. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Video Attachement format
Hello I have tested the Video-Voicemail feature with our SIP Hard phone, it works great ! I'm trying to convert the h263 file (who cannot be played with an out of stock Windows Media Player) to another format for email forwarding (mpeg or another WMP recognised format) Anyone has tried to ? (I have tried transcode or ffmpeg without success) thanks for advice. Nicolas FOURNIL Nicolas P2P manager http://www.videotel.fr PS: We actualy doing a french translation for voicemail prompts, with also local changes (date format etc...) we will release it asap. ___ 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] How expensiveare thedifferent codecs?(Regarding CPU time)
Hi! Jim Van Meggelen schrieb: [EMAIL PROTECTED] wrote: Jim Van Meggelen schrieb: You understand. I was kidding a little bit, but yes, I am also wondering just what things can be done to get a slower machine to work as well as possible. Okay. So lets try. What are you running in terms of a kernel or distro? I'm running a Debian Woody with a handmade 2.6.5 (based on a woody backport) Otherwise - without the -p option - the system had values of 400ms (and higher) converting speex when it wasn't idle. Now the value is constantly at about 210. Nice. The system is now giving Asterisk the priority it needs. Don't forget to change that in your rc.local, or wherever you're starting Asterisk from. I'm starting it from a start-stop daemon. AT the moment I'm having a little fight with it. When starting from the script it tells me Starting Asterisk PBX: Unable to set high priority. Starting it from the shell works. But I guess I can convince the script to cooperate ;-) I guess this option could help me a lot regarding the sound problems I got sometimes. Yes, it might help a lot. I will see when doing some calls over sip (there I had the most problems). Maybe at the evening. Now I have to breakfast, shower and go to work. And I need to go to bed! Good night! Bye! Michael ___ 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] Calculating required bandwidth
Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit, then it has only 23 channels to carry voice, as the 24th channel is used for the D-channel (signalling channel). Only if you're in the US. We have 30 + 1 :-) E1 == 2048kbps == 32 channels, giving 30 B + 1 D + 1 for timing roy ___ 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] Queueueueuueue position
There are 3 to 5 calls in the queue at that moment, all from different CID, hold time is over a minute. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Brian Roy Verzonden: vrijdag 17 december 2004 1:52 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Queueueueuueue position On Thu, 16 Dec 2004 15:18:10 +0100, E. Versaevel [EMAIL PROTECTED] wrote: When I call in (with an agent logged in) I get to hear the MOH on the client side, hover no matter how high the hold time is, I NEVER get an announcement over my queue position or my estimated wait time? Both the incoming call and the agent are on SIP channels. What is wrong ? Kind regards, E. Versaevel Would that be because this is the only call in queue? Try putting another call in queue and see what you get. -Chuji ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calculating required bandwidth
Hi, A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit, then it has only 23 channels to carry voice, as the 24th channel is used for the D-channel (signalling channel). Only if you're in the US. We have 30 + 1 :-) Are you sure? As far as I know, E1 is 30 + 2, not 1... Best Regards, -- Durval Menezes (durval AT tmp DOT com DOT br, http://www.tmp.com.br/) ___ 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] VoIP Termination
snip Fortunately, asterisk will do that for you to the second, looking at the cdr records and totalling up the duration column for a specific period will tell you what your bill would be at the few cents a minute you'll be charged. Actually not true, although a common mistake. You do not pay to the provider by the second, so never, ever, ever total the duration column. Usually you don't pay by the minute either, but by some intermediate, like in units of 12 seconds, but you need to check it with the provider. This is a serious mistake, and it gets worse the more calls you total. Let's say, that you pay by the minute. Having made 6 calls of under ten seconds costs you as if you've made 6 calls of 59 seconds. If you just total the duration of the calls, you'd think you're paying for one minute and you're actually paying for six. That's a big mistake. Now imagine you had 12000 calls to total. Always total the price for each call, never the duration. ___ 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] Paris Meeting Date/Time/Location
The Paris Asterisk meeting will be held Monday, December 20, 2004 at 1 p.m. at Les Vendanges--a wonderful restaurant in the 14th (tel 01.45.39.59.98). However, we have to let them know exactly how many people will attend, so PLEASE RSVP as soon as possible. The address is 40, rue Friant, and the metro station is Porte d'Orleans, the end of line 4. Take Sortie Boulevard Brune, numeros impairs (odd-numbers), and go straight on Boulevard Brune. Rue Friant is the first street that intersects Boulevard Brune on the right. Turn right on rue Friant, the restaurant will be on the left at the corner of rue Friant and rue Morere. The price is fixed regardless of the items you order: 25 euros for 2 courses (entree and main course, or main course and dessert) or 35 euros for 3 courses. There is a good wine selection, many are reasonably priced. Mark ___ 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] MD110 and analog trunks
Hello all, I was wondering if someone already wrote something to support a serial connection(ICU) on PABX's that's used for signaling. What I currently have is a connection between an Ericsson MD110 and * with analog trunks. Problem with this is, that all CallerID info is send over a serial link (ICU). Is there anyone who knows if there is support for this on * or to find the specification of ICU somewhere? Regards, Roelof Dijkstra Network Engineer EMEA Compuware Europe BV -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ 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] Has anyone connected to 7960 withconsolecablefor setup?
Randy, If this is the case, you might need this : http://www.voip-info.org/wiki-Firmware+issues+on+7940+-+7960 Might be worth a go if you suspect it to be the problem... Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay Sent: 16 December 2004 20:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Has anyone connected to 7960 withconsolecablefor setup? When I push the settings button, nothing happens. I never get a chance to put in the password. I think the previous owner may have messed up a firmware upgrade. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sean Cook Sent: Thursday, December 16, 2004 11:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup? Why can't you use the settings button? If you know the password (or using the default password) you should be able to unlock the phone and do a hard reset... Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] om] On Behalf Of Randy MacKay Sent: Thursday, December 16, 2004 1:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup? I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. Anyone have any ideas? Thanks, Randy I found a link to make a Cisco Console Cable for RJ-45. http://www.hardwarebook.net/cable/serial/cisc oconsole9.html DB9F RJ45 Receive Data 2 3 Transmit Data 3 6 Data Terminal Ready 4 7 Ground5 4 Ground5 5 Data Set Ready6 2 Request to Send 7 8 Clear to Send 8 1 The Console Access Manual, give the following cable information: Console Cable Requirements You use a serial cable with a connector to connect a PC and a phone. The cable uses an RJ-11 connector for the phone and an RJ-45 connector to an RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements for the console cable. Table D-1 Console Cable Pinouts RJ-11 Connector RJ-45 Connector Pin 2 == Pin 6 Pin 3 == Pin 4 Pin 4 == Pin 3 So, I thought I would go right from DB9F to RJ-11 DB9F RJ-45 RJ-11 Pin 2 Pin 3 Pin 4 Pin 5 Pin 4 Pin 3 Pin 3 Pin 6 Pin 2 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 _ __ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aste risk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste risk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SayUnixTime
If I use SayUnixTime to an extension, it will tell me the CORRECT time:
extension =698,1,SayUnixTime
If I use the same in the wakeup-agi, it tells me the time 14 hours in
the future:
$agi-exec('SayUnixTime', sprintf(%s||IMp, UnixDate($time, %s)));
Where is the difference? I am in Time zone Asia/Taipei (GMT+8:00)
Any ideas?
bye
Ronald
___
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] Meetme with video???
I wonder if there is an application available, what would allow me to have a conference call (meetme) with video. ___ 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] No Caller ID Name PRI NI2.
Yep, on one of my NI2 PRI's I had to add a wait(1) before I answer, otherwise CallerIDName did not show 94.52% of the time. Seems like it might be a buglet, but it didn't seem worth looking into since the fix was so simple. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 15, 2004 2:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Caller ID Name PRI NI2. For callerid on NI2 don't you have to put a wait on there before you answer? Otherwise you miss the packet with the name in it? I think brc_ had this same problem.. exten = s,1,Wait(2) exten = s,2,Answer bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Modesitt Sent: Wednesday, December 15, 2004 12:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Caller ID Name PRI NI2. Included is my debug. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, December 15, 2004 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No Caller ID Name PRI NI2. On Wed, Dec 15, 2004 at 10:52:07AM -0700, Chris Modesitt wrote: Okay, now I am really confused. I have two PRI's coming in from two different Carriers (QWEST and ELI), both of them are supposed to be setup to pass name and number on incoming calls. Problem that I am having is that I am not receiving inbound caller id name on either PRI, the only thing that both carriers have in common is that I am terminating into a DMS switch at the carrier. Observations: Caller ID Name dose not show up in the CDR records. PRI intense debug never sees the FACILITY IE message. Can you post the pri intense debug so that we can look at it? Matthew Fredrickson ___ 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] chan_sccp and 7920
I grabbed the 7920 tarball available from the sourceforge site dated sometime in Oct. I've got my 7920 working, however, asterisk seems to stop talking to it sometimes and requires a restart (like kill asterisk completely and restart it). Does anyone know if a new version is on the way? Does the current CVS fix this? Otherwise, the phone works great. Battery life sucks though. :( ~jay ___ 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] Hardware based DSP
Thank you all for your explanations related to my question. I have one follow-up question though. When I said that all dsp related stuff has to be handled by software within asterisk, I was thinking of conferencing at the time. I mean in order to be able to conference a sip session with a PSTN call, it would have to be handled by software, even if both the channels had hardware dsp capabilities. Right ?? If you are dealing with just a single channel, then the driver may handle codec/echo cancelation stuff with hardware help (??) As an aside, what is the best way to go about learnig about the aritecture of asterisk, other than using the source ?? The Wiki pages are great, but I have not (yet) found any info about asterisks architecture itself (in depth that is). Some linked websites / blogs provide good info on some topics, but is there a good high level design doc available anywhere ? It would stop people like me asking so many basic questions in the -dev list. Thanks Shahed ___ 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] instructions to get .bin firmware for 7920
Cisco no longer offers a zip file with the firmware in it, only an EXE file. If you use Winrar, you can unzip the exe. Pull out the following files and drop them somewhere: data1.hdr data1.cab data2.cab Download i6comp. If you search on google, it's all over the place. From the command line, run: i6comp e -r data1.cab This reassembles the CAB's and unpacks all of the files out of it. You'll need the cmterm-.bin and the OS7920.TXT file. Drop these on your tftp server. The phone won't even download the configs unless these 2 files exist. ~jay ___ 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] Low-latency kernel?
Rich Adamson wrote: While trying to apply the low-latency kernel patches to our RHv9 Linux 2.4.20-31.9, the patches would not apply. In comparing one of the first patch files (lowlatency.h) to that already on the system, it would appear the low latency patches were already applied by RH. The original RHv9 file (lowlatency.h) even had the patch author's name/credit in it. Does anyone know whether RH made an effort to incorporate the patches, and if so, about what kernel version? Redhat kernel's, like most Linux distro vendors, contain around 200 patches over the vanilla tree, including O(1) scheduler, VM system patches and the lowlatency patch. You can find out the exaqct version etc by downloading the kernel SRPM that include the list of patches. Gilad ___ 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] ADSI programming/TDM400P issues
My set up is quite simple: Redhat 9, Asterisk CV-HEAD from 07/11/04-13:04:22; TDM400P card, FXO card (X101P now unused) and a Fritz! AVM pci ISDN card. It works. But...just got an adsi PT390 phone and with some help from the nice people at Sayson got some basic programming into it. The programming came from dialling over a traditional phone line however and not from Asterisk. Whenever I dial into Asterisk to program it, I get the correct messages that its an adsi compliant phone, attempting upload and then I hear a short beep and the phone hangs up. When I dial into one of Sayson's servers from Asterisk, the same happens. Only going over a traditional phone line, do I get anywhere. Here's my theory: the TDM400P card is not dealing with the data signalling correctly. After a lot of googling and wiki work, I have checked the interrupts and see that: CPU0 0: 344331040 XT-PIC timer 1: 66 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC usb-ohci 5: 852647991 XT-PIC fcpci, wctdm 8: 1 XT-PIC rtc 10: 3457355287 XT-PIC eth0, wcfxo 12: 6 XT-PIC PS/2 Mouse 14:3031110 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 1 ERR: 0 Now, this looks like ethernet and the 101 card are sharing the same IRQ (probably not an issue as that card is currently unused) but that the fritz ISDN card (fcpci) and the TDM card are also sharing the same IRQ and that -could- (is?) a problem. As I say, the system works. I can dial out from the phone and receive calls into it fine. But I wonder whether the data exchange required by the ADSI programming is not working because of the IRQ sharing. Can anyone confirm this? Also, if there are other helpful hints about any other setup issue which might be causing ADSI not to program, please advise. I'm in the Netherlands, so my ISDN is from KPN. Ive tried the TDM card with loadzone=us and =nl and no difference. Also have tried changing indiciations.conf between nl and us to see if this has any affect. Still no change. In zaptel.conf I have kewl start selected figuring. Any suggestions welcome. cw -- Chris ___ 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] ADSI programming/TDM400P issues
Chris W wrote: Any suggestions welcome. Do you have the proper FDN/SEC codes for your phone located in asterisk.adsi, and have an extension created to program your phone? Matt ___ 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] Public Thanks
LoL :) Thanks, Humberto! But u posted my MSN, it does not work for mail. And U should have been posted it to asterisk-business, I gues. :)) Litnitsky Maxim Key Solutions, Moscow. http://www.asterisksupport.ru http://www.ksolutions.ru MSN [EMAIL PROTECTED] ICQ 172468035 ___ 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] AS5xx0: SS7 and SIP?
Title: AS5xx0: SS7 and SIP? We currently use Asterisk to provide a SIP-to-PSTN service. The actual conversion takes place somewhere in a softswitch owned by our SIP-to-PSTN provider, where we have an SS7 link. We would like to do that conversion ourselves. Is it possible to replace a softswitch with a Cisco AS5xx0 only (ie. AS5300, 5350, 5400), or is a *real* softswitch (ie. Cisco PGW2200) needed? Does anyone have any experience with an Asterisk--CiscoAS5xx0--SS7 configuration? As far as I know, Asterisk--CiscoAS5xx0--PRI works, but I couldn't find anything about SS7. Thank you -Manuel ___ 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] ADSI programming/TDM400P issues
On Fri, 17 Dec 2004, Matt Gibson wrote: Any suggestions welcome. Do you have the proper FDN/SEC codes for your phone located in asterisk.adsi, and have an extension created to program your phone? Not 100% sure. I've tried all the codes I could find - including those in the document I received from Sayson. The TE code on the back of the phone is TER01221 if that's any help also. However, if I call Sayson's programming system over Asterisk I also do not get into it (again, just the beep and a hangup) but if I dial it over a traditional phone line, then I a menu on the phone. I therefore suspect the locking codes could be an issue in the future but probably aren't the main issue right now. cw ___ 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[2]: [Asterisk-Users] Hardware based DSP
Dear Shahed, If you use hardware DSP for encoding and decoding you will need of less power Host CPU. There are no difference for what you will use these codecs - I mean conference or single conversation, because for conference all codecs are converted to G.711, then mixed and then back to the original codec for each party of conference group. So, if you would like to to conference with software coding you will need of very power computer. Best Regards, Miroslav Nachev S Thank you all for your explanations related to my question. S I have one follow-up question though. S When I said that all dsp related stuff has to be handled S by software within asterisk, I was thinking of S conferencing at the time. S I mean in order to be able to conference a sip session with S a PSTN call, it would have to be handled by software, even S if both the channels had hardware dsp capabilities. Right ?? S If you are dealing with just a single channel, then the driver S may handle codec/echo cancelation stuff with hardware help (??) S As an aside, what is the best way to go about learnig about S the aritecture of asterisk, other than using the source ?? S The Wiki pages are great, but I have not (yet) found any info S about asterisks architecture itself (in depth that is). S Some linked websites / blogs provide good info on some topics, S but is there a good high level design doc available anywhere ? S It would stop people like me asking so many basic questions in S the -dev list. S Thanks S Shahed S ___ S Asterisk-Users mailing list S [EMAIL PROTECTED] S http://lists.digium.com/mailman/listinfo/asterisk-users S To UNSUBSCRIBE or update options visit: Shttp://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Off Topic] humour, XMAS, ground loop - good business strategy
hi, I received this e-mail which contains a ballad, at first I thought it was junk mail, but then I read through it, for the EE members of this list, it may be quite humorous. I don't know if the ballad is original, but at least it's the XMAS season, so it's something to lighten up your day, eh? -samudra How the Ground Loop Stole Christmas Every dude down in dude-ville Liked Christmas a lot... But the Ground Loop, Who lived under Dude-ville, Did NOT! The Ground Loop hated Christmas! And Santa and Miss Claus Now please don't ask why, No one knows the cause. Maybe his connector wasn't screwed on just right. Could be, perhaps, that his crimps were too tight. But I think the most likely cause of it all Might be his cable was one conductor too small. But Whatever the reason, the crimps or the screws, He lurked there on Christmas Eve hating the dudes. They're shopping and bopping, they're trying and dipping, they're chirping and burping and buying and sipping They're hanging their stockings! He growled with a frown Tomorrow is Christmas! I must take them down! The Ground loop was angry, his face full of fury Had the dudes known, They'd have started to worry From deep in his mind, a plan did now hatch So evil and wicked, a way he could catch Dudes and dudettes in the midst of their shopping Action and motion all suddenly stopping His plan was so vile, his scheme was so clever The dudes would ponder and remember forever The dancing and singing and cash registers ringing Would grind to a halt and now become screaming The wicked intent of Ground Loop's foul mind Searched dudeville all over, hoping to find A careless young dude, oblivious to worry Who strung up his wires in too much of a hurry Probing and looking and peering and viewing The eyes of Old Ground Loop were roving and moving At last he did find on late Christmas Eve A witty and gritty technician named Steve So nifty and thrifty and clever was Steve Economy and elegance you wouldn't believe A master he was of superior design Quality, six sigma and kaizen combined Solid and robust and savvy his plan Earthquakes, hailstorms and rain to withstand A world class design, I give you no jive Except for a small problem with 485 Sarah in purchasing had called to contest A thousand dollar reduction was her request For Sarah, young Steve's heart did so yearn Shave pennies he would, her favor to earn In blueprints and drawings and plans he looked The requested amount of savings was booked Her hand at the dance he then requested With hope in her heart, his invitation accepted The music played and dancing ensued A beautiful evening with romance imbued And as she decided his company she liked Ol' Ground Loop with fury did finally strike His target of terror, the signals in town In bedlam and confusion the traffic would drown And just as Steve turned to take her back All the stoplights of dude-ville went Black Chaos and confusion did quickly arise The Ground Loop now claimed his deadly prize Only for want of optical isolation The town of dudeville suffered desolation Riots and shouting and crashes were heard Gridlock and damage and cursing incurred As Ground Loop witnessed this loss of control A tumult of joy filled his dark soul Shamefaced and panicked the couple sat glued A strenuous discussion quickly ensued You asked for cost savings Steven accused You skipped isolation Sarah diffused Upright they bolted as both realized This disaster would get them both downsized They became a team, no longer a faction They opened the door and flew into action Through gridlocked streets the couple dashed To save Dude-ville from Ground Loop's hash Back to the office they breathlessly ran So Christmas in Dude-ville could resume again Raiding his toolbox Steve rummaged with fury Never before had he worked with such hurry Widgets and gidgets and gadgets did fly Until the optical isolator caught his eye With sweat on his brow replaced the foul node And threw the failed unit in the commode The power switch flipped in a flash of commotion Steve prayed for stoplights to begin their motion Despite his best hope, the darkness remained The Christmas disaster his job record stained Down to his knees he fell in despair Until he heard sleigh bells filling the air Up he looked, and to his surprise Eight reindeer and sleigh greeted his eyes Can it really be? Can Santa be real? I thought Santa was a a mythical deal! This santa was young, and he had no fear He had the tools of a data engineer Steve looked at his nametage, now he could see This was Mike Fahrion from BB! Saving your bacon today is my job Restoring dude-ville, yessiree Bob I have now come because one thing you forgot Without changing bias, terminate you must not! He lectured to Steve all about termination Bias and cost cuts and infatuation And then with a wave of his hand he did turn The stoplights of dude-ville to once again burn All around dude-ville the happiness spread The cars and the trucks
[Asterisk-Users] Simulate back impulse
Hi I have a asterisk voip box connected to a classic pbx. The pbx use telecom back impulse (bad translation ?) for billing. To have all my billing done by the pbx I need to send back impulse to pbx from asterisk. Is it possible to simulate telecom back impulse with asterisk ? Thanks for your help. Jerome Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] s and i in context not invoked
Hi, Just made a simple test to see how the two extensions (s and i) worked but for some reason I can't seem to make then act as I would like them to. I pick up the phone and dials 100 or 200 - and in the CLI it prints out what ever I have put in the Noop() If i dial any other number, nothing happens - no indication in the CLI. Souldn't the s or i context be activated when I dial a non-existent number? I'm running Asterisk 1.0.2 Sip.conf: [741] type=friend context=test username= secret= canreinvite=no host=dynamic dtmfmode=rfc2833 Extension.conf: [test] exten = s,1,noop(s context) exten = s,2,hangup exten = i,1,noop(i context) exten = i,2,hangup exten = 100,1,noop(+++) exten = 100,2,hangup exten = 200,1,noop(---) exten = 200,2,hangup -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ 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] RE: MD110 and analog trunks
First of all sorry for my English. I'm also interesting in ICU and MD110 on Asterisk. Is there anyone who knows if there is support for this on * or to find the specification of ICU somewhere? Regards, Alberto Ribagorda ___ 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] CPU spikes with wcfxs loaded
After reading your advice I replaced the Via chipset asus board (Sempron) by a Intel chipset MSI board (Celeron). Compiled the same kernel (2.6.8.1) for new hardware, compiled asterisk (in bristuff-0.0.2 distribution), spandsp. Spandsp now works, before it doesn't ! vmstat 1 shows me, that I get almost everytime the same number of interrupts (from 17034 to 17056) procs ---memory-- ---swap-- -io --system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 1 0 0 91388 104828 14610000 0 0 17034 169 0 0 100 0 0 0 0 91388 104836 14610000 012 17035 177 0 0 100 0 0 0 0 91388 104836 14610000 0 0 17034 169 0 0 100 0 Compared to my via chipset asus mainboard, the number of interrupts are the same like above, except when the harddisc is active. Then it drops to about 16050. (bo!=0). hdparm -u1 /dev/hda (on via chipset asus system) avoids this, so number of interrupts stay almost constant. With that I got a little more of a fax page, before it failed. Setting pci latency (on via chipset asus system) by setpci -v -s XX:X.X latency_timer=00 also helped me a little, since as best result, I got almost 75 % of a fax page, before it failed. But with all these tricks, I never got a whole page (or even a several pages fax). Only by switching to the intel chipset mainboard the timing problem seems to be resolved. Thank you very much Craig Guy, for your advice! If more readers experience the same problems with VIA chipset mainboards only, we should add a warning to http://www.voip-info.org/wiki-Asterisk+Hardware Bye, Stephan Schiessling Craig Guy wrote: If you are using an Athlon then you might have a VIA chipset and apparently non-intel chipsets can have these sorts of interrupt problems (Via especially). Try changing to an intel chipset motherboard. Craig - Original Message - From: Michael Welter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, December 14, 2004 5:57 AM Subject: [Asterisk-Users] CPU spikes with wcfxs loaded I need to reopen this discussion because it's impossible to run spandsp (and VoIP) under these circumstances. With zaptel unloaded, I see the following vmstat 1 output: no swapping, an occasional disk output, +/- 1003 interrupts/sec., less than 10 context switches/sec., CPU idle 100%. A very quiet system. I load modules zaptel and wcfxo, and the system utilization stays the same. When I load wcfxs, the number of interrupts goes up to +-2004, which is normal. However, every three seconds the CPU spikes to 50%. This is system utilization, not userland. I assume it's in a wcfxs interrupt. The number of interrupts stays constant at about 2004 during each spike, leading me to the conclusion that the TDM card is holding an interrupt for 500ms every three seconds (50% of 1000ms is 500ms). This is a disaster for spandsp and VoIP in general. When I unload the wcfxs module, CPU idle goes back to a constant 100%. The TDM22B card is REV E/F, and I've tried it with several different cards. Fedora Core 3 with linux-2.6.9 downloaded from kernel.org (a stock kernel). The CPU is Athlon K7. Can anyone please give me a clue? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] ilbc and asterisk 1.0.3 - strange noises.
Title: OpSign I am using RedHat 7.2 and this noises on the codec started after I updated GCC to 3.0.4, downgrading it to gcc 2.96 made it work well again. I know, it's time to upgrade de distrubution, but it's running very stable so far, so why change... Thanks. Alessandro Ren wrote: Have someone experienced any strange noises using the ilbc codec after upgrading to asterisk 1.0.3? I had to change the codec do gsm to fix this problem. The noise is very loud, like saturation of the echo ro something, seems like the echo cancelation is amplifying itself. I'be been using ilbs since asterisl 0.70 and have never had any problem like this. Thanks. -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ 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 -- OpSign __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ 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] Disabling ! command
Hi, since I run asterisk as root with a CLI open on TTY12 I was wondering if the ! (shell) command can be disabled from the config, for safety reasons it seems me usefully. Tnx for any help ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ 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] Polycom SIP Phones
snip Why get the IP 500 when you can get the IP 600 for less? Check out www.tritechcoa.com. They have the IP 600 for $255. But, I think that this stuff should go to the -biz lists. /snip I recently bought a bunch of IP500s and before shipping / tax they were $170 / each (including power supply). We are lucky to have received such a great discount, but there's no reason to pay more than $200 for an IP500. -Ron___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugtracker Karma Hall Of Fame
At 09:01 PM 12/16/04, you wrote: Paul Crick wrote: But seriously, if you think you're owed karma for something and haven't received it, flag it to a bug marshall. I'm not one, I just did the web stuff. funny thing that karma stuff, you are never owed any, you just keep doing good stuff to prevent any bad stuff from hunting you down Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Troubleshooting Asterisk
Guys,
Ok nowhere near as complex as most of the
discussions on here ( ex telco engr for 18 years here). But thought Id
ask for some assistance.
Have just set up my first * Pbx having a play with
it and a couple of Cisco 7960 (configured as SIP) phones.
The phones are tftping into the server ok, and
picking up the configs all ok.
Everything _seems_
to be working, but I cant make any calls either internally or
externally
(apologies in advance for the copious code below)
Having a look, I have placed the following lines into the
extensions.cfg file to allow for the extensions to work
[2001]
exten = 2001,1,Dial(SIP/2001,15,t)
exten = 2001,2,Voicemail(u2001)
exten = 2001,102,Voicemail(b2001)
exten = 2001,103,Hangup
[2002]
exten = 2002,1,Dial(SIP/2002,15,t)
exten = 2002,2,Voicemail(u2002)
exten = 2002,102,Voicemail(b2002)
exten = 2002,103,Hangup
then also in extensions.cfg, I have also set these to allow
connection to voiptalk:
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten =
_00.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:2})
exten =
_09XX,1,Dial(IAX2/[EMAIL PROTECTED]/$EXTEN})
Then, since Im using an IAX connection to voiptalk :
[voiptalk]
type=peer
username=USERID
secret=PW
host=iax.voiptalk.org
Sip config :
[2001]
type=friend
host=192.168.1.151 (phone IP address)
username=2001
secret=
context=from-sip
nat=yes
callgroup=2
pickupgroup=2
mailbox=2001
(and then the same other than the IP addr for extension
2002)
And the last things are the Phone configs from the TFTP
files :
(Example is a basic one for one of the phones)
Line1_name : 2001
Line1_authname: 2001
Line1_password:
Now I can call the extensions from the console they ring,
and I can answer.
--Executing Dial(OSS/dsp,
SIP/2001|15|t) in new stack
--called 2001
--SIP/2001-c7b1 is ringing
--SIP/2001-c7b1 answered OSS/dsp
Console call has been
answered
Dec 17 12:26:26 NOTICE[7078] : rtp.c:1193
ast_rtp_raw_write: RTP Transmission error to IPADDR:23658: Network in
unreachable
(plus another 12 messages the same)
==Spawn extension (local, 2001, 1) exited
non-zero on OSS/dsp
Hangup on console
Anyone got any ideas? Since its my first setup, its
probably something glaringly obvious that Ive done wrong. But Im
starting to go stir-crazy about it
Thanks in advance
Paul
___
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] can't intstall the webmin module
Hello out there, i have little problem with putting the asterisk webmodule in webmin. Im kinda new with asterisk but webmin (latest version, fedora c2) is clear to me. Tried all the options to install with varios files (web, ftp local) but it keeps bugging me with this message: Failed to install module from http:// xxx /asterisk/webmin/webmin.tgz : Module webmin is missing a module.info file where xxx is tested from varios (http/ftp) sites. The folowing lines are in the module.info file: name=ASTERISK category=servers desc=ASTERISK PBX I'm drawing blank on this on, does somebody know the solution? Thnx in advance, Cor Kuijt [EMAIL PROTECTED] Nederlands. ___ 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] Cisco 7905g TFTP Configuration
I recently got a 7905G w/ Sip software preloaded. I got it working w/ asterisk with no problem setting it up through the phone. I am now trying to make it download the config file from the tftp server. I have set all of the options in the file and the file is definately named correctly. But the phone is simply not processing the config file for some reason. Two commands Im trying to get it o process is UIPassword and AudioMode It ignores everything though. I can post it here if you'd like ___ 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] Troubleshooting Asterisk
On Fri, 17 Dec 2004 12:32:30 - Paul Brock [EMAIL PROTECTED] wrote: Everything _seems_ to be working, but I cant make any calls - either internally or externally. I asume you can't place calls from the Cisco's... you need a context in the extension.conf for them. In sip.conf you tell them to use the [from-sip] context - that context should be in the extension.conf eg.: extension.conf: [from-sip] include = 2001 include = 2002 This allows the Cisco's to dial eachother. Hope this helps. -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ 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] erroneous errors - registration fails for grandstream phones
Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to be erroneous since the phones/extensions apparently are working. example of 1 phone, but it happens with both: *** from asterisk CLI -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 Dec 17 08:01:59 NOTICE[22259]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' -- Saved useragent Grandstream BT100 1.0.5.20 for peer 40852 -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 The date obviously changes *** from /var/log/asterisk/messages Dec 17 08:01:59 NOTICE[22259]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' The phones appear to work no traffic on the server 3Ghz P4 512MB RAM 75GB Free Disk Space Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Cisco CallManager Integration
Message: 10 Date: Thu, 16 Dec 2004 17:13:33 -0600 From: Adi Linden [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Cisco CallManager Integration To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi As a few of the others on this list knows, this is something I've been fighting with for a few months now- and it's not such an easy thing. So far, I've not had complete success- but I'm willing to give a quick status report here, and I will post the solution back to the wiki once I do get there. Right now, using gnugk, 1.0+ stable, and chan_h323, I've managed to successfully dial from CCM 3.3 into Asterisk- I use this in production every day today, and I expereince few if any problems with it. 1.02 fixed a few things that caused hanging h.323 channels, so I'd recommend starting there if you need stable.. I am NOT able to call from Asterisk back into CCM at all- CCM drops the call like a hot rock, under either gateway OR gatekeeper configuration. If you're more willing to be on the bleeding edge, the CVS-HEAD from yesterday (not before) , is able to place a call from Asterisk to CCM. Thanks to Snewpy (and I presume JerJer) for taking it this far. The bad news is that, as of last night, it's still not working properly. Configured in gatekeeper mode, I can ring the phone- but there are RTP problems, resulting in one-way audio and early (2 second) call termination. Under Gateway configuration, the result is much the same as under stable- Asterisk sends the SETUP message, and CCM tells it to go away. Right now, the best clue I have on this is that CCM only thinks of Asterisk as an IOS H.323 gateway- and it would appear that IOS gateways perform some sort of registration back to CCM when they start up- Asterisk doesn't, and I think the cause is rooted here. I'm hoping to get a good packet dump off an IOS gateway today to see what the differences are- but there's a lot of distance to go between packet traces and solution. I'm also unable to try the SIP trunk method, since I'm at CCM 3.3.3- but it would appear that it works, based on the wiki. If you want to know how I configured it under stable, contact me offlist and I'll be happy to help. -pbd ___ 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] Cisco 7905g TFTP Configuration
Matthew, If you post the TFTP Server log, maybe it will give us a little direction to help. Last week I upgraded my 7912G and it downloaded the config file with out any problems. I have a few 7905, but I just have not used the TFTP server for configuring them, just used the TFTP to upgrade the firmware. Which Sip version are you using? Have you tried just making the changes thru the web interface? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Marlowe Sent: Friday, December 17, 2004 5:09 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7905g TFTP Configuration I recently got a 7905G w/ Sip software preloaded. I got it working w/ asterisk with no problem setting it up through the phone. I am now trying to make it download the config file from the tftp server. I have set all of the options in the file and the file is definately named correctly. But the phone is simply not processing the config file for some reason. Two commands Im trying to get it o process is UIPassword and AudioMode It ignores everything though. I can post it here if you'd like ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 ___ 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] Asterisk and HylaFax
Title: Mensaje Hi all, again I try configure Hylafax with asterisk. I would like configure Asterisk in the next way: 1)An incoming fax go into through X100P 2)Asterisk detects Fax and redirect fax to Hylafax Is it possible? Any idea woluld be great idea? regards, srsergio -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.5.4 - Release Date: 15/12/2004 ___ 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] OT: DSL without voice
A lot of people are going for the VOIP only approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ 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] Hardware based DSP
[EMAIL PROTECTED] wrote: Thank you all for your explanations related to my question. I have one follow-up question though. When I said that all dsp related stuff has to be handled by software within asterisk, I was thinking of conferencing at the time. I mean in order to be able to conference a sip session with a PSTN call, it would have to be handled by software, even if both the channels had hardware dsp capabilities. Right ?? Hmmm. Ultimately, yes, I suppose there's always going to be some of that. Where the DSP work gets really intensive is when what is going into the DSP is quite different from what's coing out. If a hardware device existed for Asterisk to transcode all channels into a common format, so that internally the same codec was being used, then the CPU would have relatively little DSP work to do when connecting them together. I don't know if this would be a problem in other areas, so what I'm saying is more of a brainstorm than anything I think needs to happen. If you are dealing with just a single channel, then the driver may handle codec/echo cancelation stuff with hardware help (??) That sounds pretty much correct. I know there's been some talk about using the powerful floating-point capabilities of 3d video cards to do transcoding; other ideas are in the works as well. As an aside, what is the best way to go about learnig about the aritecture of asterisk, other than using the source ?? There simply isn't enough documentation in that regard. At least not that anyone's found in a single repository . . . other than the wiki, of course. The Wiki pages are great, but I have not (yet) found any info about asterisks architecture itself (in depth that is). Nor has anyone, to my knowledge (other than reading the source). Some linked websites / blogs provide good info on some topics, but is there a good high level design doc available anywhere ? The Asterisk Documentation Project has that very thing as one of it's goals. It's a big job, and there are few of us, so it's not happening as fast as everyone would like. There is so much to write, and so little time . . . It would stop people like me asking so many basic questions in the -dev list. There's no doubt that more documentation is needed. ___ 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] OT: DSL without voice
A lot of people are going for the VOIP only approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? SDSL service is delivered on a dedicated circuit. You should never need to have an analog voice circuit before ordering SDSL. (So is, for that matter, HDSL, though that is frequently misrepresented as T1 service) There are also a small number of providers, like Speakeasy, who are now offering ADSL-over-bare-copper services (I believe they call theirs Onelink). There is really nothing prohibiting DSL providers from doing this - they just end up paying Ma Bell for the entire cost of the copper, and that's not really all that popular. Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Definity PBX with a T100P TN767E
Hey everybody, I'm looking for some information that I'm not finding by searching the 2004 Asterisk archives. I'm currently playing with a Digium T100P card and 2 Grandstream phones, things are working well. I wanted to move on to linking our Definity G3R Rev 8.2 to the T100P. Everything that I've read so far shows that you need a TN464 to accomplish this. We have a TN767E available. Is there any way to do this without the TN464? The phone admin wanted to know if there was a way for the T100P to do just a blind dump of a call to the Definity without having to have the data channel. Or, is there some documentation of how to properly setup a Definity G3R with a TN767? I've seen some references to this working, but nothing definitive. I believe our TN767 ROM version is 10. If it's not possible, anybody have a link to a reseller of used TN464s? I found a couple sites, but they don't list prices. Thanks, Doug Lytle ___ 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] RE: Meetme with video???
I wonder if there is an application available, what would allow me to have a conference call (meetme) with video. Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this functionality: http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+video+conferencing You can add to this bounty, if you want. I'm trying to convince the money people at my company that we should add $500 to this. BTW: Is anybody working on this? ___ 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] Forcing E.164ID with chan_h323 or chan_oh323
I am trying to figure out the correct way to send my E.164 ID with chan_h323 and or chan_oh323 as my H323 provider requires this in the format of 'account-pin'. With chan_oh323 I have been able to register with the gatekeeper and can recieve incomming calls, but outgoing calls do not work. With chan_h323, I can call H323 clients (netmeeting, ATAs etc) but cannot place a call through my providers gateway. I have tried playing around with setting manually my CID, however the call fails every time. My provider has told me they aren't recieving my E.164 ID... Any ideas? ___ 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] Masive Fax Sendig with spandsp
Anybody have expiriance with masive fax sendig with spandsp I have PRI E1, plan to bye Digium E1 card and to send 30 fax's in the same time. Any working solution? P.S. Please, need for Yesterday. ___ 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] OT: DSL without voice
On December 17, 2004 09:00 am, Joe Greco wrote: (So is, for that matter, HDSL, though that is frequently misrepresented as T1 service) Also considering that almost every T1 you order these days is being delivered on HDSL2 (1 copper pair) it further muddies the waters. :-) -A. ___ 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] Troubleshooting Asterisk
I asume you can't place calls from the Cisco's... you need a context in the extension.conf for them. In sip.conf you tell them to use the [from-sip] context - that context should be in the extension.conf eg.: extension.conf: [from-sip] include = 2001 include = 2002 This allows the Cisco's to dial eachother. Hope this helps. Michael, Many thanks - have added this, but strangely enough it still doesn't work phone-phone :( /me continues to play Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] native MOH with Asterisk 1.0.3
Let me jump in. Seems that the ChanSpy patch worked just fine in pre-1.0.x. Provided MOH plus a bunch of there useful stuff. Now it seems it's gone in 1.0.3 and scant little info on why or when (or if) it will be back. Any insight is requested. Jon - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 16, 2004 8:55 PM Subject: Re: [Asterisk-Users] native MOH with Asterisk 1.0.3 Kristian Kielhofner wrote: What is the bug ID for moh stop? It's 3035. The current native moh patch in Mantis definitely will not apply to 1.0.x. I believe that anthm removed my posted patch from the bug, so it's no longer available there. If you want it I can try to scrounge up a patch that will work for you, but I don't run 1.0.x here so I can't make any promises. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: DSL without voice
Most RBOC's ILEC's have in their tariffs that the DSL subscriber MUST have a working POTS line before they can be sold DSL. Tom Chandler - Original Message - From: Joe Greco [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 17, 2004 8:00 AM Subject: Re: [Asterisk-Users] OT: DSL without voice A lot of people are going for the VOIP only approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? SDSL service is delivered on a dedicated circuit. You should never need to have an analog voice circuit before ordering SDSL. (So is, for that matter, HDSL, though that is frequently misrepresented as T1 service) There are also a small number of providers, like Speakeasy, who are now offering ADSL-over-bare-copper services (I believe they call theirs Onelink). There is really nothing prohibiting DSL providers from doing this - they just end up paying Ma Bell for the entire cost of the copper, and that's not really all that popular. Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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 --- Scanned by Bayou Internet for all known viruses. http://www.bayou.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: DSL without voice
The Georgia PSC requires RBOC's and LEC's to unbundle pots from dsl. most other states' PSC's arent as progressive. On Friday 17 December 2004 08:29, Tom Chandler wrote: Most RBOC's ILEC's have in their tariffs that the DSL subscriber MUST have a working POTS line before they can be sold DSL. Tom Chandler - Original Message - From: Joe Greco [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 17, 2004 8:00 AM Subject: Re: [Asterisk-Users] OT: DSL without voice A lot of people are going for the VOIP only approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? SDSL service is delivered on a dedicated circuit. You should never need to have an analog voice circuit before ordering SDSL. (So is, for that matter, HDSL, though that is frequently misrepresented as T1 service) There are also a small number of providers, like Speakeasy, who are now offering ADSL-over-bare-copper services (I believe they call theirs Onelink). There is really nothing prohibiting DSL providers from doing this - they just end up paying Ma Bell for the entire cost of the copper, and that's not really all that popular. Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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 --- Scanned by Bayou Internet for all known viruses. http://www.bayou.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chad Whitten Network Administrator neXband Communications [EMAIL PROTECTED] 601-944-4801 Phone 601-944-4803 Fax ___ 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] OT: DSL without voice
Joe Greco schrieb: Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. For this you can take your mobile. When my local company (T-Com) decides to allow ADSL without a phone line I will take it. I've got my mobile for cases of emergency. And since in germany there is really no danger of a hurricane the stability of the mobile nets should be sufficient. ;-) Bye! Michael ___ 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] My Boss wants background music!!!!
Why not just dial an extention for music when the user wants music from there desk. The requirement of the original poster was to mute the music at the desk when a call is in progress. It would be really nice if there was a hardphone capable of accepting a multicast high-quality stream when no call was in progress. ___ 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] Has anyone connected to 7960 withconsolecablefor setup?
Randy, Same problem here once (or more). I ended up using TFTP to load a new image. The pone I got was programmed for non-SIP. I went into config and found the TFTP IP, set up a tftp server to load a new config at boot. Be sure your not trying to load too high a version. SIP 7.x WON'T load directly into a much lower firmware version. I needed to do a two step method. First load a 6.0 image, then the 7.x image. A actually used my laptop as the tftp server as I can change the IP addr at will without out screwing up the rest of my asterisk net. Jon - Original Message - From: Randy MacKay [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 16, 2004 3:29 PM Subject: RE: [Asterisk-Users] Has anyone connected to 7960 withconsolecablefor setup? The phone is a used one I picked up from ebay. **# doesn't seem to unlock anything. The display of the phone says; Configuring VLAN, Configuring IP, (requesting ??? flashes), TFTP ??.cfg.xml , Protocol Application Invalid. If I could just somehow get to the TFTP Settings? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Brock Sent: Thursday, December 16, 2004 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup? Randy, Is it a new unit? The only reason I ask is that hitting the settings button should let you straight in. There is an Rs232 port on the bottom - however not oversure what it's used for on the 7960's. The reason I as wether it's new or not is that it might need firmware resetting as per the cisco information (not immediately to hand). If you can see the menu's and just chance change the setting, I think it's something like *# or **# to allow change. Sorry if that's suck egg territory - just trying to cover anything obvious which is easily missed!! Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay Sent: 16 December 2004 18:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup? I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. Anyone have any ideas? Thanks, Randy I found a link to make a Cisco Console Cable for RJ-45. http://www.hardwarebook.net/cable/serial/ciscoconsole9.html DB9F RJ45 Receive Data 2 3 Transmit Data 3 6 Data Terminal Ready 4 7 Ground 5 4 Ground 5 5 Data Set Ready 6 2 Request to Send 7 8 Clear to Send 8 1 The Console Access Manual, give the following cable information: Console Cable Requirements You use a serial cable with a connector to connect a PC and a phone. The cable uses an RJ-11 connector for the phone and an RJ-45 connector to an RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements for the console cable. Table D-1 Console Cable Pinouts RJ-11 Connector RJ-45 Connector Pin 2 == Pin 6 Pin 3 == Pin 4 Pin 4 == Pin 3 So, I thought I would go right from DB9F to RJ-11 DB9F RJ-45 RJ-11 Pin 2 Pin 3 Pin 4 Pin 5 Pin 4 Pin 3 Pin 3 Pin 6 Pin 2 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 ___ 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] erroneous errors - registration fails forgrandstream phones
Hi, Look in your sip.conf host=192.168.20.2 and your phone is set to use 192.168.20.25 try to change host directive in sip.conf to host=192.168.20.25 Diego Aguirre - Original Message - From: Greg - Cirelle Enterprises [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 17, 2004 11:25 AM Subject: [Asterisk-Users] erroneous errors - registration fails forgrandstream phones Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to be erroneous since the phones/extensions apparently are working. example of 1 phone, but it happens with both: *** from asterisk CLI -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 Dec 17 08:01:59 NOTICE[22259]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' -- Saved useragent Grandstream BT100 1.0.5.20 for peer 40852 -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 The date obviously changes *** from /var/log/asterisk/messages Dec 17 08:01:59 NOTICE[22259]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' The phones appear to work no traffic on the server 3Ghz P4 512MB RAM 75GB Free Disk Space Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling ! command
you could comment that portion out and rebuild? On Fri, 2004-12-17 at 13:15 +0100, Alessio Focardi wrote: Hi, since I run asterisk as root with a CLI open on TTY12 I was wondering if the ! (shell) command can be disabled from the config, for safety reasons it seems me usefully. Tnx for any help ! ___ 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] OT: DSL without voice
Most RBOC's ILEC's have in their tariffs that the DSL subscriber MUST have a working POTS line before they can be sold DSL. So, let me get this straight, /they/ chose to put in /their/ tariffs that the customer must have a working POTS line, then /they/ use that as a reason to sell more POTS lines. Why am I not crying for the ILEC's here? I am not interested in their desire to sell POTS lines through extortion. Neither are many public utilities commissions, who are generally dismantling such rules. Incidentally, such tariffs do NOT impose such a requirement on SDSL or HDSL, at least around here, and now in a lot of areas we're seeing deployment of services like OneLink, where the customer just pays a little extra for the cost of the dedicated copper. ILEC's are generally full of , by the way. Around here, they were charging the CLEC's something like $10/month for rental of the copper to provide service (POTS, whatever). SBC whined and whined that this was far below their cost and proposed a minor readjustment of only a little more than double, up to something like $22. Note that many direct SBC customers get their entire phone service for less than $22, so I'm not sure how it is that leasing the wires at wholesale rates should cost more than actually providing full service to retail customers. Hell, they used to provide dry copper for just a few bucks a month, sold in fairly large quantities to alarm companies, etc... http://www.jsonline.com/bym/news/mar04/214278.asp I have very little sympathy for the ILEC's. I would probably be fine with seeing their physical plants taken away from them, sold to a highly regulated company that was chartered only to provide wholesale wire services, and then have everyone rent copper at fair prices - including the ILEC. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Paris Meeting Date/Time/Location
01.45.39.59.98). However, we have to let them know exactly how many people will attend, so PLEASE RSVP as soon as possible. I'm in! This is just a few blocks from where I live, cool! ___ 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] Re: VOIP Phone Suggestions
On 15/12/04 22:53 -0600, Kevin Curtis wrote: I would recommend Uniden UIP200 phones. Great sound quality with inbuilt phone book, call logs etc works great with asterisk. I recently purchased from [1]www.qualvoip.com (they also provided me sample configuration files for asterisk). Kevin One gripe about these guys - They clearly use * for their PBX product, which looks like it's not much more than * with a web based config interface. There's not one mention of * on their site! No, there's nothing wrong with that legally but they should be giving props to * instead of promoting it as their PBX software. Instead of calling the product The Asterisk Based PBX System they call it The Open System Based PBX System. Are they afraid that potential customers will discover * and try to do it on their own? Jason ___ 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] My Boss wants background music!!!!
On Friday 17 December 2004 14:31, Steven Kalcevich (Lists) wrote: Why not just dial an extention for music when the user wants music from there desk. Because then the phone will be engaged on a call and will not ring when someone else wants to talk to the person at the desk? Antony. The requirement of the original poster was to mute the music at the desk when a call is in progress. It would be really nice if there was a hardphone capable of accepting a multicast high-quality stream when no call was in progress. -- I own three Windows books, published by O'Reilly. They are Windows Annoyances, Office 97 Annoyances and Windows 98 Annoyances. That pretty much sums it up for me. Please reply to the list; please don't CC me. ___ 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] OT: DSL without voice
Joe Greco schrieb: Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. For this you can take your mobile. When my local company (T-Com) decides to allow ADSL without a phone line I will take it. I've got my mobile for cases of emergency. And since in germany there is really no danger of a hurricane the stability of the mobile nets should be sufficient. ;-) I do think the thing that worries me about this trend is the unexpected scenario. Right now, we have a fairly high quality E911 system (dunno about where you are) and people expect that they can dial 911 and the right things happen. So what if you've got some friends visiting your house and you have a heart attack and no 911 on your POTS-via-VoIP? Are they expected to know your cell phone's unlock code? Are they required to bring their own cells as a prerequisite for visiting? Or is it acceptable for them to have to go finding a neighbor who has a usable POTS phone? I know it's *unlikely*, but emergencies always are. I agree with your argument for non-emergency purposes: the advent of good cell phone service may mean the demise of many landlines, as cells may be more practical for some users. As early adopters, we may not have any good solutions, and that may in fact be fine, as long as we know it. It's just worth thinking about... ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Troubleshooting Asterisk
On Fri, 17 Dec 2004 14:14:55 - Paul Brock [EMAIL PROTECTED] wrote: Many thanks - have added this, but strangely enough it still doesn't work phone-phone :( /me continues to play Could you post the output from the CLI (with verbose level at 4 or so) it might give up some clues. -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ 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] German Howto?
Hello, I'm a new user of asterisk. I'm looking for a german or simple english howto, or a description for the following configuration. City One: ISDN telefonadapter with ISDN and analog devices | Server with a Elsa MicroLink ISDN/PCF Card | Connection via VPN or direct, as needed | Server with a Elsa MicroLink ISDN/PCF Card | City Two: ISDN telefonadapter with ISDN and analog devices I've also a SIP account, which I want to integrate. I want that both cities are able to call each other, or make calls via the SIP account. The first step I want to try, is that calling our SIP No. make my ISDN Phone here ringing. Is this all possible? Thanks for any help or hint Daniel ___ 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] Newbie setup question (Voicepulse, FWD IAXTEL)
Okay, I can receive calls through Voicepulse fine. All the various attempts (too many to list) to create a workable configuration to Dial to Voicepulse has failed, from 403s to No authority found to nothing. The Voicepulse folks told me that the open access was SIP and I shouldn't have a reference in the iax.conf file, but then said that they were refering my question to the Voicepulse Connect folks whose examples clearly show an entry in the iax.conf file. I have an Asterisk 1.0.2 running under Aurora (Sun Ultra 5). Also, when I try to configure either FWD and/or Iaxtel according to the examples on the Wiki, I'm getting Bus Errors and a core dump. Can anyone help with working examples of Voicepulse, esp. if FWD and/or iaxtel are included ? THANKS ! -- Ken M Edwards, N4ZBB __ Switch to Netscape Internet Service. As low as $9.95 a month -- Sign up today at http://isp.netscape.com/register Netscape. Just the Net You Need. New! Netscape Toolbar for Internet Explorer Search from anywhere on the Web and block those annoying pop-ups. Download now at http://channels.netscape.com/ns/search/install.jsp ___ 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] Troubleshooting Asterisk
Many thanks - have added this, but strangely enough it still doesn't work phone-phone :( /me continues to play Could you post the output from the CLI (with verbose level at 4 or so) it might give up some clues. Certainly - and many thanks. Not a problem - what info would be useful to post? Paul ___ 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] Shorten the recognition time of rings on Wildcard X100P
Hi Michael, I connected my Wildcard X100P to the PSTN and created a context in extensions.conf which rings a number of SIP phones on inbound calls from the PSTN. When I compare the actual PSTN rings with Asterisk recognition of the incoming call, Asterisk rings my SIP phones on the third ring of the incoming call. Add usecallerid=no to your zapata.conf. The caller id is detected between the first and second ring. If you don't detect it the system can put the call through immediately. Thank you! Since I am not subscribed to caller id on this pstn line it won't be any hardship. I'd rather have the line processed immediately. Thanks, Adi ___ 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] Voicemail.Conf
When I specify the users voicemail can I specify more than one email address to send the recording to once its finished? Also can I set it where it only emails the voicemail recording and not stores it local to the * box? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct ___ 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] Troubleshooting Asterisk
On Fri, 17 Dec 2004 15:09:50 - Paul Brock [EMAIL PROTECTED] wrote: Certainly - and many thanks. Not a problem - what info would be useful to post? Paul No problem at all :-) Just the output it makes as you try to call from one Cisco to the other. -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ 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] OT: DSL without voice
On Fri, 17 Dec 2004, Joe Greco wrote: Joe Greco schrieb: Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. For this you can take your mobile. When my local company (T-Com) decides to allow ADSL without a phone line I will take it. I've got my mobile for cases of emergency. And since in germany there is really no danger of a hurricane the stability of the mobile nets should be sufficient. ;-) I do think the thing that worries me about this trend is the unexpected scenario. Right now, we have a fairly high quality E911 system (dunno about where you are) and people expect that they can dial 911 and the right things happen. So what if you've got some friends visiting your house and you have a heart attack and no 911 on your POTS-via-VoIP? Are they expected to know your cell phone's unlock code? Are they required to bring their own cells as a prerequisite for visiting? Or is it acceptable for them to have to go finding a neighbor who has a usable POTS phone? This random thought just popped into my head: Seems like I've read that any cell handset will place a 911 call, regardless of whether it is associated with a valid and paid-up account. Is that true? If so, then maybe we could just attach GSM interfaces to our asterisk box to provide communications in the unlikely emergency (so long as the LAN and * box have power to operate, that is). Whaddaya think? Greg ___ 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] OT: DSL without voice
www.Covad.com I have their TeleSoho dedicated loop DSL. It costs the same as the bundled loop. Michael On Fri, 17 Dec 2004 05:43:36 -0800 (PST), Steve Edwards wrote: A lot of people are going for the VOIP only approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] OT: DSL without voice
On Fri, 17 Dec 2004, Joe Greco wrote: Joe Greco schrieb: Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. For this you can take your mobile. When my local company (T-Com) decides to allow ADSL without a phone line I will take it. I've got my mobile for cases of emergency. And since in germany there is really no danger of a hurricane the stability of the mobile nets should be sufficient. ;-) I do think the thing that worries me about this trend is the unexpected scenario. Right now, we have a fairly high quality E911 system (dunno about where you are) and people expect that they can dial 911 and the right things happen. So what if you've got some friends visiting your house and you have a heart attack and no 911 on your POTS-via-VoIP? Are they expected to know your cell phone's unlock code? Are they required to bring their own cells as a prerequisite for visiting? Or is it acceptable for them to have to go finding a neighbor who has a usable POTS phone? This random thought just popped into my head: Seems like I've read that any cell handset will place a 911 call, regardless of whether it is associated with a valid and paid-up account. Is that true? If so, then maybe we could just attach GSM interfaces to our asterisk box to provide communications in the unlikely emergency (so long as the LAN and * box have power to operate, that is). Whaddaya think? In five years, when GPS cell phone location services are mature and stable, this is probably a fairly good solution. Until then, it suffers the same problems as contemporary 911-via-cell service. :-/ It's that whole early adopter thing again. Heh. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Masive Fax Sendig with spandsp
On Fri, 2004-12-17 at 08:12 -0600, Tomislav Avramovic wrote: [snip] P.S. Please, need for Yesterday. Then you should have asked before yesterday. ___ 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] OT: DSL without voice
www.Covad.com I have their TeleSoho dedicated loop DSL. It costs the same as the bundled loop. ADSL or SDSL? (I haven't looked at Covad's pricey offerings for a while) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Definity PBX with a T100P TN767E
I'm currently playing with a Digium T100P card and 2 Grandstream phones, things are working well. I wanted to move on to linking our Definity G3R Rev 8.2 to the T100P. Everything that I've read so far shows that you need a TN464 to accomplish this. We have a TN767E available. Yes, a TN767E will work and actually a TN464 may not, depending on how the G3 is setup. If I remember right the TN464 needed a different clock set up then we had on our system. I've got my G3 working with asterisk using a TN767E (v18 R11 - Ebay $100, gotta love Ebay). Inbound/outbound, DID from G3 inbound, ext./ext., etc. You just have to make sure your G3 has a spare proc. interface and of coarse you already have PRI ($ feature enabled) on the G3, right? Here's some notes from when I did mine, hope they help you. Hotplug cp (purple slot) in spare slot, ie. 01A06 add DS1 01A06 display DS1 01A06 add data module with type of 'procr-intf' and a non-DID extension number Assign the data module to a physical channel (01 to 04) Do a 'change communications-interface links' to add the information for the ISDN board. Use the same physical channel as assigned to the data module. Enable = n Est Conn = y PI Ext = Data mod created above PROT = ISDN Brd = TN767 slot Identification = whatever Do a 'change communications-interface processor-channels' and add an entry: Appl = ISDN Link = same as assigned to the data module Channel = blank Priority = h Do a 'add or change signaling-group x' Associated Signaling = y for facility associated sognaling n for non-facility associated signaling Primary D cahnnel - 767 slot, port 24 Trunk Group = ? Go back to the 'change communications-interface links' form and enable the link that you are using. Give it a few minutes to sync up and then do a 'status signaling-group x' You should see the primary as 'in-service' ___ 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] Digium Card Error
Lee, Thanks for posting the make config. That did the trick for my setup. Have a great day ___ 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] Has anyone connected to 7960 withconsolecableforsetup?
Something else to remember some of the earlier versions would not recognize more that 8 characters in the file OS79XX.txt had to be modified to P0M30300 and file had to be moved to P0M30300.bin Then at about POM3-06-00 you are force to start with the signed firmware. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] om] On Behalf Of Jon Bebeau Sent: Friday, December 17, 2004 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Has anyone connected to 7960 withconsolecableforsetup? Randy, Same problem here once (or more). I ended up using TFTP to load a new image. The pone I got was programmed for non-SIP. I went into config and found the TFTP IP, set up a tftp server to load a new config at boot. Be sure your not trying to load too high a version. SIP 7.x WON'T load directly into a much lower firmware version. I needed to do a two step method. First load a 6.0 image, then the 7.x image. A actually used my laptop as the tftp server as I can change the IP addr at will without out screwing up the rest of my asterisk net. Jon - Original Message - From: Randy MacKay [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 16, 2004 3:29 PM Subject: RE: [Asterisk-Users] Has anyone connected to 7960 withconsolecablefor setup? The phone is a used one I picked up from ebay. **# doesn't seem to unlock anything. The display of the phone says; Configuring VLAN, Configuring IP, (requesting ??? flashes), TFTP ??.cfg.xml , Protocol Application Invalid. If I could just somehow get to the TFTP Settings? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] om]On Behalf Of Paul Brock Sent: Thursday, December 16, 2004 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Has anyone connected to 7960 with consolecablefor setup? Randy, Is it a new unit? The only reason I ask is that hitting the settings button should let you straight in. There is an Rs232 port on the bottom - however not oversure what it's used for on the 7960's. The reason I as wether it's new or not is that it might need firmware resetting as per the cisco information (not immediately to hand). If you can see the menu's and just chance change the setting, I think it's something like *# or **# to allow change. Sorry if that's suck egg territory - just trying to cover anything obvious which is easily missed!! Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] om] On Behalf Of Randy MacKay Sent: 16 December 2004 18:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup? I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. Anyone have any ideas? Thanks, Randy I found a link to make a Cisco Console Cable for RJ-45. http://www.hardwarebook.net/cable/serial/cisc oconsole9.html DB9F RJ45 Receive Data 2 3 Transmit Data 3 6 Data Terminal Ready 4 7 Ground 5 4 Ground 5 5 Data Set Ready 6 2 Request to Send 7 8 Clear to Send 8 1 The Console Access Manual, give the following cable information: Console Cable Requirements You use a serial cable with a connector to connect a PC and a phone. The cable uses an RJ-11 connector for the phone and an RJ-45 connector to an RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements for the console cable. Table D-1 Console Cable Pinouts RJ-11 Connector RJ-45 Connector Pin 2 == Pin 6 Pin 3 == Pin 4 Pin 4 == Pin 3 So, I thought I would go right from DB9F to RJ-11 DB9F RJ-45 RJ-11 Pin 2 Pin 3 Pin 4 Pin 5 Pin 4 Pin 3 Pin 3 Pin 6 Pin 2 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 _ __ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aste risk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste risk-users
Re: [Asterisk-Users] OT: DSL without voice
On Fri, 17 Dec 2004, Joe Greco wrote: On Fri, 17 Dec 2004, Joe Greco wrote: Joe Greco schrieb: Don't forget, you ought to have a conventional phone line for E911 purposes, including what happens when a hurricane goes through and my ISP becomes toast. VoIP is a neat technology but it lacks the resiliency of the traditional phone system. For this you can take your mobile. When my local company (T-Com) decides to allow ADSL without a phone line I will take it. I've got my mobile for cases of emergency. And since in germany there is really no danger of a hurricane the stability of the mobile nets should be sufficient. ;-) I do think the thing that worries me about this trend is the unexpected scenario. Right now, we have a fairly high quality E911 system (dunno about where you are) and people expect that they can dial 911 and the right things happen. So what if you've got some friends visiting your house and you have a heart attack and no 911 on your POTS-via-VoIP? Are they expected to know your cell phone's unlock code? Are they required to bring their own cells as a prerequisite for visiting? Or is it acceptable for them to have to go finding a neighbor who has a usable POTS phone? This is what I'm struggling with at the moment. I want to set up PSTN - VoIP but I haven't completely settled on how to handle a 911 situation. I can certainly train my family, but others What makes it difficult for me is that I don't get cell service at my house, it's the price I pay for living in the boonies, but a sacrafice I'm willing to make. This random thought just popped into my head: Seems like I've read that any cell handset will place a 911 call, regardless of whether it is associated with a valid and paid-up account. Is that true? Yes, a volunteer firefighter in VT was just arrested because he was making 911 calls on a discarded cell phone he found. It wasn't attached to any service, but he could make all the 911 calls he wanted. He was making false calls of fire and auto accidents to watch the responders. He was finally caught because someone in the 911 service recognized his voice. If so, then maybe we could just attach GSM interfaces to our asterisk box to provide communications in the unlikely emergency (so long as the LAN and * box have power to operate, that is). Whaddaya think? In five years, when GPS cell phone location services are mature and stable, this is probably a fairly good solution. However, this will require some external interface, external meaning outside, for the GPS. I have yet to see a GPS unit that will lock on in a house or building. Until then, it suffers the same problems as contemporary 911-via-cell service. :-/ It's that whole early adopter thing again. Heh. So given all that, I'm looking for ideas and solutions that others have implemented to address this issue. Ed ___ 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] Paris Meeting Date/Time/Location
The Paris Asterisk meeting will be held Monday, December 20, 2004 at 1 p.m. at Les Vendanges--a wonderful restaurant in the 14th (tel Definite yes! ___ 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] Problems with app_realtime
Post this as a bug Brian. -Matthew - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk-users [EMAIL PROTECTED] Sent: Tuesday, December 14, 2004 3:51 AM Subject: [Asterisk-Users] Problems with app_realtime It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql describe sip_buddies; ++---+--+-+-++ | Field | Type | Null | Key | Default | Extra | ++---+--+-+-++ | uniqueid | int(11) | | PRI | NULL| auto_increment | | name | varchar(30) | | UNI | || | accountcode| varchar(30) | YES | | NULL|| | amaflags | char(1) | YES | | NULL|| | callgroup | varchar(30) | YES | | NULL|| | callerid | varchar(50) | YES | | NULL|| | canreinvite| char(1) | YES | | NULL|| | context| varchar(30) | YES | | NULL|| | defaultip | varchar(15) | YES | | NULL|| | dtmfmode | varchar(7)| YES | | NULL|| | fromuser | varchar(50) | YES | | NULL|| | fromdomain | varchar(31) | YES | | NULL|| | host | varchar(31) | | | || | incominglimit | char(2) | YES | | NULL|| | outgoinglimit | char(2) | YES | | NULL|| | insecure | char(1) | YES | | NULL|| | language | char(2) | YES | | NULL|| | mailbox| varchar(50) | YES | | NULL|| | md5secret | varchar(32) | YES | | NULL|| | nat| varchar(5)| YES | | NULL|| | permit | varchar(95) | YES | | NULL|| | deny | varchar(95) | YES | | NULL|| | pickupgroup| varchar(10) | YES | | NULL|| | port | varchar(5)| | | || | qualify| varchar(4)| YES | | NULL|| | restrictcid| char(1) | YES | | NULL|| | rtptimeout | char(3) | YES | | NULL|| | rtpholdtimeout | char(3) | YES | | NULL|| | secret | varchar(30) | YES | | NULL|| | type | varchar(6)| | | || | username | varchar(30) | | | || | allow | varchar(100) | YES | | NULL|| | disallow | varchar(100) | YES | | NULL|| | regseconds | int(11) | | | 0 || | ipaddr | varchar(15) | | | || | ts | timestamp(14) | YES | | NULL|| ++---+--+-+-++ 36 rows in set (0.01 sec) -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.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 ___ 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] OT: DSL without voice
On Fri, 17 Dec 2004 09:42:28 -0600 (CST), Joe Greco wrote: www.Covad.com I have their TeleSoho dedicated loop DSL. It costs the same as the bundled loop. ADSL or SDSL? (I haven't looked at Covad's pricey offerings for a while) ADSL 3.0 Mb down / 768k up. $99/mo. The dedicated loop service requires a professional installation that costs $175 (I think) I was having trouble with the bundled DSL dropping when my home POTS line rang. SBC and Covad were hopeless at diagnosing this, and the unbundled service was available so I simply switched. SBC droppped a clean, new pair to the house. Covad's tech did his install in less than 10 minutes. The also told me that I had to buy their DSL mode/wireless router combo. I did, but the cost was rebated. Then I put my trusty Siemens Speedstream/m0n0wall combination back in the line ;-) Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] asterisk clients (need helpdesk solution)
Asterisk looks great on the server side, but it I'm not sure I've found a good solution for my users on the desktop. I've look at few windows apps, but most are just for configuring asterisk (which I've already done). The users here are used to using an app that lets them see caller id, switch between multiple calls, and has a dial directory. Is there a good client app available for users that would cover at least some of these? tia! ___ 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] Display on OptiPoint400std SIP
Hi all,
I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient.
Any ideas? Thx!
sip.conf
[2005]
type=friend
callerid="OptiPoint" 2005
context=default
host=dynamic
disallow=all
allow=ulaw
allow=alaw
extensions.conf
exten = 2005,1,Dial(SIP/${EXTEN},10,tr)
exten = 2005,2,Congestion
Do you Yahoo!?
Yahoo! Mail - You care about security. So do we.___
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] SNIMTA_SPAM Using the Directory Feature to play a menu
All my user accounts (mailboxes) have been setup. When an outside person calls my asterisk server they can press 7 to hear a directory of extensions starting at 601 and continues. When prompted to enter the first three letters of the last name comes, I enter the letters. When entered the voice for the mailbox is played I am then asked to verify by pressing one (1). Once I do, asterisk fails to dial the extension. And re-reads the options again never getting out of the loop. You can see my extensions.conf file as well as my voice mail at: http://pastebin.ca/3059 I first started by installing asterisk with the fedora core 2 RPM's released by a member of the asterisk community. I was told possibly that those were no good. I then un-installed the RPM rebooted cleaned up and trails, and reinstalled using the latest release of the stable build. This is happening to both installs. I feel it is something I am missing in my conf files but can not pin it down. Thanks for the help. ___ 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] Latest head giving app_queue.c:340 error
Hello, After upgrading to the latest development CVS Head, I am now getting regular errors as follows: Dec 17 17:07:30 WARNING[8092]: app_queue.c:340 changethread: Can't change device with no technology! Also, my ability to answer calls with XTen Pro softphone seems to be a bit flaky now. Any ideas? = Jason Goecke www.goecke.net Ph: +31.707.504.634 Mb: +31.707.504.634 Fx: +31.847.598.006 Alt#s: +1.720.946.6451 (US) /+44.844.986.9270 (UK) [EMAIL PROTECTED] = Jason Goecke www.goecke.net Ph: +31.707.504.634 Mb: +31.707.504.634 Fx: +31.847.598.006 Alt#s: +1.720.946.6451 (US) /+44.844.986.9270 (UK) [EMAIL PROTECTED] ___ 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] Definity PBX with a T100P TN767E
Ken Godee wrote: Yes, a TN767E will work and actually a TN464 may not, depending on how the G3 is setup. If I remember right the TN464 needed a different clock set up then we had on our system. Ken, I printed that out to give to our phone Admin. Thank you very much! Doug ___ 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] Troubleshooting Asterisk
No problem at all :-) Just the output it makes as you try to call from one Cisco to the other. Stupid question on my part, but how do you specify a level to debug at when issuing the debug command??? Currently I'm running a debug, but I suspect it's at too high/low a level, since I'm not seeing a great deal (i.e. debugging is on, but it's not returning anything!!) Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and HylaFax
On 2004.12.17 05:42 Sergio Serrano wrote: Hi all, again I try configure Hylafax with asterisk. I would like configure Asterisk in the next way: 1)An incoming fax go into through X100P 2)Asterisk detects Fax and redirect fax to Hylafax Is it possible? Yes, but it may not be as pretty as you like, and it may not function as well as you hope. Using faxdetect in your zapata.conf file will get practically all of the faxes coming in to the X100P routed to the fax extension. The trick, then, is how to get HylaFAX at that fax extension. There are a number of different ways to do it, but in each case, in the end, the idea is to get HylaFAX and Asterisk communicating at an Asterisk FXS point. So you could do a number of different arrangements: X100P - Asterisk - SPA-2000 (ATA) - Modem - HylaFAX or X100P - Asterisk - TDM400P (FXS) - Modem - HylaFAX but, at that point you probably would be better off without the X100P like this: TDM400P (FXO) - Asterisk - TDM400P (FXS) - Modem - HylaFAX I will warn you now, however, that the analog-to-digital and then digital-to-analog conversions that are required in these arrangements seem to cause some problems. You may never notice the problems if you use ECM on the receiving modem, but if you pay attention to the ECM logging, you may notice that you'll get more data corruption (and thus retransmissions of data) than you would if you just had the modem plugged into the POTS line directly. I don't know if that's an error on the part of Asterisk, or on the part of all of the ATAs that I've heard mentioned used in this situation. In fact, I've even heard of that same problem when using a TDM card instead of the ATA. Maybe it's just an inherent problem with the A-D D-A conversions. I don't know. In any case, currently the best way (the way without any data corruption as I mention) to interface Asterisk and HylaFAX is to keep everything digital... TE405P - Asterisk - TE405P - T1 Modem (Patton 2977) - HylaFAX But this requires more expensive hardware and more expensive lines and is probably beyond the scope of your project since you're talking about faxdetect and X100Ps. There are some futuristic arrangements that could be done conceivably with some tools that are available, such as spandsp or t38modem, but currently there is no way to interface t38modem with Asterisk and no way to interface spandsp directly with HylaFAX. Lee. ___ 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] OT: DSL without voice
Ed Robbins wrote: So given all that, I'm looking for ideas and solutions that others have implemented to address this issue. There are at least two solutions available: 1.) Locate the emergency number for your local 911 provider - every single 911 office should have a non-911 number that is still considered an emergency number. How do you find this, simple, call your local phone company and ask them, they should know it - otherwise the local police department will. Then have * dial this number over your VoIP line whenever someone dials 911 on one of your phones and there you go, 911 service over VoIP. 2.) Locate the emergency number for your local police, fire department, and hospital and have * present an IVR asking the person that dialed 911 exactly which type of emergency they have. If they don't respond to this IVR, simply connect them to the police, as all three emergency services can and do field calls for the other services. For those of you really thinking this through both the options have one MAJOR flaw. Neither of these can provide the LOCATION of the original call. The simplest solution to this problem is simply post a printed copy of your address ON every single phone in your house. This way when the emergency operator answers the line, the person who dialed 911 will be able to tell the operator where they are calling from. It's not pretty, but it works. Oh, and I think all of this is already discussed on the wiki. -Chris Disclaimer: NONE of this has been tested, but there is no reason why it should not work. If anybody else has other ideas, please state them. -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 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
Re: [Asterisk-Users] Paris Meeting MAP
Local map for the Vendanges Restaurant: http://s91782239.onlinehome.us/asteriskresto.gif ___ 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] SNIMTA_SPAM Using the Directory Feature to play a menu
All my user accounts (mailboxes) have been setup. When an outside person calls my asterisk server they can press 7 to hear a directory of extensions starting at 601 and continues. When prompted to enter the first three letters of the last name comes, I enter the letters. When entered the voice for the mailbox is played I am then asked to verify by pressing one (1). Once I do, asterisk fails to dial the extension. And re-reads the options again never getting out of the loop. You can see my extensions.conf file as well as my voice mail at: http://pastebin.ca/3059 I first started by installing asterisk with the fedora core 2 RPM's released by a member of the asterisk community. I was told possibly that those were no good. I then un-installed the RPM rebooted cleaned up and trails, and reinstalled using the latest release of the stable build. This is happening to both installs. I feel it is something I am missing in my conf files but can not pin it down. Thanks for the help. ___ 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] Problems with app_realtime
THIS IS NOT A BUG you can't use qualify with a realtime peer as they don't live long enough in memory to bee poked/qualified in the do_monitor thread. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Friday, December 17, 2004 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems with app_realtime Post this as a bug Brian. -Matthew - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk-users [EMAIL PROTECTED] Sent: Tuesday, December 14, 2004 3:51 AM Subject: [Asterisk-Users] Problems with app_realtime It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql describe sip_buddies; ++---+--+-+-+--- -+ | Field | Type | Null | Key | Default | Extra | ++---+--+-+-+--- -+ | uniqueid | int(11) | | PRI | NULL| auto_increment | | name | varchar(30) | | UNI | | | | accountcode| varchar(30) | YES | | NULL| | | amaflags | char(1) | YES | | NULL| | | callgroup | varchar(30) | YES | | NULL| | | callerid | varchar(50) | YES | | NULL| | | canreinvite| char(1) | YES | | NULL| | | context| varchar(30) | YES | | NULL| | | defaultip | varchar(15) | YES | | NULL| | | dtmfmode | varchar(7)| YES | | NULL| | | fromuser | varchar(50) | YES | | NULL| | | fromdomain | varchar(31) | YES | | NULL| | | host | varchar(31) | | | | | | incominglimit | char(2) | YES | | NULL| | | outgoinglimit | char(2) | YES | | NULL| | | insecure | char(1) | YES | | NULL| | | language | char(2) | YES | | NULL| | | mailbox| varchar(50) | YES | | NULL| | | md5secret | varchar(32) | YES | | NULL| | | nat| varchar(5)| YES | | NULL| | | permit | varchar(95) | YES | | NULL| | | deny | varchar(95) | YES | | NULL| | | pickupgroup| varchar(10) | YES | | NULL| | | port | varchar(5)| | | | | | qualify| varchar(4)| YES | | NULL| | | restrictcid| char(1) | YES | | NULL| | | rtptimeout | char(3) | YES | | NULL| | | rtpholdtimeout | char(3) | YES | | NULL| | | secret | varchar(30) | YES | | NULL| | | type | varchar(6)| | | | | | username | varchar(30) | | | | | | allow | varchar(100) | YES | | NULL| | | disallow | varchar(100) | YES | | NULL| | | regseconds | int(11) | | | 0 | | | ipaddr | varchar(15) | | | | | | ts | timestamp(14) | YES | | NULL| | ++---+--+-+-+--- -+ 36 rows in set (0.01 sec) -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with app_realtime
Don't use Qualify. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian Wilkins Sent: Tuesday, December 14, 2004 3:52 AM To: Asterisk-users Subject: [Asterisk-Users] Problems with app_realtime It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql describe sip_buddies; ++---+--+-+-++ | Field | Type | Null | Key | Default | Extra | ++---+--+-+-++ | uniqueid | int(11) | | PRI | NULL| auto_increment | | name | varchar(30) | | UNI | || | accountcode| varchar(30) | YES | | NULL|| | amaflags | char(1) | YES | | NULL|| | callgroup | varchar(30) | YES | | NULL|| | callerid | varchar(50) | YES | | NULL|| | canreinvite| char(1) | YES | | NULL|| | context| varchar(30) | YES | | NULL|| | defaultip | varchar(15) | YES | | NULL|| | dtmfmode | varchar(7)| YES | | NULL|| | fromuser | varchar(50) | YES | | NULL|| | fromdomain | varchar(31) | YES | | NULL|| | host | varchar(31) | | | || | incominglimit | char(2) | YES | | NULL|| | outgoinglimit | char(2) | YES | | NULL|| | insecure | char(1) | YES | | NULL|| | language | char(2) | YES | | NULL|| | mailbox| varchar(50) | YES | | NULL|| | md5secret | varchar(32) | YES | | NULL|| | nat| varchar(5)| YES | | NULL|| | permit | varchar(95) | YES | | NULL|| | deny | varchar(95) | YES | | NULL|| | pickupgroup| varchar(10) | YES | | NULL|| | port | varchar(5)| | | || | qualify| varchar(4)| YES | | NULL|| | restrictcid| char(1) | YES | | NULL|| | rtptimeout | char(3) | YES | | NULL|| | rtpholdtimeout | char(3) | YES | | NULL|| | secret | varchar(30) | YES | | NULL|| | type | varchar(6)| | | || | username | varchar(30) | | | || | allow | varchar(100) | YES | | NULL|| | disallow | varchar(100) | YES | | NULL|| | regseconds | int(11) | | | 0 || | ipaddr | varchar(15) | | | || | ts | timestamp(14) | YES | | NULL|| ++---+--+-+-++ 36 rows in set (0.01 sec) -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.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 ___ 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[2]: [Asterisk-Users] Disabling ! command
Hello Justin, Friday, December 17, 2004, 3:43:12 PM, you wrote: JC you could comment that portion out and rebuild? You are right, I will do like this (well at first I have to understand where the comment has to be put) ... just wondering if maybe we can suggest a new option in the config for the purpose. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ 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] Troubleshooting Asterisk
On Fri, 17 Dec 2004 16:19:40 - Paul Brock [EMAIL PROTECTED] wrote: No problem at all :-) Just the output it makes as you try to call from one Cisco to the other. Stupid question on my part, but how do you specify a level to debug at when issuing the debug command??? Currently I'm running a debug, but I suspect it's at too high/low a level, since I'm not seeing a great deal (i.e. debugging is on, but it's not returning anything!!) Just connect to asterisk with asterisk -rv It should produce some good output. -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and HylaFax
On Friday 17 December 2004 16:22, Lee Howard wrote: On 2004.12.17 05:42 Sergio Serrano wrote: Hi all, again I try configure Hylafax with asterisk. I would like configure Asterisk in the next way: 1)An incoming fax go into through X100P 2)Asterisk detects Fax and redirect fax to Hylafax Is it possible? Yes, but it may not be as pretty as you like, and it may not function as well as you hope. Using faxdetect in your zapata.conf file will get practically all of the faxes coming in to the X100P routed to the fax extension. The trick, then, is how to get HylaFAX at that fax extension. Does there exist any sort of bypass box which could be used in the following arrangement: POTS - X100P - Asterisk - TDM400P(FXS) - Fax machine Hypothetical bypass box also plugs into POTS line and Fax machine, able to switch the X100P, Asterisk and the TDP400P out of the circuit, and just connect POTS to Fax directly on some command from the Asterisk PC. Then Asterisk uses faxdetect to send ringing to the fax machine, waits for call to be answered, and sends (RS232?) command to bypass box, allowing fax machine to take the original incoming call without all the analogue - digital - analogue conversion going on. If such a hypothetical bypass box could also detect remote hangup, and drop itself back out of circuit once the call is complete, everything returns back to normal ready for the next call to come in. Electrically it seems like a very simple solution - a 2-pole 2-way relay with RS232 control and line-voltage detection (for the automatic switchover on hangup), however whether such a thing exists and has appropriate type approvals I have no idea Regards, Antony. -- Atheism is a non-prophet-making organisation. Please reply to the list; please don't CC me. ___ 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] Old posts and the ability to search...
Gents, Just a passing thought... is there any reason why the ability to search the past posts on here isn't switched on? Just wondered, since it makes much more sense to be able to search the old archives if you have a problem, rather than ask the same question again and again... Paul ___ 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] OT: DSL without voice
Ed Robbins wrote: Perhaps I wasn't clear. The process of dialing 911 via * is a given, what I'm concerned about is those situations in which you have a power failure and * isn't available. My contingency is to have a phone that is directly connected to the PSTN and simply turn the ringer off so that it isn't bothersome during normal operation. Given a power outage we could use that phone, but again it becomes an issue of training non-family members. Ed Sorry about that, was following a half dozen threads. Wasn't sure on the exact details of what you were after. As for power failure options, you have several. The wiki lists quite a few good ideas. Depending on the type of phones you use, I think power-fail-switches are the best option, as your phones immediately become POTS connected phones. For IP phones, POE with a central LARGE UPS and/or generator would be good. The generator would have other benefits as well. -Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 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
Re: [Asterisk-Users] Problems with app_realtime
I was going to, but then just took a look at the code in chan_sip.c on line 1044 (realtime_update_peer) and the regseconds is in unix time. So, if converted to human time it translates into the correct readable date and time. So if someone has a regseconds of 1103262922, the registration time is 17-12-2004 05:55:22. On Friday 17 December 2004 03:57 pm, Matthew Boehm wrote: Post this as a bug Brian. -Matthew - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk-users [EMAIL PROTECTED] Sent: Tuesday, December 14, 2004 3:51 AM Subject: [Asterisk-Users] Problems with app_realtime It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql describe sip_buddies; ++---+--+-+-+ + | Field | Type | Null | Key | Default | Extra | | ++---+--+-+-+ + | uniqueid | int(11) | | PRI | NULL| auto_increment | | name | varchar(30) | | UNI | | | | accountcode| varchar(30) | YES | | NULL| | | amaflags | char(1) | YES | | NULL| || callgroup | varchar(30) | YES | | NULL| | | callerid | varchar(50) | YES | | NULL| | | canreinvite| char(1) | YES | | NULL| | | context| varchar(30) | YES | | NULL| || defaultip | varchar(15) | YES | | NULL| | | dtmfmode | varchar(7)| YES | | NULL| | | fromuser | varchar(50) | YES | | NULL| | | fromdomain | varchar(31) | YES | | NULL| || host | varchar(31) | | | | | | incominglimit | char(2) | YES | | NULL| | | outgoinglimit | char(2) | YES | | NULL| | | insecure | char(1) | YES | | NULL| || language | char(2) | YES | | NULL | || mailbox| varchar(50) | YES | | NULL | || md5secret | varchar(32) | YES | | NULL | || nat| varchar(5)| YES | | NULL ||| permit | varchar(95) | YES | | | NULL|| deny | varchar(95) | YES | | | NULL|| pickupgroup| varchar(10) | YES | | | NULL|| port | varchar(5)| | | | || qualify| varchar(4)| YES | || NULL|| restrictcid| char(1) | YES | | | NULL|| rtptimeout | char(3) | YES | | | NULL|| rtpholdtimeout | char(3) | YES | | | NULL|| secret | varchar(30) | | YES | | NULL|| type | varchar(6) | | | | || username | varchar(30) | | | | || allow | | varchar(100) | YES | | NULL|| disallow | | varchar(100) | YES | | NULL|| regseconds | | int(11) | | | 0 || ipaddr | | varchar(15) | | | || ts || timestamp(14) | YES | | NULL|| ++---+--+-+-+ + 36 rows in set (0.01 sec) -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.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 ___ 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 -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne,
