[asterisk-users] Asterisk or Trixbox
Hi, In my case I used them both. It depends on the features I am going to activate. There are some departments that use only a standard PABX-like, use 8 Ports FXO Gateway, 48 Ports FXS Gateway and all they need is to be able to call and be called. I used Trixbox for this application. I have some departments that use ISDN Trunks that uses E1 Gateway, Predictive/Pre-emptive dialing I use only plain Asterisk. In short, I decided on what to use base on what will be my applications. If my application needs some modification/patches to be done during the installation of Asterisk or Zaptel, I use plain Asterisk. But if, its straight-forward PABX application with IVRS, call distribution, voicemail, conferencing, call queue, fax to email and voicemail to email, I use Trixbox and I assure you it's easy to set-up (it's really freePBX in 1 to 2 hrs.) - Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: how to define a pilot number
Hi Lito, It depends on how you asked your telco provider to configure your 3 direct lines. We called it trunking the 3 lines with pilot number. Telco can figure it the way we configure our Asterisk Followme. (seized all, random, sequencial). Regards, Angel Lito Lampitoc [EMAIL PROTECTED] wrote: thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? On 3/27/07, David Cook [EMAIL PROTECTED] wrote: is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Lito I'm assuming you are talking about analog lines as PRI's will do this more-or-less naturally. This is a telco feature as opposed to an Asterisk feature. Here in Bell Canada country they call it Ringer Equivalence. Call your local carrier and they should be able to tell you what they call it in their marketing world. You tell the telco which lines you want the calls to roll to then all three will terminate calls to the pilot number. Now it doesn't work exactly as you had described - it doesn't move the call so as to free up the first port. It merely says the first port is busy and terminates the next call on the next port in sequence. This means you can't count on which line is available at any time. For outbound, you need to put the three lines in an Asterisk group and test the group for availability to select an available line to dial out on. dbc. -- David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
Hi Hind, Sorry, I haven't work for it yet. I still left on the number of endpoints assigned. Probably I will concentrate on it once I finish with my gnuDialer Project. I'll keep you informed. Angel hind habaoui [EMAIL PROTECTED] wrote: hi angel. it is about the CallerId, i have the same problem, did you resolve it??? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
Hi Yehavi, Yes, this can be done. We are currently using this features. The Secretaries making the calls to who ever her Boss wants to join the conference she then just transfer the calls into the conference room. You can even annouce the name of the newly arrived calls in the conference. Like; Mr. Mateevitsi join the conference or Mr. Mateevitsi leaved the conference if one's leave the conference. I had created one coference room for every department. Regards. Angel Victor Mateevitsi [EMAIL PROTECTED] wrote: Or, you can just transfer the calls into the conference room. On 3/19/07, Lee Jenkins [EMAIL PROTECTED] wrote: Yehavi Bourvine +972-8-9489444 wrote: Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Don't get soaked. Take a quick peek at the forecast with theYahoo! Search weather shortcut.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with MFCR2 and Meridian
Hi Artur, Just follow the information Moises had recommended you and for sure this will work. The default configurations that was exampled in the document is just fined and suited with Nortel Meridian. Just be sure that your Nortel MFC Card is installed and working in good condition with up to date Nortel patches. Regards, Angel Moises Silva [EMAIL PROTECTED] wrote: Arturo, the error does not says much really, just that either the other end timed out expecting a response from you, or your end timed out expecting a response from the other end :) However, from my experience, it may be an error in your DNIS/ANI configuration and/or an mfcr2 library error ( less likely but still possible ). Anyway, you can lear how to debug this problems with this little document I wrote: http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Good Look and happy debugging! :) - Moisés Silva - It's here! Your new message! Get new email alerts with the free Yahoo! Toolbar.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?
Hi Giorgio, I guess it will be more benefitial to all old version users to read some information regarding new version like * 1.4. In this way, they will be encourage, or probably have an idea whether to upgrade or not based on all the concerns that was posted. I for one still using 1.2.13 but I love reading * 1.4 concerns before leaping to 1.4. Regards, Angel dave cantera [EMAIL PROTECTED] wrote:here! here! they are different beasts... Giorgio Incantalupo wrote: Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Don't be flakey. Get Yahoo! Mail for Mobile and always stay connected to friends.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
Hi Steven, Thank you for your response. I had tried leaving endpoint phone number blank but when I tried making an outside call, the Audiocodes seems doen't know where to pass the call. So I need to assigned numbers. There is no problem for incoming call aside from not being displayed the Caller ID. Steven Totaro [EMAIL PROTECTED] wrote: I have no experience with AudioCodes but it seems that you need to have callerID enabled, leave endpoint phone number blank. Hope this helps. Maybe some of this info might help: http://www.voip-info.org/wiki/view/AudioCodes ++ If caller ID is turn on, then freePBX will only record the receiving number.not the line number. ++ Well, you can fix this, by using the Routing General settings, Audiocodes allows you to Prepend the Hunt Group to the number, You can then use the Manipulation tables, and strip the source number (tel--IP) after Routing. So if u set each Endpoint up to have a different Hunt Group, you can get it to ID the line. They also have a x-channel header that can be added for you to look in the SIP message at. things that help when dealing with the FXO's They are designed to work with Analog PSTN lines, 1. Caller ID is usually delivered between the 1st and 2nd ring on these lines. Also make sure it is enabled in the Supplementary services. 2. For those of you expecting the number to get delivered through to the IP side when dialing, it won't PBX's and CO's just ring the PSTN line, they don't deliver the number. Make sure you Enable AutoMatic Dialing in the Endpoint Settings, and if you want the line in port x to be the number dialed to the sip side, datafill the number there. 3. Make sure you set up the audiocodes with the proper coder like Ulaw, they come set to 723 by default which is crap for coders. they can support up to 5 so just datafill them with all the big coders U, A, 729, and whatever else 4. The Advanced Configuration pages have all their Channel settings, make sure the fax's are set to what the Trixbox supports. Audiocodes by default does t.38 now. if your pbx isn't set up for it, you need to put the Audiocodes in a transparent or events mode If you want the source number from IP to use the same datafilled Endpoint Port on the PSTN side make sure Endpoint Phone Numbers has that number datafilled, and then set up a hunt group with source number as the selection algorithm(5.0). Assign the endpoints to that hunt group. IP to Tel rouitng route all calls to that group Endpoint Phone Number - This will give you the options for either 4 or 8 ports. You do not need to place anything here. However, it is a good idea to do such to help you identify which port the call comes in on; as you can view the reports in freePBX to identify calls. In my case, since I have four PSTN ports, I used the last four digits of the telephone number to identify. Identifying which PSTN line the call came from only works if you DO NOT have caller id on the line, or your turn off caller id. If caller ID is turn on, then freePBX will only record the receiving number.not the line number. Endpoint Settings - Automatic Dialing - Define a station number located on Asterisk / Trixbox (ie 101) for all ports - Caller ID - Allowed .. turn off if you want to Identify the line they came in on. - Detect Caller ID from Tel - Enable Thanks, Steve Totaro From: Angel Heart Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO Date: Mon, 26 Feb 2007 19:22:53 -0800 (PST) Hi José, I have not resolve this issue yet. I am currently focusing in my newly arrived toy (fonebridge2) then after which I will go back to AudioCodes Issue. Still I don't received yet any response from AudioCodes Representative here in the Philippines. I had already escalated this to their Regional Office in Singapore. But still no reply for almost a month already. I will post immediately once I resolve the issue. It is important to us because we really need to now where the calls coming from. Regards Angel. José Luis Gómez wrote: Hello Angel. Did you solve this issue? I have the same problem. Thanks, José El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió: Hi, I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming outgoing calls. However, I noticed that the caller ID of the caller coming from the FXO displays its endpoints assigned number and not the actual caller's ID coming from PSTN. Hope someone is using the same scenario and could share on how to resolve the caller ID/Number. Thanks. Angel
Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
Hi José, I have not resolve this issue yet. I am currently focusing in my newly arrived toy (fonebridge2) then after which I will go back to AudioCodes Issue. Still I don't received yet any response from AudioCodes Representative here in the Philippines. I had already escalated this to their Regional Office in Singapore. But still no reply for almost a month already. I will post immediately once I resolve the issue. It is important to us because we really need to now where the calls coming from. Regards Angel. José Luis Gómez [EMAIL PROTECTED] wrote: Hello Angel. Did you solve this issue? I have the same problem. Thanks, José El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió: Hi, I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming outgoing calls. However, I noticed that the caller ID of the caller coming from the FXO displays its endpoints assigned number and not the actual caller's ID coming from PSTN. Hope someone is using the same scenario and could share on how to resolve the caller ID/Number. Thanks. Angel __ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk callerID
Hi Allen All, I had posted this kind of problem 2 weeks ago but seems nobody from here encountered yet. So I haven't received any reaction as of the moment. The problem with AudioCodes' FXO is that I cannot make it work without defining endpoints number. Once a number is defined, this number will serve as the callerID or will be displayed from a call coming from the FXO/PSTN. I guess same thing with FXO cards installed directly to an Asterisk Server. I have not find the solution yet until this time. Hope somebody from AudioCodes could share solutions on this matter. Allen Casteran [EMAIL PROTECTED] wrote: voip crazy wrote: Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I make a call directly using one of this line, the callerID is sending correctly. With the same zapata config file and the Freepbx 2.1.3, the callerId was sending correctly. Any clue will be welcome Again, your description is not clear. If your problem is that calls coming in to asterisk are not displaying caller ID on your phones, then you need to make sure that your CO lines are configured by the carrier to deliver caller ID. As a test connect a basic analog phone that has caller ID capability and call the line. If your simple phone displays the CallerID that you are calling from your line supports it and Asterisk should pick it up. If you do not see the caller ID on the analog phone when directly connected to the CO line, then call your carrier and ask them to provide Caller ID on your lines. I had this exact situation this morning, so yes it happens. If your problem is calling OUT from asterisk and your caller ID not getting displayed on the phone you are calling, that is also a function of the carrier and something you have NO control over. You should see something on the far end even if its Private unavailable or blocked. Call your carriers and ask them to check their set up for your phone lines. Allen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues do not accept calls if all agent are busy?
Hi, cud any one help me figuring out the problem... When the agent in a queue is engaged, it cannot accept anymore calls, below is the script; -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/2063||tr) in new stack -- Called 2063 -- Local/[EMAIL PROTECTED],1 is ringing -- Got SIP response 486 Busy Here back from 10.19.1.158 -- SIP/2063-084a6c18 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing NoOp(Local/[EMAIL PROTECTED],2, Returned to dolocaldial with DIALSTATUS BUSY) in new stack -- Executing Macro(Local/[EMAIL PROTECTED],2, outisbusy|) in new stack -- Executing Playback(Local/[EMAIL PROTECTED],2, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Local/[EMAIL PROTECTED],1 answered SIP/10.19.1.157-084eec28 -- Stopped music on hold on SIP/10.19.1.157-084eec28 -- Executing Playback(Local/[EMAIL PROTECTED],2, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') -- Executing Macro(Local/[EMAIL PROTECTED],2, hangupcall) in new stack Thanks Angel - Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues do not accept calls if all agent are busy?
Hi Ex vito, Thank you for your response below is my current config. I defined it into my queues_addintional.conf where the definitions of queues defined. Do I need to defined it in the general portion of queues.conf? But anyway, there's no harm for trying. I am using Asterisk 1.2.13 svn rev 47264 with AudioCodes FXO/FXS Gateway. queues.conf [general] ; ; Global settings for call queues ; (none exist currently) ; ; Note that a timeout to fail out of a queue may be passed as part of application call ; from extensions.conf: ; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) ; example: Queue(dave|t|||45) [default] ; ; Default settings for queues (currently unused) ; #include queues_custom.conf #include queues_additional.conf queues_additional.conf [7001] wrapuptime=5 timeout=20 strategy=leastrecent retry=5 queue-youarenext= queue-thereare= queue-thankyou=custom/client-in-queue queue-callswaiting= music=default monitor-join=yes monitor-format= maxlen=0 leavewhenempty=no joinempty=Yes context=ivr-6 announce-holdtime=yes announce-frequency=30 Kindest regards. Angel Ex Vitorino [EMAIL PROTECTED] wrote: On 2/15/07, Angel Heart wrote: cud any one help me figuring out the problem... When the agent in a queue is engaged, it cannot accept anymore calls, below is the script; Angel, Check your queues.conf, specifically the joinempty parameter. See below the relevant part in the queues.conf sample file: ... ; This setting controls whether callers can join a queue with no members. There ; are three choices: ; ; yes- callers can join a queue with no members or only unavailable members ; no - callers cannot join a queue with no members ; strict - callers cannot join a queue with no members or only unavailable ; members ; ; joinempty = yes ... Cheers, -- Ex Vito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bandwidth shapping device
Hi, What Network Switch you are using? I do traffic/bandwidth shapping on the edge switch where the port the voice installed, you can configure each port to 128Kbps or just plain Ethernet Port. So the link between bldg. will always be 10Mb/s, who ever uses it whether data or voice and enable switch port prioritization. HP ProCurve and Cisco switches do this features. Don't know if others can do it. Regards Angel Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 14 Feb 2007, Damon Estep wrote: Why do that? Just traffic shape each user/group of IP addresses to the total bandwidth you want them to have and then set up a low latency queue for voip traffic, that way the voip bandwidth can be used for data when there are no calls but will give VoIP traffic priority over other traffic. Any old refurbished Cisco 2611 or 2621 will do the trick. Look up low latency queuing and traffic shaping on cisco.com If you are doing NAT on the router I recommend a general deployment (GD) 12.3 IP feature set IOS image. I have to say, that unless you are quite good at driving Linux or *BSD's firewall/traffic shaping mechanisms, then I'd probably go for a Cisco - especially if this is a full-on corp-rat environment. I would use a Linux box, but then I've been using Linux boxes for a great number of years including setting up some hairy/scarey traffic management. Gordon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Wednesday, February 14, 2007 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bandwidth shapping device I'd use a MikroTik or 2 - Original Message - From: Ronald Wiplinger To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, February 14, 2007 2:19 PM Subject: [asterisk-users] Bandwidth shapping device I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to resolve CallerID from AudioCodes FXO
Hi, I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming outgoing calls. However, I noticed that the caller ID of the caller coming from the FXO displays its endpoints assigned number and not the actual caller's ID coming from PSTN. Hope someone is using the same scenario and could share on how to resolve the caller ID/Number. Thanks. Angel - Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Hi, I'm not sure, but I experienced it before with our Nortel Meridian I MFC/R2. When set to both zero(0), calls drop once answered. I tried to vary its values until I finally got it stabled. I'd been in the Datacoms/Telecoms for 16 years now, only with Asterisk I experienced beyond technical theory (out of the book). But bottom line is, it works. Magic ! Angel. Facundo Ameal [EMAIL PROTECTED] wrote: Thanks for the response, I 've already matched codecs. I have no problems with that. Do rxgain and txgain have something to do with R2 protocol errors? Regards. On 1/28/07, Angel Heart wrote: Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal [EMAIL PROTECTED] wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and
Re: [asterisk-users] Any quiet 24 port POE switches out there?
Hi, I am using these model from HP ProCurve http://www.hp.com/rnd/products/switches/switch2600series/features.htm?jumpid=reg_R1002_USEN http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/features.htm?jumpid=reg_R1002_USEN Regards, Angel Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den 03.01.2007, 16:51 -0600 schrieb John French: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. I had to browse through the list of switches on the market recently for different features. Most switches do not feature an acoustic entry in their description. Even those described as desktop devices... and just with it being named a desktop device does not necessarily give you a silent device, au contraire. All I found was the Nortel BAS220 48T (with 24 out of 48 ports PoE compliant), nominal 43.8 dB on the datasheet. I do not know that device, but noise information on PoE switches seems not to be a thing that manufacturers are proud of. I guess building a 1u-switch with an included 300W++ power adaptor requires active cooling, and the smaller the fans, the noisier the whirl. Maybe using several, smaller switches could do the trick for you. Brian Roy mentioned the Netgear FS108p (with external power adaptor, noiseless) as 8-port device. There is also a larger brother of it, the FS116P, which also comes with an external power supply, does PoE on eight of its 16 ports. I have no idea of your overall bandwidth requirements, but if it is only about phones, 100 MBit should be by far sufficient for those 20 devices, so you could cascade switches (like plugging two FS108p into non-PoE ports on a single FS116P, for instance). This is of course the cheapo way of doing it. Getting a proper multi-port switch, perhaps even a real brand one would be (ask the drooling sales droids out there) would be the real deal. Talking about NetGear switches, I once bought a 24port Gigabit Netgear switch, noiseless, external PSU. It was meant to be screwed to a table from below (in a classroom environment) with four metal brackets. The switch kept crashing (not letting any data through) in that environment, situation only changed when mounting that switch to a wall (with the CAT6 cables hanging straight down from the plugs) - temperature problem (which was not bad enough to go into warranty exchange. Just do not use the switch in a hot environment. 20°C in a boring computer lab) On a non-PoE device, with far less than 300W power to go through. I personally do not trust wall-wart (a.k.a. external power supply) switches too much. I do not think it is a problem in principle, but those devices with internal power supply just tend to be better for me. YMMV. If you find something worthy, with a decent sound, please report back to the list so others can share a good experience. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 and unicall
I don't think if somebody making upgrades for the unicall in accordance to the latest version of Asterisk. The latest patches of unicall and MFCR2 that I saw is still for Asterisk ver. 1.2.0. Haven't see any patches for latest version yet. This what making me afraid of going to upgrade our Asterisk, I am using MFCR2 as well with Asterisk 1.2.12 without any problem. I hope there will be version of upgrades that it won't delete unicall libraries and its dependencies. Rgds. Angel Anton Krall [EMAIL PROTECTED] wrote: I hope so, he is the only guy working on mfcr2 right now. I have unicall working on 1.2 perfectly but if there will be no unicall support for 1.4, that would be a show stopper unless we use a mfcr2 converter... anybody knows any? Something that can convert mfcr2 to pri? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Barzilai Spinak |Sent: Thursday, December 28, 2006 8:26 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] 1.4 and unicall | |I asked the same a while ago, without any kind of conclusive answer. |But you have to consider that these are special dates |I just spent all night studying/modifying mfcr2.c to my needs but |I've never looked at the unicall code or the asterisk channel API. |With respect to MFC/R2, and according to what it saw, it seems fairly |complete on the incoming part of the protocol, but the outgoing logic is |kind of crude. |I wonder if Steve Underwood is still actively working on it. | |BarZ | |Anton Krall wrote: | No update on unicall and 1.4? | | |-Original Message- | |From: [EMAIL PROTECTED] [mailto:asterisk-users- | |[EMAIL PROTECTED] On Behalf Of Anton Krall | |Sent: Tuesday, December 26, 2006 6:15 AM | |To: asterisk-users@lists.digium.com | |Subject: [asterisk-users] 1.4 and unicall | | | |Guys, anybody knows if 1.4 has support for unicall or if/which version of | |unicall will compile on it? | | | | | |___ | |--Bandwidth and Colocation provided by Easynews.com -- | | | |asterisk-users mailing list | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |asterisk-users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users BEGIN:VCARD VERSION:2.1 X-MS-SIGNATURE:YES N;LANGUAGE=en-us:Krall;Anton FN:Anton Krall ORG:Intruder Consulting TITLE:A Division of IntruderEnterprises S.A. de C.V. TEL;WORK;VOICE:+52 (55) 5781-5112 x 201 TEL;WORK;VOICE:+52 (55) 5985-2430 x 201 X-MS-OL-DEFAULT-POSTAL-ADDRESS:0 URL;WORK:http://www.intruder.com.mx EMAIL;PREF;INTERNET:[EMAIL PROTECTED] PHOTO;TYPE=JPEG;ENCODING=BASE64: /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa KCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCj/wAAR CAAXAEgDASIAAhEBAxEB/8QAHwAAAQUBAQEBAQEAAAECAwQFBgcICQoL/8QAtRAA AgEDAwIEAwUFBAQAAAF9AQIDAAQRBRIhMUEGE1FhByJxFDKBkaEII0KxwRVS0fAkM2JyggkK FhcYGRolJicoKSo0NTY3ODk6Q0RFRkdISUpTVFVWV1hZWmNkZWZnaGlqc3R1dnd4eXqDhIWG h4iJipKTlJWWl5iZmqKjpKWmp6ipqrKztLW2t7i5usLDxMXGx8jJytLT1NXW19jZ2uHi4+Tl 5ufo6erx8vP09fb3+Pn6/8QAHwEAAwEBAQEBAQEBAQECAwQFBgcICQoL/8QAtREA AgECBAQDBAcFBAQAAQJ3AAECAxEEBSExBhJBUQdhcRMiMoEIFEKRobHBCSMzUvAVYnLRChYk NOEl8RcYGRomJygpKjU2Nzg5OkNERUZHSElKU1RVVldYWVpjZGVmZ2hpanN0dXZ3eHl6goOE hYaHiImKkpOUlZaXmJmaoqOkpaanqKmqsrO0tba3uLm6wsPExcbHyMnK0tPU1dbX2Nna4uPk 5ebn6Onq8vP09fb3+Pn6/9oADAMBAAIRAxEAPwDI8HeCb3xFHJclhb2MY+adumfQDua62b4Q iBGluNWEcOPlZreQZJ/4DXS/DAXn/CBxS6fArXscc5gbPOflGfrgmun8NNfr4dt5vESs2oG2 lRNxy7ZZcYHryfwzXo4nGVouTi0knY+ew+CoyhHmV21e58/XvhaWz1XWLGZpC+n27TkxoGBw QOckYHPXn6VFdeDdftbV7i4050iRFkbMiblRsYYrnIXkc4xXWeJNRFp408Xosc9217YyW6mF d20llO5vQDFUdQ8TxT6pr2pi0vBaalpqafCzLwHEcanJzjGUbpXoxqVGk/66HnzpUU2n3f6/ 8D7zO1D4f67bao9lbQJdukEdw7xyKFRXAPJJ4wTjJ9M9Kxl0DVHu7a1Wzcz3MJuIUBHzxgMS w56YVj+FdnrevxXMesW8FjqaX2oaba27RPDjyzDsBPXJUhOuO9LpPijRoDo+p3UOp/bNP017 ARxxqYmyroH3k5/j6Y696FUqct2v6t/mEqNFysnZf8H07amJ4Z8CarrU1o0sRtbK4jeVZmZS 2xVJ3CMsGKkjGcY5rNs/CutXmmrfW1i727Kzod6hnVfvMqE7mA55ANdvYa7psOraXq9/b6ol /YW/9lNBEiGFpVRkBD7hjg5K4696bpnjOC30XSrqSC9huNNtjZqYrOFkkYbtpEzAsn3uQAf1 pOpVvov61/4BSo0LJN/1p/wdNzg9Q0DU9P0221C9tWhtLkBoXZl+cEZBAznGO+KKveM9Qa/b Rw1vcQfZtOhtsTLt3Fc5ZfbNFdEG2rs46qjGVo7Gh4Z8e3uhaetpHbW9wiZ8syru2Z64BrRn +KmqPaslva2ttcFCnnxIFYA9en86KKyeGpSlzOKuaRxdaK5VLQ4qHVLiOaeRisrTDD+YM8g5 B+oIpG1KdtPSzITy0xhsfMQCzAfTLMfxoorblRjzPuWItevY74XeY2l2svK8EM5c5APPzMeO
Re: [asterisk-users] Inform callers on recorded/monitored number.
Hi Paul Eric,
Thank you for you information and quick response. I had enabled Monitoring in
every SIP phone already. Made some Playback see below truncated config;
exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten = s,22,Goto(s-${DIALSTATUS},1)
exten = s,108,Noop(max channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()
;Below was an added codes for the purpose of advising caller status of their
call.
exten = s-NOANSWER,1,Playback(user)
exten = s-NOANSWER,n,Playback(is-curntly-unavail)
exten = s-NOANSWER,n,Hangup()
exten = s-ANSWER,1,Background(for-quality-purposes)
exten = s-ANSWER,n,Background(this-call-may-be)
exten = s-ANSWER,n,Background(recorded)
exten = s-CHANUNAVAIL,1,Playback(is-curntly-unavail)
exten = s-CHANUNAVAIL,n,Hangup()
exten = s-CONGESTION,1,PlayTones(congestion)
exten = s-CONGESTION,n,Wait(5)
exten = s-CONGESTION,n,StopPlayTones()
exten = s-CONGESTION,n,Hangup()
All the value of DIALSTATUS are working except if its ANSWER, it not working
neither the caller or callee doen't hear anything. I might inserted the message
at the wrong .conf file. I just thought that somebody out there had tried doing
these before.
Scenerio:
SIP phone (101) wanted to call out-side Asterisk via ISDN/PSTN (6320011). Upon
answering by user 6320011, it hears sound like For Quality Assurance Purposes,
this call might be monitored or recorded. It is more important for us that the
called 6320011 should be informed about the recorded conversation and its up to
him/her (called/6320011) to hangup or accept.
The same thing when some body called the SIP phone (101), from out-side
Asterisk via ISDN/PSTN Trunk. The caller (from PSTN) should be informed about
the recorded calls; Asterisk will send ringing tone then playback(For
Quality...) continue with music(MOH) until SIP phone(101 will answer.
Hope you could provide me a little bit specific configuration on where to
insert such scripts.
Thanks
Angel
- Original Message
From: Eric Jacksch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 19, 2006 1:02:34 PM
Subject: Re: [asterisk-users] Inform callers on recorded/monitored number.
exten = s,1,Answer
exten =
s,n,Set(REC=${URIENCODE(${STRFTIME(,America/Toronto,%Y%m%d-%H%M%S)}-${CALLERID(number)}-TESTBOARD-${UNIQUEID})})
exten = s,n,MixMonitor(${REC}.wav)
exten = s,n,Playback(this-call-may-be-monitored-or-recorded)
Note that I intentionally start the recording BEFORE advising the user that the
call may be monitored — that way the first thing on the recording is the user
being advised of the recording.
-
With the playback command?
I think we are missing something here.
PaulH
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[asterisk-users] Inform callers on recorded/monitored number.
Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller callee that thier line is monitored prior to start conversation. Thanks. Angel __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to temporarily unload modules.
Thanks Tzafrir and Marco for the info. If I want to unload modules during start-up, I have to edit my /etc/asterisk/mudules.conf and add something like; noload = app_test.so or I can unload them immediately at CLI using Mr. Cohen suggestion. Regards. /etc/asterisk/modules.conf Marco - Original Message From: Tzafrir Cohen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 8:54:04 PM Subject: Re: [asterisk-users] How to temporarily unload modules. On Wed, Dec 13, 2006 at 02:03:09AM -0800, Angel Heart wrote: Hi, In what Asterisk file can I edit for me to temporarily unload such modules. But later I woudl still be able to load them. Works fine as long as the module is not in use. asterisk -rx 'unload app_test.so' Later on: asterisk -rx 'load app_test.so' -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to temporarily unload modules.
Hi, In what Asterisk file can I edit for me to temporarily unload such modules. But later I woudl still be able to load them. Thanks Angel Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches
Hi, You may want to visit www.procurve.com and look for thier training section there are lots of training materials that can be downloaded. Prices are also posted in this website. Actually, all networking manufacturers has thier training docs posted in their websites. www.3com.com www.nortel.com www.cisco.com - Original Message From: Zeeshan Zakaria [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 11, 2006 10:53:53 PM Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches What's the price for these HP switches? And also I someone can give me a link to some document where I can read about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have a burning question? Go to www.Answers.yahoo.com and get answers from real people who know.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for QoS, PoE Switches
Hi, I am using Procurve Switches by HP for PoE. http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/overview.htm?jumpid=reg_R1002_USEN Aside from being a LIFETIME WARRANTY, I found them very easy to configure and install. Regards, Angel - Original Message From: Cory Andrews [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 10, 2006 10:35:13 PM Subject: RE: [asterisk-users] Recommendations for QoS, PoE Switches Zeeshan - I really like the Adtran Netvanta and/or Cisco Catalyst 3560 series switches if your customer has the pocket depth for them. You are going to want a good, VLAN capable managed switch. There are cheaper alternatives from Linksys, Netgear and DLink as well. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: Sunday, December 10, 2006 2:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recommendations for QoS, PoE Switches Hi all, For a top quality setup, I will need to install high quality VoIP switches with QoS and PoE. My potential customer should not have any problem with call quality. Experienced folks, Please advice me what switches to install and at what price. I may need it for upto 100 phones. What else should I consider so that phones work without problem along with the computers on the same network? Phones will use their bridged ethernet connections, so that both computer and phones can work on the same connections. Thanks -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 200+ analog phones connected to FXS modules
Hi guys, We are using AudioCodes, but still looking an alternative a cheaper one for our expansion. We are currenly running 4x24-ports FXS VoIP Gateway with 2 Asterisk Server each server has Dual-Port Card interfaced with E1 PRI ISDN to PSTN and E1 MFC/R2 to PABX. Our E1 ISDNs to PSTN are already 30-day working, I am still working for the interconnection to MFC/R2 PABX and its been a month doing some trial and error thing. I was able to compiled those libraries but still getting some protocol error when calls are made. Once am able to make this run we'll push through with the expansion to our branch. Can this TDMoE able to call each other (phones) within a single 24-ports without pass-through Asterisk? Like calling each other within the branch. It will only pass through the Asterisk when a call made to an inter-branch or going to Main Office. Or in other word, can I use TDMoE without Asterisk? Regards. Angel - Original Message From: Zoa [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, December 1, 2006 2:21:37 AM Subject: Re: [asterisk-users] 200+ analog phones connected to FXS modules Interesting product, I didn't know about this one until just now. I've heard that TDMoE is more trouble than it's worth, though, and may eventually be phased out of Asterisk. Can anyone from Digium give some more information or suggestions? -A. I'm not from digium but am the proud owner of a preproduction sample of the spidermux, i also took it to Astricon Dallas. (they are already being produced but are not being sold yet). The TDMoE implementation in asterisk works, but is not used by a lot of people or hardware yet, so it needs some work (Especially to make it work with recent kernels). I know the spidermux people already have a bunch of patches ready to be released to fix the issues that exist now. I've never heard something about tdmoe being phased out of asterisk. Zoa. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Any questions? Get answers on any topic at www.Answers.yahoo.com. Try it now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_unicall.c isntallation problem
I am in the middle of installing chan_unicall.c and channels_Makefile.patch in my channels directory as instructed at the bottom of this doc/site: http://soft-switch.org/unicall/installing-mfcr2.html. I am at lost 'coz I don't know where is my channel directory is 'til someone told me that it is where all my chan_xx.so resides, so I copied them to this directory. Next, I have to issue this command patch channels_Makefile.patch and it gave me a question File to patch: which I don't know what file to patch. There were no other files under my channels directory (/usr/lib/asterisk/modules) except .so files no Makefile make files. Until I found this site: http://zarzamora.com.mx/asterisk/48. In this site, I need to put these two files in /usr/src/asterisk/channels but I do not have this directory created during the installation of Asterisk using Trixbox bootable CD. It is also stated that in the /usr/src/asterisk/channels i have to patch Makefile novo_patch.patch (or whatever patch version I want to use) then make and make install at /usr/src/asterisk directory which i do not have this directory. On what module/libraries or directory should I use the Makefile and issue command make make install? Can somebody share a more detailed instruction on how to install this chan_unical module. Thank and best regards Angel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installation of Unicall for MFC/R2
Hi Moises, 1. On what directory/folder should I copy the chan_unicall.c, channels_makefile.patch?into wherever you had put the source code of your asteriskinstallation. There, you must have a folder named "channels", whereyou will see several files named chan_sip, chan_zap and in generalchan_xxx, there you should put chan_unicall.c Thank you very for this information, this will be a good start for me to search for this. 2. On what directory/folder should I commandpatch channels_makefile.patch?You need to learn somewhere else how to apply patches. Yeahhh, that's why I'am searching the WorlWideWeb. 3. On what directory/folder should I command make and make install?God, I think you need a basic linux install course, read in googleabout "Makefiles" Frankly speaking, YES, I need training for Linux Install. But before going to formal training I need to make this * box up running ASAP before my CIO fired me. I just thought learning through forums is faster and easier because we are guided based on actual experience. In fact, this is the reason I ended up in this forum because of my persistence to learn ina fastest way and of course in a most economical way, reason why we are into Open Source. (Savings). 4. What actually the asterisk do with patch channels_makefile.patch?Apply the patch to the Makefile code I am really very thankful to all of you guys, hope I would be able to help in some other subject matter and rest assured, I will post of any successful development with my project. Desperately need help.Everyone does. Thanks God, there are FORUMS as our references. Cheers! Angel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installation of Unicall for MFC/R2
Hi, Anyone there could figure me out on how to install my unicall. I followed the instruction belowin the statedsite at; http://soft-switch.org/unicall/installing-mfcr2.html. Questions: 1. On what directory/folder should I copy the chan_unicall.c, channels_makefile.patch? 2. On what directory/folder should I command patch channels_makefile.patch? 3. On what directory/folder should I command make and make install? 4. What actually the asterisk do with patch channels_makefile.patch? Sorry guys, cannot find more detailed information regarding installation of unicall. Particularly chan_unicall.c and channels_makefile.patch. Desperately need help. Thanks Angel___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] some simple newbie help with dialplan needed...
Hi, Could anyone knows what this error codes means; -- Got SIP response 481 "Call/Transaction Does Not Exist" back from SIP Gateway IP AddressThanks Angel___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall Installation
Hi Tzafrir, Thanks for your quick reply, I will look some downloads and install it as per your suggestion. I am using CentOS 4.3, kernel-2.6.9-34.01.EL Thanks again. Angel - Original Message From: Tzafrir Cohen [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, October 23, 2006 5:43:12 PMSubject: Re: [asterisk-users] Unicall Installation On Mon, Oct 23, 2006 at 02:11:22AM -0700, Angel Heart wrote: Hi, Thank you for your comment; Below was the result of./configure checking how to run the C++ preprocessor... /lib/cpp configure: error: C++ preprocessor "/lib/cpp" fails sanity check See `config.log' for more details. [EMAIL PROTECTED] libsupertone-0.0.2]# You don't have g++/gcc-c++ installed. You just need to install somepackages.Which Linux distribution do you use?-- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafriricq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406jabber:[EMAIL PROTECTED] http://www.xorcom.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall Installation
Hi, Could anyone knows whatwent wrong with theerror below result of installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf libsupertone-0.0.2.tarlibsupertone-0.0.2/libsupertone-0.0.2/AUTHORSlibsupertone-0.0.2/Makefile.amlibsupertone-0.0.2/COPYINGlibsupertone-0.0.2/config/libsupertone-0.0.2/config/ltmain.shlibsupertone-0.0.2/config/missinglibsupertone-0.0.2/config/install-shlibsupertone-0.0.2/config/config.guesslibsupertone-0.0.2/config/depcomplibsupertone-0.0.2/config/config.sublibsupertone-0.0.2/configurelibsupertone-0.0.2/NEWSlibsupertone-0.0.2/libsupertone.speclibsupertone-0.0.2/ChangeLoglibsupertone-0.0.2/Makefile.inlibsupertone-0.0.2/supertone.clibsupertone-0.0.2/configure.inlibsupertone-0.0.2/libsupertone.hlibsupertone-0.0.2/INSTALLlibsupertone-0.0.2/supertone.hlibsupertone-0.0.2/libsupertone.spec.inlibsupertone-0.0.2/READMElibsupertone-0.0.2/supertone_tests.clibsupertone-0.0.2/config-h.inlibsupertone-0.0.2/aclocal.m4[EMAIL PROTECTED] latest]# ./configure --prefix=/usr/local/lib-bash: ./configure: No such file or directory[EMAIL PROTECTED] latest]# Help, pleeeaaassseee... Angel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall Installation
Hi, Thank you for your comment; Below was the result of ./configure checking how to run the C++ preprocessor... /lib/cppconfigure: error: C++ preprocessor "/lib/cpp" fails sanity checkSee `config.log' for more details.[EMAIL PROTECTED] libsupertone-0.0.2]# Please comment. Thanks again. - Original Message From: Hadley Rich [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, October 23, 2006 5:01:16 PMSubject: Re: [asterisk-users] Unicall Installation On Monday 23 October 2006 21:45, Angel Heart wrote: Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf libsupertone-0.0.2.tar[snip] libsupertone-0.0.2/aclocal.m4 [EMAIL PROTECTED] latest]# ./configure --prefix=/usr/local/lib -bash: ./configure: No such file or directory [EMAIL PROTECTED] latest]# Help, pleeeaaassseee...You probably shouldn't blindly follow instructions if you don't know what they do../configure should be running the script called configure in the current directory. Which, as the error message states, doesn't exist. You need to change into the correct directory (cd) before you execute the script.-- http://nicegear.co.nzNew Zealand's VoIP Supplier___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN PRI and R2 Trunks in one Dual Port Card
Hi Guys,Anyone can tell where can I look all your previous post, I am wondering what could my zapata.conf be if I wanted to use two(2) different Trunk Protocol (ISDN R2) in a single Dual Port Digium Card. Sorry, I'm a new user in this forum and new asterisk user as well. Hope somebody could lend a hand/knowledge about this set-up.Thanks.Angel Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?
Hello Guys,Same thing with RR, we are currently intalling Asterisk with Digium TE210P Dual Card. We wanted to interconnect its one(1) E1 Port to a Nortel Meridian PABX and we have this kind of status at PABX side; (see below)DTI2 LOOP 18 - ENBL REF CLK: DSBLSERVICE RESTORE: YES ALARM STATUS: ACCEPTABLECH 01 - LCKO TIE VOD CH 02 - LCKO TIE VOD CH 03 - LCKO TIE VOD CH 04 - LCKO TIE VOD CH 05 - LCKO TIE VOD CH 06 - LCKO TIE VOD CH 07 - LCKO TIE VOD CH 08 - LCKO TIE VOD CH 09 - LCKO TIE VOD CH 10 - LCKO TIE VOD CH 11 - LCKO TIE VOD CH 12 - LCKO TIE VOD CH 13 - LCKO TIE VOD CH 14 - LCKO TIE VOD CH 15 - LCKO TIE VOD CH 16 - LCKO TIE VOD CH 17 - LCKO TIE VOD CH 18 - LCKO TIE VOD CH 19 - LCKO TIE VOD CH 20 - LCKO TIE VOD CH 21 - LCKO TIE VOD CH 22 - LCKO TIE VOD CH 23 - LCKO TIE VOD CH 24 - LCKO TIE VOD CH 25 - LCKO TIE VOD CH 26 - LCKO TIE VOD CH 27 - LCKO TIE VOD CH 28 - LCKO TIE VOD CH 29 - LCKO TIE VOD CH 30 - LCKO TIE VOD DCH000 .***OVL000 OVL000 err dta204 DTA0204 loop eIncomplete or incorrect ANI was received by CDTI2/CSDTI2 FW (outgoingtrunk) which reports this fact to the main CPU by SSD messin 9 (NIproblem report).This is a "warning only" message. No additional actions are taken. Thecall will be processed regularly. However, FW will use the specialhardcoded in FW value as ANI.Severity: Minor I wonder whatwe missed in our configurations. Zaptel.conf is as follows; span=1,1,0,cas,hdb3 # cas=1-15:1101 cas=17-31:1101 dchan=16loadzone= usdefaultzone= us Any assistance is greatly appreciated. Another thing, can we possibly configure the other E1 port for PSTN (ISDN)TieTrunk?Thank you in advance.RegardsAngel "R.R. Libera" [EMAIL PROTECTED] wrote: Hello,Has somebody installed this configuration: Asterisk + E1 with MFC/R2 (Telefónica Argentina) in Argentina before? I need to know if it´s possible with MFC/R2 argentine variation.Thanks in advance.R.R. Libera___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
