RE: [Asterisk-Users] T1-PRI deployment questions...
I have been on vacation so didn't jump in earlier. Some of what I say here has been gone over earlier in this thread but I will repeat the results as a summery. AMI is not lossy, but it is almost always used in conjunction with a ones density technique called bit7. Bit7 will change bit 7 to a 1 when a word (8 bits) are all zeros. In North America when someone says AMI they really mean AMI with bit7. PRI ISDN will not work on an AMI (with bit7) T1. When validating ones density B8ZS does introduce errors but in such a way that the far end will know that they are errors and remove them. The only reasonable choice for PRI ISDN is B8ZS Robbed bit signaling uses bit 8 of each word of every 6th frame for signaling. This does not introduce errors on the line but it does make those bits unavailable for data. Usually data services assumes that the 8th bit of every work of every frame are not available so that you have 7 of each 8 bit word. That makes 56Kbit/s of 64Kbit/s of bandwidth available. PRI ISDN does not use Robbed bit signaling. Instead it uses the D channel. Slips are caused by a timing problem and have nothing to do with AMI or B8ZS. Framing of either SF (D4) or ESF can be used on a T1 regardless of the line coding. (some mistakenly believe that B8ZS must be used with ESF and AMI must be used with SF) ESF is preferred because more robust performance monitoring is possible while the T1 is in service. Don Pobanz Ps. For a while the group I worked with back at the phone company was known as 'T1s are us' so I should be familiar with this stuff. ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring spans
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, No matter what I configure my spans at (on a E400P) ztcfg -v always shows: SPAN x: D4/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Currently I've configured my spans as "ccs,hdb3,crc4", so shouldn't D4/AMI be showing "ccs/hdb3" instead? - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE+21FX2TEAILET3McRAsNoAJ9NqHzDlsaIe+mWJs1hvjtlKp+KxQCeJc45 mwTqE3/vWva1TlMYJbE3cS0= =O//P -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does anyone know how to get rid of this warning message?
Hi, I searched the archives about this, but didn't find any references. When I make an outbound SIP call, the call completes and everything is fine, but in the Asterisk console, I keep getting a huge stream of warning messages: "WARNING[1200876848]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames" I thought I saw this in a post earlier, but I don't get it in the search. Does anyone know what needs to be set to stop these? Thanks in advance. --- Paul Cheng Mátyás király ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP & Caller ID & outgoing line
Hi, I have same problem with identification of the caller. Callerid in sip.conf or LookupCIDName app in extensions.conf wont work. Also having 2 snom100`s dialing each others. What`s the name of the problem? -Johanna Andy Powell wrote in Sat,03 May 2003: Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => [EMAIL PROTECTED]/1000 register => [EMAIL PROTECTED]/1000 register => [EMAIL PROTECTED]/1000 then each phone has an entry similar to this: [phone1] type=friend ;secret=blah host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=192.168.172.189 mailbox=1000; Mailbox for message waiting indicator context=sip callerid="[EMAIL PROTECTED]" This all works fine, I can dial internally,externally and have voicemail. The problem lies in the identification of the caller. When someone calls me, either internally or externally the caller id on the phone always says that the call is comming from asterisk@ as you can imagine, it's not much help. What I (and i guess others) need is a way to pass the REAL caller information to the phones. Although asterisk IS calling me it's only routing the call. The other problem is what is my outgoing sip number in this situation? since I /will/ want to change which line I am calling from how can I go about that? Should I change my sip.conf so that each of the registered domains appears on a different extention, then register the phone with multiple connections one for each exten and then use the phone to set the outgoing line before dialing? Any help appreciated here... (and a special thanks to all on IRC who have got me going) many thanks Andy[Asterisk-Users] SIP & Caller ID & outgoing line ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN4Linux + Asterisk and Europe
Remember that capi 2.0 doesn't have echo suppressor routines. Only Eicon Diva server cards have on board DSPs, that can be enabled with Eicon custon CAPI commands. (the great * chan_capi already do that). Matteo. Il lun, 2003-06-02 alle 12:39, Michiel Betel ha scritto: > My Fritz paasive PCI hasn't crashed so far and works fine, relatively > low latency so not too much echo. However for professional use, get an > active CAPI card so you can use the CAPI echo supp. routines. > > Michiel > > Oliver Brandt said: > > > On Mon, Jun 02, 2003 at 12:33:06AM +0200, Piotr Adamiak wrote: > >> Hello, > >> > >> Anyone on this group using / implementing * and hardware certified for > >> use in Europe ? I believe that ISDN4Linux cards mostly have telecomm > >> certificates, so using them should be safe on the client side. Are there > >> any major issues / problems associated with using such cards with * ? > >> I am talking about a small / very small office with single - few lines. > > > > I tried ISDN4Linux but I had the problem that high voices were > > recognized as DTMF signal wich ended up in beping through the whole > > call. I belive there is a patch out (maybe eve imcluded in the regular > > asterisk code) but I have not tried it. I'm using chan_capi and since I > > swiched to an AVM B1 it works great. With the AVM Fritz (passive) it's > > suppose to work but it actually cause my whole system to crash every > > once in a while... > > Just buy a B1 or so at ebay and you should be fine. > > CU > > Oliver > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP ATA 186
Yes, the remote ATA is behind a NAT and it´s no forced to regiter SIP, I´ll try. Ariel P. AichinoCisbCorrientes 314 Of. 18 y 19S2000CTP Rosario ArgentinaTel. Fax. +54 341 4484004http://cisb.mine.nuemail: [EMAIL PROTECTED] - Original Message - From: Richard Alexander To: [EMAIL PROTECTED] Sent: Saturday, May 31, 2003 10:26 PM Subject: RE: [Asterisk-Users] HELP ATA 186 Is the remote ATA behind NAT ? Is it set to register SIP ?
Re: [Asterisk-Users] enum.conf file
Hi, Thanks, Dan - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, June 02, 2003 1:02 PM Subject: Re: [Asterisk-Users] enum.conf file > ; ENUM Configuration for resolving phone numbers over DNS > > asterisk/configs/enum.conf.sample > > just copy this to /etc/asterisk/enum.conf > > -- > Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK > #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 > VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 > > This mail is confidential & intended solely for the use of the addressee. > > On Mon, 2 Jun 2003, Dan wrote: > > > Hi all, > > > > I get the following warning when starting Asterisk. > > > > Parsing '/etc/asterisk/: == Parsing '/etc/asterisk/enum.conf': Not > > found (No such file or directory) > > > > This file must exist? > > Which is the role of this file? > > > > > > Thanks, > > Dan > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum.conf file
; ENUM Configuration for resolving phone numbers over DNS asterisk/configs/enum.conf.sample just copy this to /etc/asterisk/enum.conf -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential & intended solely for the use of the addressee. On Mon, 2 Jun 2003, Dan wrote: > Hi all, > > I get the following warning when starting Asterisk. > > Parsing '/etc/asterisk/: == Parsing '/etc/asterisk/enum.conf': Not > found (No such file or directory) > > This file must exist? > Which is the role of this file? > > > Thanks, > Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E400P cable
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'm trying to connect an E400P card to a Siemens Digital ISDN S2M Module. (TMS2). How should I connect these two? Straight or crossed? (receive tip <-> receive tip or receive tip <-> transmit tip) Also, I can't seem to figure out how to configure the channels. Does anyone here have any experience with the Siemens module mentioned above? - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE+2yq02TEAILET3McRAncMAKCElMySQ+OUOgtFd2B56cbyp0gVUQCfSfF6 XdYjR1ZEMs1XewHo+ZZ1Cc4= =qATa -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN4Linux + Asterisk and Europe
My Fritz paasive PCI hasn't crashed so far and works fine, relatively low latency so not too much echo. However for professional use, get an active CAPI card so you can use the CAPI echo supp. routines. Michiel Oliver Brandt said: > On Mon, Jun 02, 2003 at 12:33:06AM +0200, Piotr Adamiak wrote: >> Hello, >> >> Anyone on this group using / implementing * and hardware certified for >> use in Europe ? I believe that ISDN4Linux cards mostly have telecomm >> certificates, so using them should be safe on the client side. Are there >> any major issues / problems associated with using such cards with * ? >> I am talking about a small / very small office with single - few lines. > > I tried ISDN4Linux but I had the problem that high voices were > recognized as DTMF signal wich ended up in beping through the whole > call. I belive there is a patch out (maybe eve imcluded in the regular > asterisk code) but I have not tried it. I'm using chan_capi and since I > swiched to an AVM B1 it works great. With the AVM Fritz (passive) it's > suppose to work but it actually cause my whole system to crash every > once in a while... > Just buy a B1 or so at ebay and you should be fine. > CU > Oliver > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN4Linux + Asterisk and Europe
On Mon, Jun 02, 2003 at 12:33:06AM +0200, Piotr Adamiak wrote: > Hello, > > Anyone on this group using / implementing * and hardware certified for > use in Europe ? I believe that ISDN4Linux cards mostly have telecomm > certificates, so using them should be safe on the client side. Are there > any major issues / problems associated with using such cards with * ? > I am talking about a small / very small office with single - few lines. I tried ISDN4Linux but I had the problem that high voices were recognized as DTMF signal wich ended up in beping through the whole call. I belive there is a patch out (maybe eve imcluded in the regular asterisk code) but I have not tried it. I'm using chan_capi and since I swiched to an AVM B1 it works great. With the AVM Fritz (passive) it's suppose to work but it actually cause my whole system to crash every once in a while... Just buy a B1 or so at ebay and you should be fine. CU Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] enum.conf file
Hi all, I get the following warning when starting Asterisk. Parsing '/etc/asterisk/: == Parsing '/etc/asterisk/enum.conf': Not found (No such file or directory) This file must exist? Which is the role of this file? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nagios plugin to check asterisk
> what's the best (read most suitable) network monitoring > tool that's suitable for asterisk (things like > remote process check, remote stats and remote restart) ? We use nagios. I happen to like nagios, since you can make all sorts of plugins yourself. it's just perl scripts (or basic, pascal, C, whatever programs) being executed at intervals. If something comes up (server crashes etc) you can fire off event handlers to restart stuff, you can tell asterisk to call someone and yell at them etc etc etc. > That's because we're going to deploy some asterisk servers > and want to monitor them from remote, including the > poss. to restart it from the same tool if something > is wrong (like Jeremy suggests). I'd split the two into one reporting / surveillence system and one manager (for manually restarting asterisk or similar). Or what do you think? > I'm currently using zabbix (good but little plugin support). See above for asterisk and plugins. Asterisk is generally a framework as compared to what I can see about zabbix. The bad thing about asterisk, is that changing the system's outfit isn't done in 10 minutes ... roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 problems
Makerere University wrote: i am trying to make calls between two workstations using netmeeting and asterisk. i get the popup on both when i call the extensions 665 and 667 but when accept, i get this error *CLI> 0:18.190H225 Caller:8112978 H225Received connect PDU. 0:18.288 H245:810b388 H245Read error: Bad file descriptor 0:18.318 H323 Cleaner H323Connection ip$localhost/25430 terminated. WARNING[23576]: File pbx.c, Line 1702 (ast_pbx_run): Timeout, but no rule 't' in context 'voip-h323' 0:28.313 H225 Answer:80fc988 H225Read error (4): Interrupted system call 0:28.349 H323 Cleaner H323Connection ip$172.18.1.184:2127/5227 terminated. Can't figure out what might be wrong here. Have you specified common codecs on both ends (Asterisk+Netmeeting)? Michael. This is what i have in oh323.conf context=voip-h323 [register] alias=asterisk alias=123 alias=665 alias=667 and in extensions.conf exten => 665,1,Dial(OH323/172.18.1.133) exten => 667,1,Dial(OH323/172.18.1.184) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What they are used for?
Hi, I see the following channels when Asterisk is started: chan_modem_bestdata.so; BestData (Conexant V.90 Chipset) VoiceModem Driver chan_modem.so ; Generic Voice Modem Driver What they are used for? It means that we can use a voice modem as a FX0 interface? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN4Linux + Asterisk and Europe
There has been a lot of discussion about ISDN BRI on the list - a search will turn up plenty of discussion! You're right about there being a lot of ISDN cards available that are certified for use in Europe. They fall into two categories - active and passive. Passive cards are cheap and generally operate through ISDN4linux and asterisk's chan_modemi4l. The reported disadvantages of this approach are: - delay during calls - echo (it's disputed what the cause of it is but it's a bit of a nuisance) - call tones don't follow PSTN patterns. The active cards (AVM is the major supplier, I think) are better. If you get one with a CAPI interface then you can use the asterisk chan_capi driver. I haven't any experience of this type of card - maybe someone else can provide feedback. Iain --On Monday, June 2, 2003 12:33 am +0200 Piotr Adamiak <[EMAIL PROTECTED]> wrote: Hello, Anyone on this group using / implementing * and hardware certified for use in Europe ? I believe that ISDN4Linux cards mostly have telecomm certificates, so using them should be safe on the client side. Are there any major issues / problems associated with using such cards with * ? I am talking about a small / very small office with single - few lines. All the best, Piotr Adamiak -- I am updating my DNA. 'Microsoft Genome 2.0' says I need to reboot my body to continue. I am worried that I may not have enough free chromosomes to allow the full installation to complete. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer Problem
U get the following output when u execute the "show application Dial" command in the Asterisk prompt, -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):Requests one or more channels and places specified outgoing calls on them.As soon as a channel answers, the Dial app will answer the originatingchannel (if it needs to be answered) and will bridge a call with the channelwhich first answered. All other calls placed by the Dial app will be hunp upf a timeout is not specified, the Dial application will wait indefinitelyuntil either one of the called channels answers, the user hangs up, or allchannels return busy or error. In general, the dialler will return 0 if itwas unable to place the call, or the timeout expired. However, if allchannels were busy, and there exists an extension with priority n+101 (wheren is the priority of the dialler instance), then it will be the nextexecuted extension (this allows you to setup different behavior on busy fromno-answer). This application returns -1 if the originating channel hangs up, or if thecall is bridged and either of the parties in the bridge terminate the call.The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then pickedup by another user. The optionnal URL will be sent to the called party if the channel supportsit. Surajee - Original Message - From: George Lin To: [EMAIL PROTECTED] Sent: Monday, June 02, 2003 1:11 PM Subject: FW: [Asterisk-Users] Call Transfer Problem Hi, Which document describes the Dial with “T” option ? Could you let me know or email it to me. Thanks, George Lin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Surajee RatnayakeSent: Sunday, June 01, 2003 9:10 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Call Transfer Problem hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee
[Asterisk-Users] asterisk with vocal
hi all have anyone configured asterisk with vocal and get it to work if yes he can send me steps of configuration on both vocal and asterisk any help is appreciated ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Modem + Soundcard Driver
The problem with using Voice Modems is that they fall into two categories: 1) Hardware Modems which only have half-duplex transmission of voice 2) Soft/Win/Lin modems which are proprietry and don't have asterisk drivers Please shoot down this recipe before I waste any time trying to acheive it: Rationalisation: In Australia and I assume other places, there is no asterisk-compatible low-end hardware which is legal to use. This could be a way to use existing certified hardware with asterisk. Ingredients: 2 Phone lines 2 Banksia Wave SP 336 Modems - these have 3.5mm female jacks for speaker/microphone 1 Full Duplex Soundcard 4 3.5mm mono male-male audio cables 2 3.5mm stereo male -> 3.5mm female left + 3.5mm female right adapters 1 Linux Box with 2 serial ports and spare PCI slot for Full Duplex sound card. Asterisk Programmer Method: --- Plug modems into phone lines and serial ports as per normal Plug adapters into lineout and linein of sound card Plug modem 1 speaker into left of soundcard linein adapter Plug modem 2 speaker into right of soundcard linein adapter Plug modem 1 mic into left of soundcard lineout adapter Plug modem 2 mic into right of soundcard lineout adapter Adapt soundcard driver and/or other drivers to: initialise modems to use Caller-ID via the serial lines expect RING and caller_id info from the serial lines send AT commands to answer calls, end calls, originate calls get voice data for lines 1&2 from left & right channels of soundcard line in send voice data for lines 1&2 to left & right channels of soundcard line out Diagram: POTS-Line 2 | | Line 1 | | | Line Line - -- |=mic-\ /--mic=| | Modem1 || | | Modem2 | |=spkr--\ /-|---|-spkr=| | | | | | | | | L---R L---R | | | | \ / \ / \ / \ / | | |---Line In---Line Out---| || | Sound Card | || Am I out of my tree? Comments, flames? cheers, -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zapata 3.3v PCI version...
I second the question and request. There are more and more server class machines that won't take the old PCI cards at all. -Original Message- From: Gene Kochanowsky [mailto:[EMAIL PROTECTED] Sent: Sunday, June 01, 2003 6:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zapata 3.3v PCI version... Does anyone know if there are any plans for Zapata or anyone else for that matter to come out with a 3.3v PCI version or PCI-X version of those cards? Gene ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer Problem
hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee
RE: [Asterisk-Users] Channel Banks
Hello all, Can someone point me where I can buy a E1 channel bank ( incluidng model and vendors ) which is compatible with digium E400P card. Thanks, George Lin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, April 10, 2003 9:20 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Banks On Thu, 2003-04-10 at 10:37, Jon Pounder wrote: > Anyone on the list actually tried one of those fxs/fxo converter boxes with > a channel bank ? > If so how was it ? I seem to remember someone recently offering up free british toll free access via an ata186 and a box either the same one mentioned here or similar on the FWD list. If you browse the archives there you should see what people thought of the experience, and/or contact the person to see what type of longer term satisfaction they had. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax support over MGCP, H323 and SIP in asterisk
On Sun, 2003-06-01 at 18:32, George Lin wrote: > Hello, > > Can someone tell us if current asterisk supports FAX over MGCP, SIP and H323 Asterisk doesn't care what audio you pass over those protocols. Your problems will be due to packetization and transport adding delay to the audio stream. You also will not be able to use any compression for this type of call. So you take your risks and possibly will have to deal with aborted faxes due to modem disconnections. If you can, put the modem at the phone line termination and use something like hylafax to move the images over IP. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax support over MGCP, H323 and SIP in asterisk
Hello, Can someone tell us if current asterisk supports FAX over MGCP, SIP and H323 ?? Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata 3.3v PCI version...
Does anyone know if there are any plans for Zapata or anyone else for that matter to come out with a 3.3v PCI version or PCI-X version of those cards? Gene ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN4Linux + Asterisk and Europe
Hello, Anyone on this group using / implementing * and hardware certified for use in Europe ? I believe that ISDN4Linux cards mostly have telecomm certificates, so using them should be safe on the client side. Are there any major issues / problems associated with using such cards with * ? I am talking about a small / very small office with single - few lines. All the best, Piotr Adamiak -- I am updating my DNA. 'Microsoft Genome 2.0' says I need to reboot my body to continue. I am worried that I may not have enough free chromosomes to allow the full installation to complete. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forcing intermachine codecs ?
Jeremy McNamara wrote:> A working example for restricting codec's by peer
[general]
port=5036
bind=0.0.0.0
allow=all
tos=lowdelay
[NuFone]
type=peer
disallow=all
allow=iLBC
trunk=yes
host=switch-1.nufone.net
context=NANPA
I have this iax.conf setup on many of the customer boxes I babysit. This
way they can use any codec they want for other user's or peer's, but
force everything to iLBC when calling out to NuFone. This works the
same with SIP, but not H.323. I haven't personally tested MGCP, since it
uses SDP I would guess it should work also.
I added those entries to my contexts for NuFone, and everything works
just fine.
So I added them to the rest of my iax.conf contexts, most of which move
calls around between my three different asterisk instances.
But for whatever reason, my IAX2 calls to everywhere else all go with
"format=4"
I'm not sure which side is dominant. . .Here is a typical entry from the
CLI on "my side:"
-- Called foo:[EMAIL PROTECTED]/6060
-- Call accepted by 1.2.3.4 (format 4)
-- Format for call is 4
And on the other:
-- Accepting AUTHENTICATED call from 5.6.7.8, requested format = 4,
actual format = 4
-- Executing Dial("[EMAIL PROTECTED]:4569]/1", "Zap/2/BYEXTENSION")
in new stack
I have put the disallow/allow clauses in every context that is used by
my servers, and I've checked that all the asterisk instances have the
same version and the ilbc modules.
What am I doing wrong?
Thx.
B.
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Re: [Asterisk-Users] Any plans for a .....
On Sun, 2003-06-01 at 15:49, Gene Kochanowsky wrote: > I saw that Digium now has a 4 port FXS card. Any plans to add a 4 port > FXO card? The 4 port cards are actually modular. It is a backplane and daughtercards to hook the phone jacks to the system. There has been an anouncment that they plan on releasing the FXO daughterboards. I believe the last I heard was they are going through the approval process. I think the plan is for it to be mix and match when the FSO cards are available. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any plans for a .....
I saw that Digium now has a 4 port FXS card. Any plans to add a 4 port FXO card? Gene ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
