Re: [Asterisk-Users] Would DECT cordless phones work with Asterisk and VOIP?
On Sun, November 27, 2005 1:46, Larry Alkoff said: I've just heard about DECT which is used for about 50 million phones in Europe and is just starting to appear in the US. DECT stands for Digitally Enhanced Cordless Telephone and supposedly has much greater range than other cordless telephony. Additionally, you can purchase repeaters that will greatly range. Since reading about poor audio quality and echo issues caused by repeated conversions as the signal traverses the path from the (possibly POTS analogue) station, over tcp/ip and to destination, would DECT (another digital form) agravate this? In my house, a Uniden 5.8 and Panasonic 2.4 cordless system would only work over about 35 feet indoors - not enough for a large house. Does anyone have any hands-on experience with DECT? Larry I have a Siemens Gigaset 4030 DECT station at home. It is an ISDN deskphone with built-in DECT access point. I have finally got 2 HFC-PCI cards running simultaneously in TE and NT mode, and will be connecting the 4030 to the NT mode HFC-PCI card soon. (I'd like to do it today, but my MIL has 'invited' us for ST.Nicholas (Dutch fest, usually on 5 Dec, with lots of children's gifts, similar to Xmas Eve in USA) today!) As soon as I have some test-experience with the setup I will: - Let the list know (both this thread and the '2 hfc-pci cards do not work' thread) - Write an extensive wiki article on the setup for others to learn from... ;-) Cheers -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] Would DECT cordless phones work with Asterisk and VOIP?
On Sun, Nov 27, 2005 at 09:11:46AM +0100, Francesco Peeters wrote: As soon as I have some test-experience with the setup I will: - Let the list know (both this thread and the '2 hfc-pci cards do not work' thread) - Write an extensive wiki article on the setup for others to learn from... ;-) Siemens DECT 4XXX, two Longshine LCS8051 HFC Cards, VISDN as driver (www.visdn.org), one NT Mode (NTBA connectd, DECT on NTBA), one TE Mode to POTS. Box: AMD 400Mhz, 256 M Ram. works like a charme, either ISDN-SIP, or ISDN-ISDN. Btw, i have not soldered anything on the cards, why did you crossconnect the timers on your cards? -- http://www.ukeer.de/about.html I can assure you that data processing is a fad that won't last the year. --Chief Business Editor, Prentice Hall, 1957 ___ --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] Problem about outgoing calls with verizon.
It works. For all the verizon lines, I add w before the dialstring in [EMAIL PROTECTED] setup. It now works fine. Thanks a lot. On 11/26/05, Steve Totaro [EMAIL PROTECTED] wrote: Add a w (wait) to your dialstring.Chances are asterisk is dialingbefore you are getting dialtone.You can put a butt set on the line and listen while dialing out to verify this before adding the w.Sometimesit may take ww to get it working correctly.Thanks,Steve_From: Zheng Fang [mailto: [EMAIL PROTECTED]]Sent: Friday, November 25, 2005 9:45 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Problem about outgoing calls with verizon. Hi, everyone,I have met a very strange problem. We use a Asterisk PBX connected withtwo rollover PSTN phone line provided by Verizon by Digium TDM cards.The incoming calls are always OK. But when I make outgoing calls, sometimes it works, sometimes it just get a busy tone, doesn't work atall. The strange thing is that when I use the same server with SBC linesperviously, it worked fine.What is the problem here? This drives me crazy. Thanks/frank___--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
RE: [Asterisk-Users] A rather big setup.
Remember that the E1 only gives you a 30 lines from each floor then! If you use a dedicated 100mbs ethernet and uses IAX trunks you can have much more lines from each floor. Just my 5 cents. Best regards --On Saturday, November 26, 2005 12:35:16 PM +0100 Vedran Dakic [EMAIL PROTECTED] wrote: You mean 240 / 1000 simultaneous calls or 240 outside lines and 1000 internal phones ? I can only guess that I should have the ability to deliver a solution that can do some 100/500 simultaneously. The only question is how powerful should be a machine (or machines) that could do around 100/500 simultaneously. And, just for the sake of knowing, what should the setup be alike if it was 240/1000 simultaneously? In the second case there's no need for a cluster, a good server will do, (obviously a second server for backup is a good idea ). I'm assuming you can use a/ulaw to transmit the data, if bandwidth is a problem and you must compress cpu usage becomes a boottleneck to keep in mind. A/ulaw? I saw some reports that G.729 uses very little bandwidth and has a quality part granted (audio quality). It's not a question of hardware and/or CPU power, I have two dual Opteron configurations and could install some more, it's just the question of that setup running with quality audio and no unwanted events. I presume that I should have all of the phones using the same codec (so, no transcoding), and preferrably the same VoIP protocol. I have a choice there so everything's possible. Let's say - IP10s has H.323, SIP and MGCP firmwares, although I'd like to leave H.323 out of the story. I'm having ~80 concurrent calls from iax/sip to pri in alaw from an userbase of ~150 clients and the cpu is around 6% on a dual 2.8 Ghz. 1000 phones are a lot, and sip sometimes is an hassle (mostly nat), I don't know your network topology, but maybe you can consider to connect every group of phones to an asterisk pc and the pcs to the server via iax, which uses a little less bandwidth and most of all works out of the box. A pentium 400 can handle ~8 calls with ilbc, so every modern pc will do. Maybe I have a better idea, now that I come to think of it. Maybe I should install one Asterisk server per floor (8-9 floors) and use IAX to connect to the central server with E1 connections. Does that sound reasonable? ___ --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 -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B pgp25NPsnNKhT.pgp Description: PGP signature ___ --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] Would DECT cordless phones work with Asterisk and VOIP?
Larry Alkoff wrote: In my house, a Uniden 5.8 and Panasonic 2.4 cordless system would only work over about 35 feet indoors - not enough for a large house. Does anyone have any hands-on experience with DECT? I have recently discovered kirk (kirktelecom.com) wich also uses something like DECT, but according to their website, a modified version. Now, what makes this interesting is, you can use 'repeaters' so you can even bridge larger distances. (you can use multiple repeaters, but there is a maximum, check their website) To summarize, I'm using: IP600 and a Kirk4020 and 4040 and a repeater-box. The only drawback is, you can't use regular dect-phones with this system. I found them to be very reliable, and when using chan_sccp, you can interface very smoothly with asterisk. cheers! ___ --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] Would DECT cordless phones work with Asterisk and VOIP?
Larry Alkoff wrote: Snip ... Does anyone have any hands-on experience with DECT? We have an old BT DECT phone in the house, connectd to an FXS port on a TDM400 card. We get a burst of white noise (inaudible to the guys at the other end) for about half a second when we pick up, but apart from that it's fine. Range is about 100m in clear air. We have 3 ft thick stone walls and it copes with that very well. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
Francesco Peeters wrote: Reshuffled the cards in my machine (actually inverted the order of the PCI cards) and problem solved! Indeed, if the bios is is not working 'with' you, but more against you, then shuffling could solve the problem, glad it did :-) It appears the bottom 2 slots of the MoBo share an IRQ line, period... Now the cards have different IRQs, and both work, *with* the sync connection in place (i.e. the NT running of the TE clock) Brilliant! Now to play with the dialplan and integrate a zap-channel phone (or actually: 4 phones... G) Excellent, now the 'really' fun part starts out :-) 1.2 has 'soo' many possibilities :) cheers! ___ --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] Small office with all employee's offsite
Aldo Bergamini wrote: [EMAIL PROTECTED] is believed to have said: Jason, I'm sure these questions have been answered at some point, but I'm too new to this stuff to know the right words to plug into the search function to find what I need. well, yes of course. I have never touched Asterisk before, but have wanted to for some time. Now I finally think I'm going to bite the bullet, as I have a real-world application for it! You are in for some fun and satisfaction; with some small price to pay... My office consists of two employees, neither of whom work in the office physically. Here is what I'd like to do. Hopefully someone can tell me what I need to do/buy/configure/install to make it work... As a minimum set up you will need a CPU plus an interface to your incoming phone lines and most likely to an extension line in the main office. I want all calls to come into the Asterisk box in the main office. Obvious. I want all incoming calls to be recorded (not as concerned about outgoing calls) Can be done from the dialplan. Both employees have regular POTS telephone lines (one fellow has a land line and a cell, the other has just a land-line). I'd like callers to be presented with a short menu of options, the behavior of which might change depending on the time of day (for instance, at night, I'd like both the sales and support calls to go to one employee, while during the day I'd like sales to go to one person, and support to go to another. I'd also like to have an answering machine (built into Asterisk?) pick up calls that go unanswered. Can be done from the dialplan. Voicemail is an Asterisk application. I guess that's about it. I looked at the Digium TDMxx cards, but don't really know what I need in the way of FXO's and FXS's to pull off what I want to do. That's a very good option. As an added bonus, if someone knows of a VOIP adapter that allows one to plug an analog phone into it AND accept both VOIP and normal phone calls to the same phone, that would be cool (and might make things easier to configure, without making each extension 100% dependent on VOIP). You could look into products from Sipura or from Grandstream. Thanks in advance. I'm really looking forward to finally doing something with Asterisk, one of the most exciting projects I've looked at for a while!! But the very best advice I can give you is to start getting used to the Asterisk wiki and get the O'Reilly book on Asterik: it will be your friend. That's the small price to be paid. I found it worth. Regards Aldo ___ --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 As mentioned the SPA3000 has two ports - one for a handset, one for a phone line. They hook into your asterisk as *two* (SIP) devices, giving four ways to use them: - incomming call from telco passed to asterisk (inbound call routing) - asterisk can make outgoing calls through this line (outbound call routing) - asterisk can ring the handset as an extension ( --- you want this one ) - handset can be used to ring other extension (--- possibly this also - ring your partner for free over voip) In the event of a power cut, the SPA joins the lines together - so you will still have local calling. Your going to want a 'dial plan' typed into the SPA3000 config so that normal calls are routed out of the analogue line rather than to asterisk and back. Chris ___ --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] Context mix-up
I have two fwd accounts, and I want them to behave differently. It took me a while to figure out why it wouldn't work, but finally I realized that the last definition in sip.conf is the one that steals the show. Simplified, I have this: register = account1:[EMAIL PROTECTED]/88 register = account2:[EMAIL PROTECTED]/87 [fwdaccount1] context = context1 host=fwd.pulver.com . [fwdaccount2] context = context2 host=fwd.pulver.com . In extensions.conf: [context1] exten = 88,1,NoOp(Testing context1) [context2] exten = 87,1,NoOp(Testing context2) What happens in my case, is that every call goes into the context defined _last_ in sip.conf. So any call to account1 will be branded context2 and fail, because extension 88 is not defined in context2. Calls to account2 will work ok. If the two definitions in sip.conf trade places, the whole thing will work the other way around. [fwdaccount2] context = context2 host=fwd.pulver.com . [fwdaccount1] context = context1 host=fwd.pulver.com . Calls to either account will be branded context1 and fail if account 2 was called. If this is how it is supposed to work, the workaround must be to let both accounts enter the same context and differentiate their behavior based on the extension dialed. Not difficult, but I thought it would be possible to let them have different contexts from the start. Thor ___ --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] speex ilbc
-- off-topic --- Sorry for bothering the list, but my messages to the list disappear in cyberspace. Just want to see if answering an existing message makes any difference. ___ --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] speex ilbc
-- still off-topic -- Yes it does arrive now. Are my previous posts stopped by some anti-spam mechanism, or is there a moderator that is blocking me (for some obscure reason)? Of course, the message that never arrives is not spam, it's about some dialplan problem. I sent it twice, none arrived. Strange. Thor ___ --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] A rather big setup.
Hmm, maybe I'm missing something here. So, just to be sure... I was thinking about having a separate Asterisk server/cluster in the -1 floor server room where all of the telco/other wires come in (with that 240 lines via 8 E1 wires), and one asterisk server per floor connected to Asterisk server/cluster in the basement. I don't understand what did you mean by this 30 lines from each floor :) Confused a bit here Couldn't I have these per-floor Asterisk servers connected directly via Ethernet/IAX to the -1 floor server room, and have those servers/cluster/whatever manage the calls? I wasn't thinking about installing one asterisk server per floor with E1 card inside. I was thinking about connecting all of those servers to the central server with all of the E1 lines inside. Isn't that possible? Cheers, Vedran. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Saell Sent: Sunday, November 27, 2005 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] A rather big setup. Remember that the E1 only gives you a 30 lines from each floor then! If you use a dedicated 100mbs ethernet and uses IAX trunks you can have much more lines from each floor. Just my 5 cents. Best regards ___ --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] Context mix-up
Help, my messages to the list disappear. I will post a follow-up to this message in just a sec. ___ --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] Context mix-up
I have two fwd accounts, and I want them to behave differently. It took me a while to figure out why it wouldn't work, but finally I realized that the last definition in sip.conf is the one that steals the show. Simplified, I have this: register = account1:[EMAIL PROTECTED]/88 register = account2:[EMAIL PROTECTED]/87 [fwdaccount1] context = context1 host=fwd.pulver.com . [fwdaccount2] context = context2 host=fwd.pulver.com . In extensions.conf: [context1] exten = 88,1,NoOp(Testing context1) [context2] exten = 87,1,NoOp(Testing context2) What happens in my case, is that every call goes into the context defined _last_ in sip.conf. So any call to account1 will be branded context2 and fail, because extension 88 is not defined in context2. Calls to account2 will work ok. If the two definitions in sip.conf trade places, the whole thing will work the other way around. [fwdaccount2] context = context2 host=fwd.pulver.com . [fwdaccount1] context = context1 host=fwd.pulver.com . Calls to either account will be branded context1 and fail if account 2 was called. If this is how it is supposed to work, the workaround must be to let both accounts enter the same context and differentiate their behavior based on the extension dialed. Not difficult, but I thought it would be possible to let them have different contexts from the start. Thor ___ --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] Context mix-up
Thor: All your messages seem to be making it to the list ok - I've seen this email at least 3 times. Are you perhaps blocking the list somewhere in your anti-spam setup? Roger Thor Atle Rustad wrote: I have two fwd accounts, and I want them to behave differently. It took me a while to figure out why it wouldn't work, but finally I realized that the last definition in sip.conf is the one that steals the show. Simplified, I have this: register = account1:[EMAIL PROTECTED]/88 register = account2:[EMAIL PROTECTED]/87 [fwdaccount1] context = context1 host=fwd.pulver.com . [fwdaccount2] context = context2 host=fwd.pulver.com . In extensions.conf: [context1] exten = 88,1,NoOp(Testing context1) [context2] exten = 87,1,NoOp(Testing context2) What happens in my case, is that every call goes into the context defined _last_ in sip.conf. So any call to account1 will be branded context2 and fail, because extension 88 is not defined in context2. Calls to account2 will work ok. If the two definitions in sip.conf trade places, the whole thing will work the other way around. [fwdaccount2] context = context2 host=fwd.pulver.com . [fwdaccount1] context = context1 host=fwd.pulver.com . Calls to either account will be branded context1 and fail if account 2 was called. If this is how it is supposed to work, the workaround must be to let both accounts enter the same context and differentiate their behavior based on the extension dialed. Not difficult, but I thought it would be possible to let them have different contexts from the start. Thor ___ --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 -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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] A rather big setup.
Well you are exactly right! You can have one box per floor. But you said that they where going to connect to the main astrisk with a E1 and then i guessed that you where tinking of using a E1 car in the box to the main one and not ethernet. Thats why i said what i did. So yes - connect them with Ethernet/IAX and is should work fine! Best regards jan --On Sunday, November 27, 2005 12:02:53 PM +0100 Vedran Dakic [EMAIL PROTECTED] wrote: Hmm, maybe I'm missing something here. So, just to be sure... I was thinking about having a separate Asterisk server/cluster in the -1 floor server room where all of the telco/other wires come in (with that 240 lines via 8 E1 wires), and one asterisk server per floor connected to Asterisk server/cluster in the basement. I don't understand what did you mean by this 30 lines from each floor :) Confused a bit here Couldn't I have these per-floor Asterisk servers connected directly via Ethernet/IAX to the -1 floor server room, and have those servers/cluster/whatever manage the calls? I wasn't thinking about installing one asterisk server per floor with E1 card inside. I was thinking about connecting all of those servers to the central server with all of the E1 lines inside. Isn't that possible? Cheers, Vedran. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Saell Sent: Sunday, November 27, 2005 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] A rather big setup. Remember that the E1 only gives you a 30 lines from each floor then! If you use a dedicated 100mbs ethernet and uses IAX trunks you can have much more lines from each floor. Sphinx Just my 5 cents. Best regards ___ --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 -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B pgpc93sOHzcm8.pgp Description: PGP signature ___ --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] Would DECT cordless phones work with Asterisk and VOIP?
On Sun, November 27, 2005 9:24, Rico -mc- Gloeckner said: On Sun, Nov 27, 2005 at 09:11:46AM +0100, Francesco Peeters wrote: As soon as I have some test-experience with the setup I will: - Let the list know (both this thread and the '2 hfc-pci cards do not work' thread) - Write an extensive wiki article on the setup for others to learn from... ;-) Siemens DECT 4XXX, two Longshine LCS8051 HFC Cards, VISDN as driver (www.visdn.org), one NT Mode (NTBA connectd, DECT on NTBA), one TE Mode to POTS. Box: AMD 400Mhz, 256 M Ram. works like a charme, either ISDN-SIP, or ISDN-ISDN. Btw, i have not soldered anything on the cards, why did you crossconnect the timers on your cards? The Florz patch allows for timer slave mode, which means the buffer timers for the NT cards are derived from (one of) the TE card(s), That way the NT cards are always synchronized to the ISDN PSTN, which would mean that there ir virtually nill chance of buffer over- or underruns, thus improving call quality when crossconnecting the cards for PSTN calls... It is not really crossconnecting, it is parallel connecting the cards, ie all NT cards and *one* TE card have their cologne chip's pins 54 connected to eachother, as well as all their pins 53 and 52... I have followed Florian's tip to make connectors, so you can connect them using a cable, rather than having them always connected. This also allows me to test with both configurations: double master or master/slave... More to come... ;-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] Intel G729 Codec Install error on [EMAIL PROTECTED] 2.0
hi there: I try to install the Intel G729 codec(http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/) on [EMAIL PROTECTED] 2.0 box, the Intel library can be installed successfully. When I try to compile the codec sample, It always return me the complied error. But I try to install this codec on the [EMAIL PROTECTED] 2.0b4 version, everything is working properly. The errors as the following: make: *** [samples/codec_g729.o] Error 1 Does anybody know what factor of AAH 2.0 cause this problem happend? Thank you for your help. ___ --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] Dialplan help
hi, can anyone please guide me as to how i can implement this in extensions.conf: my PSTN line normally has its longdistance capability locked which can be opened by dialing some keys and the PIN. if i wanted some users to be allowed to call long distance using the zap channel, how can i initially offhook then send some DTMF then Flash twice before proceeding with the dialing EXTEN. ___ --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] calling to mgcp device
I've just tested a mgcp setup. I'm trying to use mgcp devices (mediatrix 1102) with dynamic ip. First one with static ip: [192.168.1.104] context=from-internal host= 192.168.1.104 wcardep = aaln/* callerid= 306 306 callwaiting = no callreturn = no cancallforward = no canreinvite = no transfer= no dtmfmode= rfc2833 line = aaln/1 callerid= 307 307 callwaiting = no callreturn = no cancallforward = no canreinvite = no transfer= no dtmfmode= rfc2833 line = aaln/2 The device works ok and I can call other phones. To call 306 I set up to dial to MGCP/aaln/[EMAIL PROTECTED] It works ok. The problem is now with dynamic ip. The set up I made is this: [anv] context=from-internal host= dynamic wcardep = aaln/* callerid= 308 308 callwaiting = no callreturn = no cancallforward = no canreinvite = no transfer= no dtmfmode= rfc2833 line = aaln/1 callerid= 309 309 callwaiting = no callreturn = no cancallforward = no canreinvite = no transfer= no dtmfmode= rfc2833 line = aaln/2 The device, again works ok but I can call to 308 dialing to MGCP/aaln/[EMAIL PROTECTED] What can be the problem? Is correct to dial MGCP/aaln/[EMAIL PROTECTED] I'm using [EMAIL PROTECTED] 2.0 beta 6 -- Alejandro Vargas ___ --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] Context mix-up
I have now received the messages I sent today, this seems to have happened after I updated some settings at digium.com's list server. Why that would matter, I don't know. According to the list server, I had a bounce score of 1 (of 5). Therefore I changed a setting or two just let the server I still exist. Maybe the fault lies within gmail.com? Still, the two I sent yesterday remain in cyberspace. I have been able to post follow-ups all along, but yesterday, when creating a new thread, I didn't see it, nor any replies. Thor ___ --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] trunk not registering -newbie
Hello: My by-the-book [EMAIL PROTECTED]/Broadvoice installation doesn't register in the SIP Registry, IAX2 SIP Registry, or with SIP peers. The Asterisk server is behind a firewall using NAT. The checkpoint firewall opens up all IP Telephony ports and I manually opened up ports 4000-2. There do not seem to be any issues related to the NAT, firewall or network. I tried re-directing port 5060 to the asterisk server and adding this to the sip_additional.conf: port = 5060 externIP = 62.219.212.2 localnet = 192.168.10.0 localmask 255.255.255.0 nat=1 I got the same results when I set the trunk up as IAX2. This is what show up in the Asterisk info: Sip Registry Name/usernameHostDyn Nat ACL Mask Port Status Verbosity is at least 3 -- Remote UNIX connection disconnected Sip Peers HostUsername Refresh State -- Remote UNIX connection IAX2 Sip Registry Host UsernamePerceived Refresh State -- Remote UNIX connection -- Remote UNIX connection disconnected IAX2 Peers Name/UsernameHost Mask Port Status bv/3109439023147.135.20.128 (S) 255.255.255.255 4569 Unmonitored -- Remote UNIX connection -- Remote UNIX connection disconnected These are the errors that appear in the log file: Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'IAX2/BV' in family 'cfb' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'IAX2/SIP.BROADVOICE.COM' in family 'cfb' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'SIP/1000' in family 'cfb' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'IAX2/BV' in family 'dnd' Also: Nov 25 11:51:10 WARNING[2019]: mybvpassword is not a valid port number at line 1 Also: Nov 26 06:22:54 WARNING[2018]: Unknown directive 'permit=192.168.1.0/255.255.255.0' at line 18 of manager_custom.conf even though this is rem'ed out in manager_custom.conf #permit=192.168.1.0/255.255.255.0 I followed the instruction listed at: http://mundy.org/blog/index.php?p=66 this is my sip_additional.conf: [EMAIL PROTECTED]:mybvpassword:[EMAIL PROTECTED] [bv] username=3109439023 user=phone type=peer secret=mybvpassword nat=yes insecure=very host=sip.broadvoice.com fromuser=3109439023 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband canreinvite=no authname=3109439023 [sip.broadvoice.com] username=3109439023 user=3109439023 type=user secret=mybvpassword nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=rfc2833 dtmf=rfc2833 context=from-pstn Any help would be appreciated. Thanks, Billy Troper __ Yahoo! Mail - PC Magazine Editors' Choice 2005 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
[Asterisk-Users] zaptel 1.2.0 and correct settings in zapata.conf for Germany
Hi, everything works fine with zaptel 1.2.0 and TE405P. The only thing i am missing is the callerid for incoming calls. It is always empty. That worked with 1.0.9. -- Accepting overlap call from '' to '9671987' on channel 0/2, span 1 Are there any missing setting in the zapata.conf to make the incoming callerid number visible? Thanks and regards BK ___ --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] A rather big setup.
Vedran Dakic ha scritto: I can only guess that I should have the ability to deliver a solution that can do some 100/500 simultaneously. The only question is how powerful should be a machine (or machines) that could do around 100/500 simultaneously. And, just for the sake of knowing, what should the setup be alike if it was 240/1000 simultaneously? My suggestion is to buy the E1 cards first of all and put them in a test server, equipped with asterisk and all the relevant agi / db connections / moh etc.. Then loop the card with a crossover cable and run some test script to generate the medium and upper bound call flows. That should give you an idea of your cpu/ram requirements. In the second case there's no need for a cluster, a good server will do, (obviously a second server for backup is a good idea ). I'm assuming you can use a/ulaw to transmit the data, if bandwidth is a problem and you must compress cpu usage becomes a boottleneck to keep in mind. A/ulaw? I saw some reports that G.729 uses very little bandwidth and has a quality part granted (audio quality). It's not a question of hardware and/or CPU power, I have two dual Opteron configurations and could install some more, it's just the question of that setup running with quality audio and no unwanted events. G729 has a very good quality -considered the bandwidth used-, but if your customers are used to conventional telephony they will no doubt notice the difference, so go with G711 (probably alaw, since you use E1 I suppose you are in europe) Anyway if bandwidth is a problem consider ilbc / speex which are free and have good audio qualities also. Lastly a lot of the quality comes from a well configured phone, tweak with volumes and timeouts. I presume that I should have all of the phones using the same codec (so, no transcoding), and preferrably the same VoIP protocol. I have a choice there so everything's possible. Let's say - IP10s has H.323, SIP and MGCP firmwares, although I'd like to leave H.323 out of the story. Yes, leaving H323 out of the story is a good way to start the project :) ___ --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] Dialplan help
hi, can anyone please guide me as to how i can implement this in extensions.conf: my PSTN line normally has its longdistance capability locked which can be opened by dialing some keys and the PIN. if i wanted some users to be allowed to call long distance using the zap channel, how can i initially offhook then send some DTMF then Flash twice before proceeding with the dialing EXTEN. Do you want asterisk to take care of the DTMF for you? When you say users, do you mean certain extensions? Why not cancel your restrictions with the telco and implement restrictions within asterisk. You can use authenticate before your dial statement to prompt for a pin when dialing long distance or you can create different contexts for phones that you want to access long distance and for phones that you don't Thanks, Steve ___ --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] IAx/g729 client for MAC
Is there a good quality stable (not free) IAX2 client for MAC? I have a client wants to travel and make calls and I want to avoid the SIP blocking that is a problem for travellers. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --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] Intel G729 Codec Install error on [EMAIL PROTECTED] 2.0
On Sun, Nov 27, 2005 at 10:56:50PM +1100, wei li wrote: hi there: I try to install the Intel G729 codec(http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/) on [EMAIL PROTECTED] 2.0 box, the Intel library can be installed successfully. When I try to compile the codec sample, It always return me the complied error. But I try to install this codec on the [EMAIL PROTECTED] 2.0b4 version, everything is working properly. The errors as the following: make: *** [samples/codec_g729.o] Error 1 I really have no idea about this subject , but this error from make means that there was a more specific error message a bit higher. Could you please paste a more complete log? Try: make 21 | tee log -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Context mix-up
I have two fwd accounts, and I want them to behave differently. It
took me a while to figure out why it wouldn't work, but finally I
realized that the last definition in sip.conf is the one that steals
the show.
Its a common issue with sip since it matches on ip address, etc. Check
the archives on 'how' sip finds a matching sip.conf entry. Change your
fwd accounts to iax and you will have more control.
In my case with fwd #61890, incoming calls include the fwd number, so
extensions.conf entries like this:
exten = 61890,1,NoOp,${CALLERID}
exten = 61890,2,Goto(bus-ivr-main|s|1)
exten = 61890,3,Hangup
work just fine. Your second number would simply have a different
exten = statement.
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Re: [Asterisk-Users] Context mix-up
I have now received the messages I sent today, this seems to have happened after I updated some settings at digium.com's list server. Why that would matter, I don't know. According to the list server, I had a bounce score of 1 (of 5). Therefore I changed a setting or two just let the server I still exist. Maybe the fault lies within gmail.com? Still, the two I sent yesterday remain in cyberspace. I have been able to post follow-ups all along, but yesterday, when creating a new thread, I didn't see it, nor any replies. I had the same problem which resulted from our broadband connection being down for a couple of days over Thanksgiving. Apparently an undeliverable email from the list server triggers a 'stop' function, and revisiting the list server page returns the sending of email again. Not a problem for me as long as one is aware of the functionality. (Kind of hard to miss it though with 200+ emails per day.) ___ --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] Dialplan help
thanks steve, the reason i cannot remove the restriction on the telco line is that an analog fone is connected to the phone jack of the x101p and some visitors occasionally use the fone and they're supposed to only call local toll free numbers. Your suggestion of doing the restrictions within asterisk is good but i have yet to buy an FXS port for the analog fone so i cannot do so. and for the extensions connected via softphones to asterisk, i wanted asterisk to take care of opening the lock before dialing the desired phone number. i was thinking of using the SendDTMF but i dont know how to get the zap line go off hook.On 11/27/05, Steve Totaro [EMAIL PROTECTED] wrote: hi, can anyone please guide me as to how i can implement this in extensions.conf: my PSTN line normally has its longdistance capability locked whichcan be opened by dialing some keys and the PIN. if i wanted some users to be allowed to call long distance using the zap channel, how can i initially offhook then send some DTMF then Flashtwice before proceeding with the dialing EXTEN.Do you want asterisk to take care of the DTMF for you?When you say users, do you mean certain extensions?Why not cancel your restrictions with the telco and implementrestrictions within asterisk.You can use authenticate before your dialstatement to prompt for a pin when dialing long distance or you can create different contexts for phones that you want to access longdistance and for phones that you don'tThanks,Steve___--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
Re: [Asterisk-Users] IAx/g729 client for MAC
On 27 Nov 2005, at 13:43, Chris Mason (Lists) wrote:Is there a good quality stable (not free) IAX2 client for MAC? I have a client wants to travel and make calls and I want to avoid the SIP blocking that is a problem for travellers.I have heard good things about http://www.loudhush.ro/ But haven't used it (yet).I couldn't get any of the other IAX2 clients to be stable on the MAC.I very much doubt you will find a g729 client for the mac. The thing with g729 is that youhave to license quite large numbers of clients just to get the patent holders to talk toyou.I'd go for GSM, it is nearly as effecient as g729, people are used to the way it sounds (frommobiles) and it is patent free.My traveling (windows carrying) users are getting on fine with IAX2 over GSM.Tim. http://www.westhawk.co.uk/ ___ --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] Zaptel errors on Debian
Hello, I am trying to install zaptel wcfxo with X101.P board on Debian sarge without success. (previously compiled and worked OK on Redhat kernel) On boot asterisk I have [chan_zap.so] = (Zapata Telephony w/PRI) !!!??? but I have no PRI, I have fxo, x1001P digium . As a matter of fact when you type version on Asterissk-1.2.0 : Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] ... or something. Anyone up there experienced that problem with Asterisk on Debian ? Any trick ? Eventually, how you define module type wcfxo in zconfig.h when you do not have PRI interface ? Thx, Geo ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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] Voicepulse Open Access status?
I have not had my 2 voicepulse open access numbers work for about 3 weeks now. I have even tried to make them work on a new server build using asterisk 1.2 I have several SIP and IAX DID's working fine. They worked fine on asterisk 1.0.x and easily were moved to the 1.2 setup. sip show registry indicates they are registered oaky but calls to both numbers go to voicepulse voice mail. If I setup hunt and fileters at the voicepulse web portal, that seems to work. For example, I can make the numbers ring my cell phone instead of going to voice mail. The primary number on the account works with the SPA-2000 ATA fine. I just used it to call the vonage number that comes into the asterisk system and I am using echo test as I type this. If anyone here has working voicepulse open access numbers could you please post sample lines from sip.conf and extensions.conf? If anyone here has been experiencing the same type of extended outage, I would like to hear about it because I am going to ask them to waive charges for at least one month of these softphone numbers. I was very tempted to put Please help!!! in the subject line today. If I don't get any replies I guess that means voicepulse sucks and I should cancel these numbers :) ___ --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] OH323 channel in asterisk 1.2 ...Ouch ... error while writing audio data!
You should have more info in full log messages, look to this file and send output. Adam Cytowanie Rafael R. GV [EMAIL PROTECTED]: Hello I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk 1.2libraries, must be oh323-0.7.3, now I have compiled this version but when reload asterisk i have this error: [chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe Any idea??? -- rrgv ___ --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] Voicepulse Open Access status?
Debug info and posting your .confs will help to get replys. -Original Message- From: Paul [mailto:[EMAIL PROTECTED] Sent: Sunday, November 27, 2005 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicepulse Open Access status? I have not had my 2 voicepulse open access numbers work for about 3 weeks now. I have even tried to make them work on a new server build using asterisk 1.2 I have several SIP and IAX DID's working fine. They worked fine on asterisk 1.0.x and easily were moved to the 1.2 setup. sip show registry indicates they are registered oaky but calls to both numbers go to voicepulse voice mail. If I setup hunt and fileters at the voicepulse web portal, that seems to work. For example, I can make the numbers ring my cell phone instead of going to voice mail. The primary number on the account works with the SPA-2000 ATA fine. I just used it to call the vonage number that comes into the asterisk system and I am using echo test as I type this. If anyone here has working voicepulse open access numbers could you please post sample lines from sip.conf and extensions.conf? If anyone here has been experiencing the same type of extended outage, I would like to hear about it because I am going to ask them to waive charges for at least one month of these softphone numbers. I was very tempted to put Please help!!! in the subject line today. If I don't get any replies I guess that means voicepulse sucks and I should cancel these numbers :) ___ --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
Re: [Asterisk-Users] WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.
On Saturday November 26 2005 1:41 pm, John Millican wrote: On Saturday November 26 2005 1:26 pm, John Millican wrote: Hello all, I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls as expected. I have been trying to get atxfer working and am getting the error message: WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. whenever I try a transfer. In features.conf: [general] parkext = 700; parkpos = 701-720; context = parkedcalls; ;parkingtime = 45; transferdigittimeout = 3; courtesytone = beep; xfersound = beep; [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer [applicationmap] testfeature = #9,callee,Playback,tt-monkeys; in extensions.conf [globals] DYNAMIC_FEATURES=automon#blindxfer#disconect#atxfer#testfeature in CLI when attempting a transfer: SIP/677-8544 answered Zap/1-1 -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') Nov 26 13:13:33 WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. -- Playing 'beeperr' (language 'en') -- Stopped music on hold on Zap/1-1 and then the channels are joined again as if nothing had happened. I googled for the error message and searched voip-info.org but no results on either. Sorry for the second post but thought I should add some info. setup is: PSTN --- X100P in asterisk box - Linksys PA2-NA phone1 on port 1 and Phone2 on port 2 the linsys is set to g711 ulaw with inband signaling I am trying to transfer an incoming call from phone1 to phone 2 Okay let me try once again. When I attempt a transfer either blind or attended i get the transfer prompt and then dial tone as I should. Then what happens is when I press a digit the dial tone may or may not go away. If I repeat that first digit I can sometimes get the dial tone to go away and asterisk accepts the remaining digits for the transfer without problem and the transfer happens. I have tried increasing the dtmf playback level in the PAP2 from -16db all the way up to 0db. This has not made any noticeable difference in detection. I have also increased the DTMF playback length from .1 to .3 again no success. Any help would be greatly appreciated. Thank You John Millican ___ --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] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Hi Folks, My TD400P cards will not dial out until someone dials in first. Error is : -- Executing Dial(SIP/8438-3e69, Zap/g2/94248438|25|t) in new stack Nov 27 11:57:12 NOTICE[5599]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/8438-3e69, ) in new stack Using latest CVS's. Any reset I can do on rebooting the system to set these channels up properly? thanks, JES smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Zaptel errors on Debian
Hi On Sun, Nov 27, 2005 at 03:40:39PM -0800, Geotrix wrote: Hello, I am trying to install zaptel wcfxo with X101.P board on Debian sarge without success. (previously compiled and worked OK on Redhat kernel) On boot asterisk I have [chan_zap.so] = (Zapata Telephony w/PRI) !!!??? but I have no PRI, I have fxo, x1001P digium . As a matter of fact when you type version on Asterissk-1.2.0 : Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] ... or something. Do you use the debs or build your own? apt-get install zaptel-source and use 'm-a a-i zaptel' to get a zaptel modules package. Anyone up there experienced that problem with Asterisk on Debian ? Any trick ? Eventually, how you define module type wcfxo in zconfig.h when you do not have PRI interface ? Begin: self promotion Hmmm. why bother? http://xorcom.com/rapid will give you a fully-functioning Debian Sarge system with Zaptel pre-compiled. End: self promotion -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Voicepulse Open Access status?
I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. Steve Totaro wrote: Debug info and posting your .confs will help to get replys. -Original Message- From: Paul [mailto:[EMAIL PROTECTED] Sent: Sunday, November 27, 2005 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicepulse Open Access status? I have not had my 2 voicepulse open access numbers work for about 3 weeks now. I have even tried to make them work on a new server build using asterisk 1.2 I have several SIP and IAX DID's working fine. They worked fine on asterisk 1.0.x and easily were moved to the 1.2 setup. sip show registry indicates they are registered oaky but calls to both numbers go to voicepulse voice mail. If I setup hunt and fileters at the voicepulse web portal, that seems to work. For example, I can make the numbers ring my cell phone instead of going to voice mail. The primary number on the account works with the SPA-2000 ATA fine. I just used it to call the vonage number that comes into the asterisk system and I am using echo test as I type this. If anyone here has working voicepulse open access numbers could you please post sample lines from sip.conf and extensions.conf? If anyone here has been experiencing the same type of extended outage, I would like to hear about it because I am going to ask them to waive charges for at least one month of these softphone numbers. I was very tempted to put Please help!!! in the subject line today. If I don't get any replies I guess that means voicepulse sucks and I should cancel these numbers :) ___ --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
Re: [Asterisk-Users] Would DECT cordless phones work with Asterisk and VOIP?
On Sat, 26 Nov 2005 18:46:17 -0600, Larry Alkoff wrote: I've just heard about DECT which is used for about 50 million phones in Europe and is just starting to appear in the US. DECT stands for Digitally Enhanced Cordless Telephone and supposedly has much greater range than other cordless telephony. Additionally, you can purchase repeaters that will greatly range. Since reading about poor audio quality and echo issues caused by repeated conversions as the signal traverses the path from the (possibly POTS analogue) station, over tcp/ip and to destination, would DECT (another digital form) agravate this? In my house, a Uniden 5.8 and Panasonic 2.4 cordless system would only work over about 35 feet indoors - not enough for a large house. Does anyone have any hands-on experience with DECT? Larry The DECT implementation we see in the US are I presume deployed with normal analog line connections, unlike in Europe where many phones are ISDN based. Thus you may not be able to avoid the D A D conversion. ISDN cards that work with US standard ISDN lines are essentially unheard of. Many of us who have been dissatisfied with small FXO adapters would LOVE to order up ISDN lines and skip the whole FXO problem altogether. There are several/many ISDN cards that work with Asterisk in Europe. I've yet to find even one that would work with an SBC ISDN drop. I'd welcome someone proving me wrong on this. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --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] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
There are no resets required. You apparently have something misconfigured in either zapata.conf or extensions.conf, but without seeing these it is difficult to guess at what you've done. Best guess is that zapata.conf is messed up, and likely associated with the group (g2) definitions. From: James B. MacLean [EMAIL PROTECTED] My TD400P cards will not dial out until someone dials in first. Error is : -- Executing Dial(SIP/8438-3e69, Zap/g2/94248438|25|t) in new stack Nov 27 11:57:12 NOTICE[5599]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/8438-3e69, ) in new stack Using latest CVS's. Any reset I can do on rebooting the system to set these channels up properly? ___ --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] Voicepulse Open Access status?
I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. At the CLI, type 'sip debug' and call the numbers again. There should be something in the debug messages that point to the problem. Post the results. ___ --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] UK, London B ased DID £1 per month
This should be posted to the Asterisk-Biz list. This list for Non-Commercial Discussion of Asterisk. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Hi Rich, Files attached. As mentioned, once the first call is made in on the lines (3 and 4) then outgoing are great :). JES Rich Adamson wrote: There are no resets required. You apparently have something misconfigured in either zapata.conf or extensions.conf, but without seeing these it is difficult to guess at what you've done. Best guess is that zapata.conf is messed up, and likely associated with the group (g2) definitions. From: James B. MacLean [EMAIL PROTECTED] My TD400P cards will not dial out until someone dials in first. Error is : -- Executing Dial(SIP/8438-3e69, Zap/g2/94248438|25|t) in new stack Nov 27 11:57:12 NOTICE[5599]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/8438-3e69, ) in new stack Using latest CVS's. Any reset I can do on rebooting the system to set these channels up properly? ___ --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 loadzone = us defaultzone=us fxoks=1-2 fxsks=3-4 fxoks=5-6 fxsks=7-8 [trunkgroups] [channels] language=en context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=128 callerid=asreceived echocancelwhenbridged=yes echotraining=yes rxgain=15 txgain=-4 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=8 callprogress=no musiconhold=default rxgain=10.5 txgain=-4.5 signalling=fxo_ks context=5th-floor callerid=Back Door (902) 424- channel = 1 rxgain=10.5 txgain=-4.5 signalling=fxo_ks context=5th-floor callerid=Front Desk Door (902) 424-9998 channel = 2 rxgain=10.5 txgain=-4.5 group=5 signalling=fxo_ks context=inbound-ednet callerid=EDnet (902) 424-6800 channel = 5 rxgain=10.5 txgain=-4.5 group=5 signalling=fxo_ks context=inbound-ednet callerid=EDnet (902) 424-6800 channel = 6 rxgain=5.0 txgain=-5.0 signalling=fxs_ks group = 2 context=mainmenu callerid=asreceived channel = 3 signalling=fxs_ks context=mainmenu group = 2 callerid=asreceived channel = 4 rxgain=10.0 txgain=-5.0 signalling=fxs_ks context=outbound-ednet group = 3 callerid=asreceived channel = 7 rxgain=10.0 txgain=-5.0 signalling=fxs_ks context=outbound-ednet group = 3 callerid=asreceived channel = 8 smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Voicepulse Open Access status?
I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. At the CLI, type 'sip debug' and call the numbers again. There should be something in the debug messages that point to the problem. Post the results. IAX debug as well. ___ --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] A question about transfering calls
Hi all, I have a question about transfering calls. If I transfer a call to extension 4000 and nobody answers I want the call to be returned bak to me at extension 1000. How do I do that? Any help is apreciated! many thanks! ___ --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] A question about transfering calls
I have a question about transfering calls. If I transfer a
call to extension 4000 and nobody answers I want the call to
be returned bak to me at extension 1000. How do I do that?
Any help is apreciated! many thanks!
Try something like this:
macro internal (dialstring, fallback, timeout) {
Dial (${dialstring},${timeout});
switch (${DIALSTATUS}) {
default:
if (${fallback} != ) {
internal (${fallback},,,${timeout});
} else {
// however you normally handle non-answering (voicemail, etc.)
};
};
};
The dialstatus switch could be used to handle busy calls differently, for
example.
If you dial extensions in your dialplan like this:
exten = _2XX,1,Dial(SIP/${EXTEN},20)
Then try this instead:
exten = _2XX,1,Macro(internal,SIP/${EXTEN},SIP/${CALLERID(number)},20)
So the fallback route is defined as the originating callerid. You might want
to use a different method of identifying your fallback route.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
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RE: [Asterisk-Users] Dialplan help
How about this?
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
-or-
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
I have not seen restrictions set before dialing since local numbers
would not fall under the restriction but that is what you said. Usually
you dial the number first and if it is a long distance call, they prompt
for a code with a tone or something. The second example shoud give you
a two second pause before dialing the rest of the number ().
Thanks,
Steve
thanks steve,
the reason i cannot remove the restriction on the telco line is that
an
analog fone is connected to the phone jack of the x101p and some
visitors
occasionally use the fone and they're supposed to only call local toll
free numbers.
Your suggestion of doing the restrictions within asterisk is good but
i
have yet to buy an FXS port for the analog fone so i cannot do so.
and for the extensions connected via softphones to asterisk, i wanted
asterisk to take care of opening the lock before dialing the desired
phone
number.
i was thinking of using the SendDTMF but i dont know how to get the
zap
line go off hook.
On 11/27/05, Steve Totaro [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
hi,
can anyone please guide me as to how i can implement this in
extensions.conf:
my PSTN line normally has its longdistance capability locked
which
can
be opened by dialing some keys and the PIN.
if i wanted some users to be allowed to call long distance
using
the
zap
channel, how can i initially offhook then send some DTMF then
Flash
twice
before proceeding with the dialing EXTEN.
Do you want asterisk to take care of the DTMF for you? When you
say
users, do you mean certain extensions?
Why not cancel your restrictions with the telco and implement
restrictions within asterisk. You can use authenticate before
your
dial
statement to prompt for a pin when dialing long distance or you
can
create different contexts for phones that you want to access
long
distance and for phones that you don't
Thanks,
Steve
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Re: [Asterisk-Users] Voicepulse Open Access status?
Steve Totaro wrote: I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. At the CLI, type 'sip debug' and call the numbers again. There should be something in the debug messages that point to the problem. Post the results. IAX debug as well. This is voicepulse retail open access softphone - SIP only. I appreciate the replies but I don't see any from people reporting they are able to use voicepulse softphone accounts. Interesting. Anyway, I saw the following type thing: Looking for s0022 in default where s0022 is the username. So I added extensions that match and incoming now works. Looks to me like they have changed something at voicepulse with the usual policy of not notifying subscribers. ___ --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] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
From: James B. MacLean [EMAIL PROTECTED] ---End of Original Message- I hope the files that you attached have different names then what was shown since there is nothing used by * called zap.conf or zapa.conf. The parameters in zaptel.conf look correct if they match up with the appropriate modules (red green) installed on the TDM boards. The parameters in zapata.conf appear to be a little wild. As the parameters are read, they are inherited by each section below it unless otherwise changed. So, usecallingpres=yes (as an example only) is inherited by each channel (regardless of whether you know it as a fxo or fxs port). Restructuring the contents might be a good place to start. I'd suggest removing the majority of the statements and starting out with very basic definitions, then when things are working reasonably well, add in other statements (like callerid, etc). You might also take a look at 'zap show status' and 'zap show channels' to ensure definitions are reasonable. Might also try 'set debug 99' and/or 'set verbose 99' and watch an outgoing call to see if additional info is provided on the CLI. ___ --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] Bad Lines - What can the phone company do?
If everything else fails, check to see if 1) DSL is available and 2) if you can cancel within 15 days and not get a cencelation fee. Then order DSL for that line. The telco will HAVE to fix any really significant issues on the line before getting DSL to work on the line. Obviously, try to get the problem fixed before trying this. 8-) Justin Selleck ([EMAIL PROTECTED]) wrote: We suffer with some bad CO lines in the Seattle Redmond area. To compensate our gains have been tuned 10 rx and 2 tx. We have also had to add a 3 second wait to outgoing calls because many times the front of the number gets missed by the telco. Is there anything we can request from the phone company? They have checked our lines (probably just for tone) and say there is nothing wrong. Thanks! -Justin ___ --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] A question about transfering calls
Hi Chris,
Many thanks, will try it!
Thanks,
Christian
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Sunday, November 27, 2005 6:03 PM
Subject: RE: [Asterisk-Users] A question about transfering calls
I have a question about transfering calls. If I transfer a
call to extension 4000 and nobody answers I want the call to
be returned bak to me at extension 1000. How do I do that?
Any help is apreciated! many thanks!
Try something like this:
macro internal (dialstring, fallback, timeout) {
Dial (${dialstring},${timeout});
switch (${DIALSTATUS}) {
default:
if (${fallback} != ) {
internal (${fallback},,,${timeout});
} else {
// however you normally handle non-answering (voicemail, etc.)
};
};
};
The dialstatus switch could be used to handle busy calls differently, for
example.
If you dial extensions in your dialplan like this:
exten = _2XX,1,Dial(SIP/${EXTEN},20)
Then try this instead:
exten = _2XX,1,Macro(internal,SIP/${EXTEN},SIP/${CALLERID(number)},20)
So the fallback route is defined as the originating callerid. You might
want
to use a different method of identifying your fallback route.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
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Re: [Asterisk-Users] Dialplan help
Thanks Steve,
But this will not work for me because after yourcodehere the line
will give a confirmation tone (similar to a congestion tone only
faster) then after flashing or certain period will turn into a busytone
and to get the dialtone again i need to Flash again before i can dial
${EXTEN}.
On 11/28/05, Steve Totaro [EMAIL PROTECTED] wrote:
How about this?exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})-or-exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})I have not seen restrictions set before dialing since local numbers
would not fall under the restriction but that is what you said.Usuallyyou dial the number first and if it is a long distance call, they promptfor a code with a tone or something.The second example shoud give you
a two second pause before dialing the rest of the number ().Thanks,Steve thanks steve, the reason i cannot remove the restriction on the telco line is thatan
analog fone is connected to the phone jack of the x101p and somevisitors occasionally use the fone and they're supposed to only call local toll free numbers. Your suggestion of doing the restrictions within asterisk is good but
i have yet to buy an FXS port for the analog fone so i cannot do so. and for the extensions connected via softphones to asterisk, i wanted asterisk to take care of opening the lock before dialing the desired
phone number. i was thinking of using the SendDTMF but i dont know how to get thezap line go off hook. On 11/27/05, Steve Totaro
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi,
can anyone please guide me as to how i can implement this in extensions.conf: my PSTN line normally has its longdistance capability locked
which can be opened by dialing some keys and the PIN. if i wanted some users to be allowed to call long distanceusing the zap
channel, how can i initially offhook then send some DTMF then Flash twice before proceeding with the dialing EXTEN. Do you want asterisk to take care of the DTMF for you?When you
say users, do you mean certain extensions? Why not cancel your restrictions with the telco and implement restrictions within asterisk.You can use authenticate before
your dial statement to prompt for a pin when dialing long distance or youcan create different contexts for phones that you want to accesslong distance and for phones that you don't
Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com --
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Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Rich Adamson wrote: From: James B. MacLean [EMAIL PROTECTED] ---End of Original Message- I hope the files that you attached have different names then what was shown since there is nothing used by * called zap.conf or zapa.conf. :) Sorry, I renamed them after I grepped out the comments :(. The parameters in zaptel.conf look correct if they match up with the appropriate modules (red green) installed on the TDM boards. Asterisk:/usr/local/src/VOIP/zaptel# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) 8 channels configured. The parameters in zapata.conf appear to be a little wild. As the parameters are read, they are inherited by each section below it unless otherwise changed. So, usecallingpres=yes (as an example only) is inherited by each channel (regardless of whether you know it as a fxo or fxs port). Restructuring the contents might be a good place to start. Ah good eye :). I did not notice that was set to yes. It is the default from the install. I'll try it off. I'd suggest removing the majority of the statements and starting out with very basic definitions, then when things are working reasonably well, add in other statements (like callerid, etc). That was how things started in my setup... Very simple :). Actually they started with just one card :). Rarely rebooting. Now I have two and for other reasons have had to reboot a couple of times and started noticing this. I did notice the problem with one card, which means you can cut half of the information out at the end of the zapa.conf file. You might also take a look at 'zap show status' and 'zap show channels' to ensure definitions are reasonable. Asterisk*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 Wildcard TDM400P REV I Board 2 OK 0 0 0 Asterisk*CLI zap show channels Chan Extension Context Language MusicOnHold pseudooutbound-ednet en default 15th-floor en default 25th-floor en default 3mainmenuen default 4mainmenuen default 5inbound-ednet en default 6inbound-ednet en default 7outbound-ednet en default 8outbound-ednet en default Might also try 'set debug 99' and/or 'set verbose 99' and watch an outgoing call to see if additional info is provided on the CLI. Thanks. I'll try that. Calling 4 numbers to get things going is not to bad, but if we go for the 24 port version, I don't want to be the one calling in :). JES smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Dialplan help
Wow, what a pain. I would just pickup an FXS and be done with it.
Thanks Steve,
But this will not work for me because after yourcodehere the line
will
give a confirmation tone (similar to a congestion tone only faster)
then
after flashing or certain period will turn into a busytone and to get
the
dialtone again i need to Flash again before i can dial ${EXTEN}.
On 11/28/05, Steve Totaro [EMAIL PROTECTED] wrote:
How about this?
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
-or-
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
I have not seen restrictions set before dialing since local
numbers
would not fall under the restriction but that is what you said.
Usually
you dial the number first and if it is a long distance call,
they
prompt
for a code with a tone or something. The second example shoud
give
you
a two second pause before dialing the rest of the number ().
Thanks,
Steve
thanks steve,
the reason i cannot remove the restriction on the telco line
is
that
an
analog fone is connected to the phone jack of the x101p and
some
visitors
occasionally use the fone and they're supposed to only call
local
toll
free numbers.
Your suggestion of doing the restrictions within asterisk is
good
but
i
have yet to buy an FXS port for the analog fone so i cannot do
so.
and for the extensions connected via softphones to asterisk, i
wanted
asterisk to take care of opening the lock before dialing the
desired
phone
number.
i was thinking of using the SendDTMF but i dont know how to
get
the
zap
line go off hook.
On 11/27/05, Steve Totaro [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
hi,
can anyone please guide me as to how i can implement
this
in
extensions.conf:
my PSTN line normally has its longdistance capability
locked
which
can
be opened by dialing some keys and the PIN.
if i wanted some users to be allowed to call long
distance
using
the
zap
channel, how can i initially offhook then send some
DTMF
then
Flash
twice
before proceeding with the dialing EXTEN.
Do you want asterisk to take care of the DTMF for you?
When
you
say
users, do you mean certain extensions?
Why not cancel your restrictions with the telco and
implement
restrictions within asterisk. You can use authenticate
before
your
dial
statement to prompt for a pin when dialing long distance
or
you
can
create different contexts for phones that you want to
access
long
distance and for phones that you don't
Thanks,
Steve
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[Asterisk-Users] ANNOUNCEMENT: Asterisk-Java 0.2 released
Asterisk-Java 0.2, a Java control for the Asterisk PBX, has been released. The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API. The 0.2 release focuses on the new features of the Asterisk 1.2 series though it still supports Asterisk 1.0.x. Since 0.2-rc2 some minor bugs have been fixed and support for several last minute additions to Asterisk 1.2 has been added. Asterisk-Java is used in several commercial environments and by the following Open Source projects: * Asterisk-IM A plugin for the Jive Messenger XMPP (jabber) server. It provides integrated presence between your IM client and phone, notification of incoming calls by IM and originate calls from supported IM clients. * Asterisk Desktop Manager (ADM) A desktop application that will allow for automatic on-call volume reduction, one click dial from clipboard, integrated phonebook and more. Asterisk-Java is available under Apache 2.0 license at http://www.asteriskjava.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
Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Rich Adamson wrote: I'd suggest removing the majority of the statements and starting out with very basic definitions, then when things are working reasonably well, add in other statements (like callerid, etc). [trunkgroups] [channels] signalling=fxs_ks group = 2 context=mainmenu channel = 3 Still fails :(. Even after a reboot. Any other suggestions welcome :). JES smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Dialplan help
This might work if you switch it around a little.
http://www.voip-info.org/wiki-Asterisk+cmd+Flash
-Original Message-
From: MZ [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 27, 2005 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialplan help
Thanks Steve,
But this will not work for me because after yourcodehere the line
will
give a confirmation tone (similar to a congestion tone only faster)
then
after flashing or certain period will turn into a busytone and to get
the
dialtone again i need to Flash again before i can dial ${EXTEN}.
On 11/28/05, Steve Totaro [EMAIL PROTECTED] wrote:
How about this?
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
-or-
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN})
I have not seen restrictions set before dialing since local
numbers
would not fall under the restriction but that is what you said.
Usually
you dial the number first and if it is a long distance call,
they
prompt
for a code with a tone or something. The second example shoud
give
you
a two second pause before dialing the rest of the number ().
Thanks,
Steve
thanks steve,
the reason i cannot remove the restriction on the telco line
is
that
an
analog fone is connected to the phone jack of the x101p and
some
visitors
occasionally use the fone and they're supposed to only call
local
toll
free numbers.
Your suggestion of doing the restrictions within asterisk is
good
but
i
have yet to buy an FXS port for the analog fone so i cannot do
so.
and for the extensions connected via softphones to asterisk, i
wanted
asterisk to take care of opening the lock before dialing the
desired
phone
number.
i was thinking of using the SendDTMF but i dont know how to
get
the
zap
line go off hook.
On 11/27/05, Steve Totaro [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
hi,
can anyone please guide me as to how i can implement
this
in
extensions.conf:
my PSTN line normally has its longdistance capability
locked
which
can
be opened by dialing some keys and the PIN.
if i wanted some users to be allowed to call long
distance
using
the
zap
channel, how can i initially offhook then send some
DTMF
then
Flash
twice
before proceeding with the dialing EXTEN.
Do you want asterisk to take care of the DTMF for you?
When
you
say
users, do you mean certain extensions?
Why not cancel your restrictions with the telco and
implement
restrictions within asterisk. You can use authenticate
before
your
dial
statement to prompt for a pin when dialing long distance
or
you
can
create different contexts for phones that you want to
access
long
distance and for phones that you don't
Thanks,
Steve
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Re: [Asterisk-Users] Dialplan help
Yeah, and unlocked ATAs are not available in the market here.
I even had to pay almost twice the cost of my X101p clone for shippingOn 11/28/05, Steve Totaro [EMAIL PROTECTED]
wrote:Wow, what a pain.I would just pickup an FXS and be done with it.
Thanks Steve, But this will not work for me because after yourcodehere the linewill give a confirmation tone (similar to a congestion tone only faster)then
after flashing or certain period will turn into a busytone and to getthe dialtone again i need to Flash again before i can dial ${EXTEN}. On 11/28/05, Steve Totaro
[EMAIL PROTECTED] wrote: How about this? exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN}) -or-
exten = 1234567890,1,Dial(Zap/g1/yourcodehere${EXTEN}) I have not seen restrictions set before dialing since localnumbers would not fall under the restriction but that is what you said.
Usually you dial the number first and if it is a long distance call,they prompt for a code with a tone or something.The second example shoudgive you a two second pause before dialing the rest of the number ().
Thanks, Steve thanks steve, the reason i cannot remove the restriction on the telco line
is that an analog fone is connected to the phone jack of the x101p andsome visitors occasionally use the fone and they're supposed to only call
local toll free numbers. Your suggestion of doing the restrictions within asterisk isgood but i have yet to buy an FXS port for the analog fone so i cannot do
so. and for the extensions connected via softphones to asterisk, i wanted asterisk to take care of opening the lock before dialing the desired
phone number. i was thinking of using the SendDTMF but i dont know how toget the zap line go off hook.
On 11/27/05, Steve Totaro [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi,
can anyone please guide
me as to how i can implementthis in extensions.conf:
my PSTN line normally has
its longdistance capability locked which can
be opened by dialing some
keys and the PIN.
if i wanted some users to
be allowed to call longdistance using the zap
channel, how can i
initially offhook then send someDTMF then Flash twice before proceeding with the dialing EXTEN.
Do you want asterisk to take
care of the DTMF for you?When you say users, do you mean certain extensions?
Why not cancel your
restrictions with the telco and implement
restrictions within
asterisk.You can use authenticate before your dial
statement to prompt for a pin
when dialing long distanceor you can
create different contexts for
phones that you want toaccess long distance and for phones that you don't Thanks, Steve___
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[Asterisk-Users] chan_bluetooth with Plantronics Heaset (some good stuff)
Hi: I have made some success in attaching my plantronics M2500 headset to asterisk. I can get the headset to ring, however there is no Audio I have tried using the connection below by having the api manager dial the headset, then when answered, dial an internal extension (2000). I get no audio on the channel. I also have an internal extension for MOH, and that doesn't produce any audio as well... I have also tried to connect to a toll free information service, 18005551212 and that doesn't produce any audio. HS connecting: Nov 27 10:29:51 NOTICE[7862]: /usr/src/chan_bluetooth/chan_bluetooth.c:2226 try_connect: Initialised bluetooth link to device HS-1 [HS] HS-1 AT+BRSF=24 [HS] HS-1 +BRSF: 23 [HS] HS-1 OK [HS] HS-1 AT+CIND=? [HS] HS-1 +CIND: (service,(0,1)),(call,(0,1)),(callsetup,(0-4)) [HS] HS-1 OK [HS] HS-1 AT+CIND? [HS] HS-1 +CIND: 1,0,0 [HS] HS-1 OK [HS] HS-1 AT+CMER=3, 0, 0, 1 [HS] HS-1 OK [HS] HS-1 AT+VGS=07 [HS] HS-1 OK HS Getting a call: [HS] HS-1 +CIEV: 3,1 [HS] HS-1 RING [HS] HS-1 ATA [HS] HS-1 +CIEV: 2,1 [HS] HS-1 +CIEV: 3,0 [HS] HS-1 OK Channel BLT/HS-1 was answered. Nov 27 10:23:37 WARNING[7667]: /usr/src/chan_bluetooth/chan_bluetooth.c:621 sco_thread: SCO thread started on fd 30, pid 7622 -- Executing Dial(BLT/HS-1, SIP/2000|30|rtT) in new stack -- SIP Seeding peer from astdb: '2000' at [EMAIL PROTECTED]:5060 for 3600 -- Called 2000 -- SIP/2000-e947 is ringing -- SIP/2000-e947 answered BLT/HS-1 Nov 27 10:23:41 WARNING[7668]: /usr/src/chan_bluetooth/chan_bluetooth.c:1139 blt_indicate: Don't know how to condition -1 The warning above: i dont understand... Can someone please help...?? my bluetooth.conf file is: [general] ; Channel we listen on as a HS (Headset) rfchannel_hs = 2 ; Channel we listen on as an AG (AudioGateway) rfchannel_ag = 3 ; hci interface to use (number - e.g '0') interface = 0 [00:03:89:5E:0F:2A] name= HS-1 type= HS ; ; ; RFCOMM channel to connect to. For a HandsSet: ;sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111E ; or,for an AudioGateway (Phone): ;sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111F ; ; Find the 'channel' value under RFCOMM. ; channel = 2 ; Automatically conenct? autoconnect = yes The channel that I got back was channel 2 from the rfcomm search command Thanks. ___ --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: Re: [Asterisk-Users] Zaptel errors on Debian
Well, thanks, it might be great your package yet I would like to know how to adapt. I wouldn't like to rewrite Debian neither Asterisk but is somebody able to advice how you define modules in zconfig.h or whatever ? Any tip ? Geo Hi On Sun, Nov 27, 2005 at 03:40:39PM -0800, Geotrix wrote: Hello, I am trying to install zaptel wcfxo with X101.P board on Debian sarge without success. (previously compiled and worked OK on Redhat kernel) On boot asterisk I have [chan_zap.so] = (Zapata Telephony w/PRI) !!!??? but I have no PRI, I have fxo, x1001P digium . As a matter of fact when you type version on Asterissk-1.2.0 : Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] ... or something. Do you use the debs or build your own? apt-get install zaptel-source and use 'm-a a-i zaptel' to get a zaptel modules package. Anyone up there experienced that problem with Asterisk on Debian ? Any trick ? Eventually, how you define module type wcfxo in zconfig.h when you do not have PRI interface ? Begin: self promotion Hmmm. why bother? http://xorcom.com/rapid will give you a fully-functioning Debian Sarge system with Zaptel pre-compiled. End: self promotion -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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] Would DECT cordless phones work with Asterisk and VOIP?
On Sun, November 27, 2005 17:21, Michael Graves said: The DECT implementation we see in the US are I presume deployed with normal analog line connections, unlike in Europe where many phones are ISDN based. Thus you may not be able to avoid the D A D conversion. ISDN cards that work with US standard ISDN lines are essentially unheard of. Many of us who have been dissatisfied with small FXO adapters would LOVE to order up ISDN lines and skip the whole FXO problem altogether. There are several/many ISDN cards that work with Asterisk in Europe. I've yet to find even one that would work with an SBC ISDN drop. I'd welcome someone proving me wrong on this. Michael Graves I don't know about all of that, but you can *always* get a European ISDN DECT system with a HFC-PCI card to take care of the wireless extensions part... That at least gets rid of the D-A-D part going in to *... Especially when using a VoIP provider like GoIAX, FWD or VoipBuster, you will remain all digital! I'm sure there are webstores that'll ship Euro-ISDN phones and cards to the US! ;-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] IAXmodem fax polling
Jean-Denis Girard wrote: Hi list, With txfax it is straightforward. With iaxmodem, the only way I found is to send both calls to a meetme room: the customer hears a message/music while the fax is processed by Hylafax, and then iaxmodem is bridged to the caller's channel. It seems to work in my very limited testing, but I'm a bit worried about degradation going through the meetme pseudo-channel, which would result in less reliability. What do you think about reliability? Is there a better way? Thanks, Humm, no reply... :( Did I miss something obvious or is there no solution ? I found app_bridge on the bug tracker (http://bugs.digium.com/view.php?id=5841), which seems what I need, but doesn't seem ready for production yet. Comments please :) Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --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] Failover with 1.2 Dial applications
Previously I would have used n+101 to effect a dialplan failover scheme. With the new 1.2 applications, is there a better way to provide fail over? -- Chris Mason NetConcepts ___ --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] Asterisk cdr mysql
Hi all, Did anyone installed asterisk-addons successfull? Becuase i am getting some error in installation. Error: cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:292: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 rm app_saycountpl.o Please help me how i can load this mysql cdr module? -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --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] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
From: James B. MacLean [EMAIL PROTECTED] Asterisk*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 Wildcard TDM400P REV I Board 2 OK 0 0 0 ---End of Original Message- The above does indicate a problem. The Rev E/F card is known to have issues, and most of the issues revolved around unusual failures after a week or so. But there have been several other changes leading up to the Rev I card (the latest is Rev J with only minor changes since Rev I). I don't know of anyone that has attempted to mix to Rev's of the TDM card in a system, so unknown whether that might be an issue or not. I'd contact digium support and have that Rev E/F card rma'ed under warranty. (All TDM cards are still under warranty.) ___ --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] Zaptel errors on Debian
On 11/27/05, Geotrix [EMAIL PROTECTED] wrote: Hello, I am trying to install zaptel wcfxo with X101.P board on Debian sarge without success. (previously compiled and worked OK on Redhat kernel) and in debian ? for compiling it in debian i suggest you to install kernel-headers (tha same version of your running kernel), module-assistant and dpatch (install all with apt or aptitude) and zaptel and zaptel-source, of course :) When i compile it i copy the /boot/config-running.kernel.version to /usr/src/kernel-headers-running.kernel.version/.config and then make cd /usr/src; m-a build zaptel this will generate a .deb in . then just dpkg -i name.of.the.deb.file.generated and reboot. Then you should can modprobe zaptel, etc... On boot asterisk I have [chan_zap.so] = (Zapata Telephony w/PRI) !!!??? but I have no PRI, I have fxo, x1001P digium . As a matter of fact when you type version on Asterissk-1.2.0 : Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] ... or something. Taht your version is that, is OK. Because you are using Debian Stable and installing asterix from those repositories... and in stable thats the version of asterisk aviable Anyone up there experienced that problem with Asterisk on Debian ? Any trick ? Eventually, how you define module type wcfxo in zconfig.h when you do not have PRI interface ? Thx, Geo Thanks, i hope this could help you :) Rodrigo ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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 ___ --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] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Rich Adamson wrote: From: James B. MacLean [EMAIL PROTECTED] Asterisk*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 Wildcard TDM400P REV I Board 2 OK 0 0 0 ---End of Original Message- The above does indicate a problem. The Rev E/F card is known to have issues, and most of the issues revolved around unusual failures after a week or so. But there have been several other changes leading up to the Rev I card (the latest is Rev J with only minor changes since Rev I). I don't know of anyone that has attempted to mix to Rev's of the TDM card in a system, so unknown whether that might be an issue or not. I'd contact digium support and have that Rev E/F card rma'ed under warranty. (All TDM cards are still under warranty.) Thanks for the heads up. More dissappointing is that the E/F card is the newer card purchased. Where can I go to see when certain revisions were released? Surprising that the newer card just purchased (to me) is the older rev :(. Next I'll try with just one card, but that will be another day as the machine is not local. thanks again, JES smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Problem about outgoing calls with verizon.
I have met a very strange problem. We use a Asterisk PBX connected with two
rollover
PSTN phone line
provided by Verizon by Digium TDM cards. The incoming calls are always OK.
But when I
make outgoing calls,
sometimes it works, sometimes it just get a busy tone, doesn't work at all.
The
strange thing is that when I use the
same server with SBC lines perviously, it worked fine.
What is the problem here? This drives me crazy.
Asterisk is dialing the number before the central office is ready to
accept it. Just add a w in the dial string like this:
exten = _9XXX,1,Dial(Zap/4/w${EXTEN})
Each w is worth about a 1/4 to 1/2 second delay. If one doesn't fix
the problem, try two or three.
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Re: [Asterisk-Users] Narrowing RTP port range
I'm trying to lock down my asterisk install as much as possible and I keep reading about people saying 'you can narrow the range of ports in rtp.con' (by default it's from 1 to 2 I think). My question is this - how much can I narrow it down? Can I narrow it to 10 ports, or can the ports not be reused for additional conversations? I guess what I'm asking is - does the number of ports in the range have anything to do with the number of simultaneous connections or anything like that? Yes, you can narrow it down. One port will be required for each leg of a call. So if sip/123 called sip/345, that's two ports. I'd suspect that some ports are used for other purposes besides just a conversation, so adding more is certainly more in your best interest then cutting them short. ___ --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] OH323 channel in asterisk 1.2 ...Ouch ... error while writing audio data!
/var/log/asterisk/full.1 output: Nov 26 21:25:39 VERBOSE[14215] logger.c: [chan_oh323.so]Nov 26 21:25:39 WARNING[14215] loader.c: /usr/lib/asterisk/modules/chan_oh3 23.so: undefined symbol: _ZNK8PChannel7IsClassEPKc Nov 26 21:25:39 WARNING[14215] loader.c: Loading module chan_oh323.so failed! thanks rafael On 11/27/05, Adam Rybak [EMAIL PROTECTED] wrote: You should have more info in full log messages, look to this file and sendoutput.AdamCytowanie Rafael R. GV [EMAIL PROTECTED]: Hello I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk 1.2libraries, must be oh323-0.7.3, now I have compiled this version but when reload asterisk i have this error: [chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe Any idea??? -- rrgv___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- rrgv ___ --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] OH323 channel in asterisk 1.2 ...Ouch ... error while writing audio data!
It looks like compiling oh323 with wrong version of headers or wrong version of open323/pwlib. Are you completly sure that you deleted old headers and libraries when upgraded asterisk to new version? Adam Rybak Cytowanie Rafael R. GV [EMAIL PROTECTED]: /var/log/asterisk/full.1 output: Nov 26 21:25:39 VERBOSE[14215] logger.c: [chan_oh323.so]Nov 26 21:25:39 WARNING[14215] loader.c: /usr/lib/asterisk/modules/chan_oh3 23.so: undefined symbol: _ZNK8PChannel7IsClassEPKc Nov 26 21:25:39 WARNING[14215] loader.c: Loading module chan_oh323.so failed! thanks rafael On 11/27/05, Adam Rybak [EMAIL PROTECTED] wrote: You should have more info in full log messages, look to this file and send output. Adam Cytowanie Rafael R. GV [EMAIL PROTECTED]: Hello I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk 1.2libraries, must be oh323-0.7.3, now I have compiled this version but when reload asterisk i have this error: [chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe Any idea??? -- rrgv ___ --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 -- rrgv Pozdrawiam, Adam Rybak ___ --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] Eicon Diva Server query
Hey guys, I just wanted to thank everyone (particularly Armin) for the assistance with my new Eicon Diva. I installed it into a new Asterisk 1.2 box this weekend and the installation was simple and painless, thanks to all your input. If anyone else is considering one of these cards, let me just say that the installation on CentOS 4.2 (RHEL4 clone) was incredibly simple: The source code RPM installed and compiled with no problems, the ./Config script found my card and set everything up for me. All up, loading CentOS 4.2, downloading ~200mb of updates (I only had the 4.1 CDs, so yum did a LOT of upgradeing), the Eicon Diva drivers, Asterisk 1.2 (with chan_capi-cm) and Asterisk Management Portal only took 3 hours from start to fully configured finish. :) cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / Walter Turnbull Bldg T: +61 (0) 2 6233 0607 44 Sydney Ave, F: +61 (0) 2 6233 0696 Forrest, W: http://www.squiz.net/ ACT 2603 . Open Source - Own it - Squiz.net ./ ___ --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] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
On Sun, November 27, 2005 11:12, Kristof Hardy said: Francesco Peeters wrote: Reshuffled the cards in my machine (actually inverted the order of the PCI cards) and problem solved! Indeed, if the bios is is not working 'with' you, but more against you, then shuffling could solve the problem, glad it did :-) It appears the bottom 2 slots of the MoBo share an IRQ line, period... Now the cards have different IRQs, and both work, *with* the sync connection in place (i.e. the NT running of the TE clock) Brilliant! Now to play with the dialplan and integrate a zap-channel phone (or actually: 4 phones... G) Excellent, now the 'really' fun part starts out :-) 1.2 has 'soo' many possibilities :) cheers! I have the 4030 running now, and it works fine, however when I restart asterisk (amportal stop/start) I get an error... Not sure yet what the exact cause is, but I do know it has something to do with the AMP file 'zapata_additional.conf', as the error goes when I disable the include statement in 'zapata.conf' (but then the ZAP extensions don't work!) The error I get is: Nov 27 21:05:52 WARNING[15764] chan_zap.c: Ignoring echocancelwhenbridge Nov 27 21:05:52 ERROR[15764] chan_zap.c: Syntax error parsing 'g11/2010' at 'g11/2010' Nov 27 21:05:52 WARNING[15764] chan_zap.c: Reload of chan_zap.so is unsuccessful! I suspect it is because it tries to use the channel group (g11 is the group for channels 1 and 2 on the NT card) before chan_zap has completed loading. As a workaround I have tried the following: - comment out the zapata_additional line in zapata.conf - start asterisk (amportal start) - uncomment the zapata_additional line - reload the config Which works, but of course is not a viable solution for the long term. The zapata_additional, as defined by AMP: ;;[2010] signalling=fxo_ks record_out=On-Demand record_in=On-Demand [EMAIL PROTECTED] echotraining=800 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=no callerid=Basis 2010 busydetect=no busycount=7 channel=g11/2010 ;;[2011] signalling=fxo_ks record_out=On-Demand record_in=On-Demand [EMAIL PROTECTED] echotraining=800 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=no callerid=Woonkamer 2011 busydetect=no busycount=7 channel=g11/2011 ;;[2012] signalling=fxo_ks record_out=On-Demand record_in=On-Demand [EMAIL PROTECTED] echotraining=800 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=no callerid=Slaapkamer 2012 busydetect=no busycount=7 channel=g11/2012 ;;[2013] signalling=fxo_ks record_out=On-Demand record_in=On-Demand [EMAIL PROTECTED] echotraining=800 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=no callerid=Miriam 2013 busydetect=no busycount=7 channel=g11/2013 ;;[2014] signalling=fxo_ks record_out=On-Demand record_in=On-Demand [EMAIL PROTECTED] echotraining=800 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=no callerid=Francesco 2014 busydetect=no busycount=7 channel=g11/2014 ;;[2020] signalling=fxo_ks record_out=On-Demand record_in=On-Demand [EMAIL PROTECTED] echotraining=800 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=no callerid=All 2020 busydetect=no busycount=7 channel=g11/2020 To be continued... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] OH323 channel in asterisk 1.2 ...Ouch ...error while writing audio data!
I had similar problems... I downloaded the OH323 package for [EMAIL PROTECTED] and installed it (https://sourceforge.net/project/showfiles.php?group_id=123387)... Seems to work much better. It includes: gnugk 2.2.1 pwlib 1.6 open h323 1.13 I have setup my Cisco uBR924 with H.323 and I can place outbound calls. The only issue I have is sending inbound calls to the Cisco device. Any thoughts??? Daryl - Original Message - From: Adam Rybak [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 27, 2005 4:21 PM Subject: Re: [Asterisk-Users] OH323 channel in asterisk 1.2 ...Ouch ...error while writing audio data! It looks like compiling oh323 with wrong version of headers or wrong version of open323/pwlib. Are you completly sure that you deleted old headers and libraries when upgraded asterisk to new version? Adam Rybak Cytowanie Rafael R. GV [EMAIL PROTECTED]: /var/log/asterisk/full.1 output: Nov 26 21:25:39 VERBOSE[14215] logger.c: [chan_oh323.so]Nov 26 21:25:39 WARNING[14215] loader.c: /usr/lib/asterisk/modules/chan_oh3 23.so: undefined symbol: _ZNK8PChannel7IsClassEPKc Nov 26 21:25:39 WARNING[14215] loader.c: Loading module chan_oh323.so failed! thanks rafael On 11/27/05, Adam Rybak [EMAIL PROTECTED] wrote: You should have more info in full log messages, look to this file and send output. Adam Cytowanie Rafael R. GV [EMAIL PROTECTED]: Hello I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk 1.2libraries, must be oh323-0.7.3, now I have compiled this version but when reload asterisk i have this error: [chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe Any idea??? -- rrgv ___ --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 -- rrgv Pozdrawiam, Adam Rybak ___ --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
Re: [Asterisk-Users] A rather big setup.
Spreading * servers across multiple floors sounds like a bad idea since it'd increase maintenance time. With your projected call volume there's no way you can reliably run g729 or any other CPU hog of a codec on a single box. For this kind of a setup you'd need 2-3 boxes and a SER/heartbeat box to handle registration and call distribution. I would also isolate CDR recording to a separate box running a database like Postgres (IMHO better choice due to WAL) or MySQL. ScriptHead On 11/27/05, Simone Cittadini [EMAIL PROTECTED] wrote: Vedran Dakic ha scritto:I can only guess that I should have the ability to deliver a solution thatcan do some 100/500 simultaneously. The only question is how powerful shouldbe a machine (or machines) that could do around 100/500 simultaneously. And, just for the sake of knowing, what should the setup be alike if it was240/1000 simultaneously?My suggestion is to buy the E1 cards first of all and put them in a testserver, equipped with asterisk and all the relevant agi / db connections / moh etc..Then loop the card with a crossover cable and run some test script togenerate themedium and upper bound call flows.That should give you an idea of your cpu/ram requirements. In the second case there's no need for a cluster, a good server will do,(obviously a second server for backup is a good idea ). I'm assuming youcan use a/ulaw to transmit the data, if bandwidth is a problem and you must compress cpu usage becomes a boottleneck to keep in mind.A/ulaw? I saw some reports that G.729 uses very little bandwidth and hasa quality part granted (audio quality). It's not a question of hardware and/or CPU power, I have two dual Opteron configurations and could installsome more, it's just the question of that setup running with quality audioand no unwanted events.G729 has a very good quality -considered the bandwidth used-, but if your customers are used to conventional telephony they will no doubtnotice the difference, so go with G711 (probably alaw, since you use E1I suppose you are in europe)Anyway if bandwidth is a problem consider ilbc / speex which are free and have good audio qualities also.Lastly a lot of the quality comes from a well configured phone, tweakwith volumes and timeouts.I presume that I should have all of the phones using the same codec (so, no transcoding), and preferrably the same VoIP protocol. I have a choicethere so everything's possible. Let's say - IP10s has H.323, SIP and MGCPfirmwares, although I'd like to leave H.323 out of the story. Yes, leaving H323 out of the story is a good way to start the project :)___--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] Asterisk 1.2 and Athlon64 platforms
Good evening all, Are there any folks out there running Asterisk on Athlon64 platforms with 64-bit operating systems? I have a couple of new asterisk servers to build up this week and I'm debating whether to order some Athlon64 CPUs and boards for them. I usually install Gentoo onto the boxes, so any experiences (or pitfalls) that folks running Gentoo + Asterisk on Athlon64 platforms would be especially helpful, particularly regarding compatibility with asterisk hardware (HFC-based ISDN cards and Digium TDM400 cards specifically). Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
On Sun, November 27, 2005 22:59, Francesco Peeters said: On Sun, November 27, 2005 11:12, Kristof Hardy said: Francesco Peeters wrote: Reshuffled the cards in my machine (actually inverted the order of the PCI cards) and problem solved! Indeed, if the bios is is not working 'with' you, but more against you, then shuffling could solve the problem, glad it did :-) It appears the bottom 2 slots of the MoBo share an IRQ line, period... Now the cards have different IRQs, and both work, *with* the sync connection in place (i.e. the NT running of the TE clock) Brilliant! Now to play with the dialplan and integrate a zap-channel phone (or actually: 4 phones... G) Excellent, now the 'really' fun part starts out :-) 1.2 has 'soo' many possibilities :) cheers! I have the 4030 running now, and it works fine, however when I restart asterisk (amportal stop/start) I get an error... Not sure yet what the exact cause is, but I do know it has something to do with the AMP file 'zapata_additional.conf', as the error goes when I disable the include statement in 'zapata.conf' (but then the ZAP extensions don't work!) The error I get is: Nov 27 21:05:52 WARNING[15764] chan_zap.c: Ignoring echocancelwhenbridge Nov 27 21:05:52 ERROR[15764] chan_zap.c: Syntax error parsing 'g11/2010' at 'g11/2010' Nov 27 21:05:52 WARNING[15764] chan_zap.c: Reload of chan_zap.so is unsuccessful! I suspect it is because it tries to use the channel group (g11 is the group for channels 1 and 2 on the NT card) before chan_zap has completed loading. As a workaround I have tried the following: - comment out the zapata_additional line in zapata.conf - start asterisk (amportal start) - uncomment the zapata_additional line - reload the config Which works, but of course is not a viable solution for the long term. The zapata_additional, as defined by AMP: SNIP Well, I replaced the g11/201* entries with 1-2/201* and all seems to work, but I really would like to use the groups, so I do not have to redo all extensions when I change the order or grouping of the cards... I'll have to check what exactly is happening there, and how to circumvent it... Bedtime now! :-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] A rather big setup.
On Sun, 2005-11-27 at 12:02 +0100, Vedran Dakic wrote: Hmm, maybe I'm missing something here. So, just to be sure... I was thinking about having a separate Asterisk server/cluster in the -1 floor server room where all of the telco/other wires come in (with that 240 lines via 8 E1 wires), and one asterisk server per floor connected to Asterisk server/cluster in the basement. I don't understand what did you mean by this 30 lines from each floor :) Confused a bit here Couldn't I have these per-floor Asterisk servers connected directly via Ethernet/IAX to the -1 floor server room, and have those servers/cluster/whatever manage the calls? I wasn't thinking about installing one asterisk server per floor with E1 card inside. I was thinking about connecting all of those servers to the central server with all of the E1 lines inside. Isn't that possible? Cheers, Vedran. I think there is more to consider. One or two fat machines in the basement forr connecting to the PSTN is very fine. But are all the people allready using voip handsets, or old fashioned analoge handsets? If so, you need quite a large number of channelbanks. You speak of 300/1500 concurrent phone calls? If so how many handsets are you considering? Is the lan capable of handling this load? Is the lan 100% dedicated for voip, or are there a bunch of servers/workstations also using this lan? Interesting project Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.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
[Asterisk-Users] launching 2 scripts
Hi: i tried to lauch the callback.agi script and astcc.agi script together but i failed to do that ,i tried this at extensions.conf: [incoming] exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,DeadAGI(callback.agi) exten = s,4,DeadAGI(astcc.agi) exten = s,5,Hangup i tried to make astcc.agi launch when the call answered when it callback but i failed. __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. 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] launching 2 scripts
I'm not exactly sure what you're trying to do. I don't know which call back script you are using, but you should be able to set which context and extension you want the call connected to. I do that using a callback script that I found on the internet somewhere. I did some work to it and it is available @ www.aleph-com.net/astpp. This is the way I run that one: [callback] exten = 1,1,AGI(callback.agi,ACCOUNTCODE,99,,9959,meetme,enhanced-outgoing) 99 can be a CID number. If that number dials in it gets connected to instead of 9959 meetme is the context to throw the call into when it's connected enhanced-outgoing is where I send the outgoing calls through. I use the local channel. Make sense? Darren Wiebe [EMAIL PROTECTED] chawki hammoud wrote: Hi: i tried to lauch the callback.agi script and astcc.agi script together but i failed to do that ,i tried this at extensions.conf: [incoming] exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,DeadAGI(callback.agi) exten = s,4,DeadAGI(astcc.agi) exten = s,5,Hangup i tried to make astcc.agi launch when the call answered when it callback but i failed. __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. 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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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_bluetooth background scanner
Hi: After a bunch of trial and error: i was able to put together a little package for using bluetooth headsets with my asterisk system: Overview: allow headsets to place calls out using asterisk channels Allow other scripts to see the status of headsets like a follow me type of system Can be downloaded at: http://www.itsngroup.com/software/asterisk/downloads/ Have a good day.. Ben ___ --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] New Asterisk user - Dumb Questions
Hi All, This is my first posting and if am asking dumb questions please let me apologize. I'm in no way a telephony or pbx expert. I have tried googling for answers but can't seem to find the answers I need. Probably because I'm not using the right words. I have installed [EMAIL PROTECTED] 2.0 which uses Asterisk 1.2. Everything appears to be running great with two exceptions: 1) On outbound dialing it appears that about 50% of the time the leading dial digit is lost by the phone company. I have 8 analog lines connected into two digium TDM400P cards. I think Asterisk is dialing the number before the carrier (SBC) is ready. Is there a way to create a delay in the dialing to allow SBC to be ready to take the dialing? Kind of like the w command in modem dialing. If so, what do I need to put and in which configuration file? 2) Is there a way to increase the volume of received vmail in email? I've tried listening to them on 2 different PCs and a laptop. Even with volume turned up all the way, they are very faint. Thanks for your answers and patience, Mike ___ --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] Does it mean I was blocked by STUN?
Hi all, I have 2 respectively networks, LAN A and LAN B, connected via my wireless links and routers. I have setup an asterisk machine at LAN A. It works fine when i was in LAN A. But, when i was in LAN B, xlite client can get connected to the server. But, it has no sound when i try to make an echo test. Does it mean, i was blocked by STUN? I have a wild search on google, i found that asterisk doesnt really support STUN. What is the workaround to make two network clients enjoy the intercalling via asterisk? IAX please advise... thanks ___ --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] New Asterisk user - Dumb Questions
Mike McMullen wrote: Hi All, This is my first posting and if am asking dumb questions please let me apologize. I'm in no way a telephony or pbx expert. I have tried googling for answers but can't seem to find the answers I need. Probably because I'm not using the right words. Some may chastise you for not using the [EMAIL PROTECTED] list, but your problems are certainly more generic. Not dumb questions at all. I have installed [EMAIL PROTECTED] 2.0 which uses Asterisk 1.2. Everything appears to be running great with two exceptions: 1) On outbound dialing it appears that about 50% of the time the leading dial digit is lost by the phone company. I have 8 analog lines connected into two digium TDM400P cards. I think Asterisk is dialing the number before the carrier (SBC) is ready. Is there a way to create a delay in the dialing to allow SBC to be ready to take the dialing? Kind of like the w command in modem dialing. If so, what do I need to put and in which configuration file? You are absolutely correct. Asterisk does NOT listen for dialtone, and no one seems able or cares to fix that problem IF, and only IF, you are dialing with DTMF, you can insert a series of w into the dial string. Search the list archives for exactly where, then you will have to struggle with [EMAIL PROTECTED] to make the change and keep it from being overwritten. If, on the other hand, one uses pulse dial, then the w in the dialstring will not work. 2) Is there a way to increase the volume of received vmail in email? I've tried listening to them on 2 different PCs and a laptop. Even with volume turned up all the way, they are very faint. Another chronic Asterisk problem. I am not sure there has been a fix to this either. Seems to only be there with the TDM400, or perhaps the TDM400 and the X100 cards.. I believe some have changed the format of the Vmail , either to or away from wav, and gotten better results. Again, search the list archives for answers. Best of luck John Novack Thanks for your answers and patience, Mike ___ --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
[Asterisk-Users] Script to update externip for [EMAIL PROTECTED]/AMP [was Re: SIP Extension behind NAT, Asterisk on NAT (DMZ)]
On Nov 26, 2005, at 3:06 PM, Manny A. Wise wrote: [snip] The problem is not updating the FQDN name in dyndns.org..that part is working great...the problem now is..how to get the IP change into the sip_nat.conf... but I am sure has to be a way... :) How about this? You have to add back in the first shebang line that defines /bin/sh as the program to use to run the script. BEGIN SCRIPT- # Script to update Asterisk's externip= setting with the current # IP address. Writes changes to sip_nat.conf for [EMAIL PROTECTED]/AMP # This script is only useful if you have a dynamic IP Address and # are using NAT. # define where to write temporary files Tmp=/tmp/externipupdate$$.txt # define the hostname to lookup host=myhost.mydomain.dom # Use dig to get our current IP address (assuming that it has been # properly updated via DynDNS client or otherwise. Set the variable # ip_address to the value of the IP address. ip_address=`dig $host +short` # Write the new settings to a temporary file and then overwrite the # exisiting /etc/asterisk/sip_nat.conf file with the temporary file # using the mv command. echo nat=yes $Tmp echo externip=$ip_address $Tmp # Change the following line to reflect your local network. Add multiple # localnet= lines if you have more than one local network. echo localnet=10.0.0.0/255.255.255.0 $Tmp mv $Tmp /etc/asterisk/sip_nat.conf # Tell Asterisk to reload SIP to make the changes take effect /usr/sbin/asterisk -rx sip reload --END SCRIPT That ought to do the trick. I tested it with my [EMAIL PROTECTED] config. I will leave the task of finding a way to automatically run this program when needed as an exercise to the reader. (cron would work if it changes at regular time intervals, I suppose) Keep in mind that there will be a delay between when the address changes to when it is updated in DynDNS and then another delay as the change propagates throughout the DNS system. Lastly, depending on how you call the script, there will be a delay between when it propagates through DNS and when this script is run. Maybe there is a way to use the DynDNS client to get the new IP address and write it to sip_nat.conf at the same time it updates the DynDNS service? Are there DynDNS clients that allow you to run an external program every time the IP changes? Kind of like Comedian mail's externnotify parameter? Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --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] Does it mean I was blocked by STUN?
Nat=yes may help -Original Message- From: Hiu Yen Onn [mailto:[EMAIL PROTECTED] Sent: Sunday, November 27, 2005 8:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Does it mean I was blocked by STUN? Hi all, I have 2 respectively networks, LAN A and LAN B, connected via my wireless links and routers. I have setup an asterisk machine at LAN A. It works fine when i was in LAN A. But, when i was in LAN B, xlite client can get connected to the server. But, it has no sound when i try to make an echo test. Does it mean, i was blocked by STUN? I have a wild search on google, i found that asterisk doesnt really support STUN. What is the workaround to make two network clients enjoy the intercalling via asterisk? IAX please advise... thanks ___ --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
Re: [Asterisk-Users] Re: oh323 channel disappears
[EMAIL PROTECTED] wrote: cd /root wget http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/pwlib-Mimas_patch2-src-tar.gz wget http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/openh323-Mimas_patch2-src-tar.gz cd /usr/src wget http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/download/asterisk-oh323-0.6.7.tar.gz cd /root tar zxvf pwlib-Mimas_patch2-src-tar.gz tar zxvf openh323-Mimas_patch2-src-tar.gz mv pwlib_Mimas_patch2 pwlib mv openh323_Mimas_patch2 openh323 cd /usr/src tar zxvf asterisk-oh323-0.6.7.tar.gz PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH Don't put libraries in /root. If you build from source, then put it in a directory below /usr/local. the only thing I am absolutely not hayy to did was that chmod 777 /root; Don't do that or set a directory to 777 unless you want anyone to be able to write there. In this case, you definitely don't want that. I think that it should be not necessary at all, I did it becouse asterisk run as asterisk user, and peraphs i thought some problems aboutr accessing pwlib or oh323; Try running asterisk as root. I have been told asterisk has trouble running under a non-root user. I tried to reboot a box WITHOUT exiting from asterisk, and the running conversetion (with more then 2000 billsec) was not recorded in the cdr Make sure you don't do this. Asterisk keeps state, and you need to save that state before you shutdown the system. HTH, Mike ___ --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] A rather big setup.
Hello, Those people currently aren't using any kind of phones, but the investment company that has this building in the works wants to deliver everything for them so they just have to - move in and do business. What worries me is the fact that when you have 100-200 offices - they're used to having 2-3 lines only for them - one for fax, two for voice, etc. So, in a way, having in mind around 200-300 outbound calls at peak time is pretty much normal. Also, when you think of the number of phones - it would only be normal to assume for people to have up to 1000 internal phone conversations peak (the less transcoding - the better, of course). I have a freedom of making whatever I want, so I can have a separate LAN for VoIP purposes only - a bunch of dedicated patch panels, VLANs on Cisco switches, or whatever. I'm just considering this setup way before it has to go online because of the price of traditional PBX for this kind of setup which can only make you hurl. And you know how much potential upgrades cost for a setup like this - a traditional PBX can be a nightmare :( Cheers, Vedran. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet Sent: Monday, November 28, 2005 12:08 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] A rather big setup. I think there is more to consider. One or two fat machines in the basement forr connecting to the PSTN is very fine. But are all the people allready using voip handsets, or old fashioned analoge handsets? If so, you need quite a large number of channelbanks. You speak of 300/1500 concurrent phone calls? If so how many handsets are you considering? Is the lan capable of handling this load? Is the lan 100% dedicated for voip, or are there a bunch of servers/workstations also using this lan? Interesting project Hans ___ --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] A rather big setup.
I'm very aware of the maintenance issue, but it would also give me a great deal of freedom for some various other things as well. This of course depends on the Asterisk architecture, which I'm not completely aware of - in detail anyway. How does Asterisk handle this kind of setup with one-two/cluster central server(s) and a bunch of other servers connected with IAX(2)? If you have local calls, do they go directly from phone to phone, do they go from phone to per-floor-Asterisk server, or they have to be interconnected via the main Asterisk server(s)/cluster? I mean, there's little point of doing this kind of setup with dedicated Asterisk servers on each floor if you don't get your central server/cluster free of some work - at least of internal calls. Also, it kills scalability, which is always an issue. Anyone? Cheers, Vedran. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Script Head Sent: Sunday, November 27, 2005 11:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A rather big setup. Spreading * servers across multiple floors sounds like a bad idea since it'd increase maintenance time. With your projected call volume there's no way you can reliably run g729 or any other CPU hog of a codec on a single box. For this kind of a setup you'd need 2-3 boxes and a SER/heartbeat box to handle registration and call distribution. I would also isolate CDR recording to a separate box running a database like Postgres (IMHO better choice due to WAL) or MySQL. ScriptHead ___ --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] CPE does not support Call Waiting Caller*ID?
I'm having a strange problem that I can't quite explain. I feel like I'm missing something simple but I can't quite find it. I'm not getting call-waiting CID whenever the incoming call is delivered over IAX. However, when the same caller, coming in over IAX, hits an empty ZAP channel, the channel rings and the CID *is* delivered as expected. I hear the CID bleep-boop, but the display indicates nothing and asterisk spits out CPE does not support Call Waiting Caller*ID. If I reverse this and have the initial call come in via IAX, ringing the ZAP channel, and the second call coming in via ZAP or SIP, the initial call shows the correct CID, I answer the phone, then hear the CID bleep-boop, the display indicates the correct CID, and asterisk spits out CPE supports Call Waiting Caller*ID. Sending Desk Phone/ 0202. I can't for the life of me figure out what the difference is. It seems that if it can do one it can do the other, so I must have some configuration missing somewhere. Inside zapata.conf I have: usecallingpres=yes usecallerid=yes cidstart=ring cidsignalling=bell callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=no callreturn=no This is on Asterisk-1.2 out of CVS, fully up-to-date. I have two ZAP channels, 1 coming in from the telco and 2 going out to an analog phone. I also have a SIP desk phone (which always shows the right CID). Anyone have any ideas? ___ --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] beginner questions
Dear List Members,I am trying to setup a small asterisk box. My configure is pretty basic for now. my zaptel.conf is as follows fxoks=1 fxsks=2 loadzone=us defaultzone=uswhen I run # /sbin/ztcfg -vv I get Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured.my zapata.conf is as follows[trunkgroups]; define any trunk groups[channels]; hardware channels; defaultusecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechotraining=yesimmediate=no; define channelscontext=internal ; Uses the [internal] context in extensions.confsignalling=fxo_ks ; Use FXO signalling for an FXS channelchannel = 1 ; Telephone attached to port 1context=incoming ; Incoming calls go to [incoming] in extensions.confsignalling=fxs_ks ; Use FXS signalling for an FXO channelchannel = 2 ; PSTN at tached to port 2I have also configured two soft phones (xten). I can dial from one softphone to the other and vice versa. I am using SIP. I am using O'Reilly Asterisk: The Future of Telephony. Now to the questions I hope I have provided enough information I am using the latest release of Redhat Linux and Asterisk. I have got 1 FXS and 3 FXO channels.1. I do not get any kind of tone in my analogie phone connected to the FXS port. 2. when I run ZTTOOL I see the card but are there any other tools that give more information I do not see any activity going on. 3. when I turn on computer all the lights at the back of digium card are green and are list even though no wire is inserted. Is that normal? 4. when I insert analogue phone line from telco in FXO port what am I suppose to look at in the zttool. I do not see any kind of activity so far. 5. Are there any open source graphical tools that I can also instal to configure, monitor and troubleshoot asterisk. 6. What other books/links can be helpful in learning this interesting software.I thank you all for you help in advance.Regards, Amir Aziz __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] beginner questions
Amir, http://www.voip-info.org ;) - Original Message - From: Amir Aziz To: asterisk-users@lists.digium.com Sent: Monday, November 28, 2005 1:31 AM Subject: [Asterisk-Users] beginner questions Dear List Members, I am trying to setup a small asterisk box. My configure is pretty basic for now. my zaptel.conf is as follows fxoks=1 fxsks=2 loadzone=us defaultzone=us when I run # /sbin/ztcfg -vv I get Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. my zapata.conf is as follows [trunkgroups]; define any trunk groups[channels]; hardware channels; defaultusecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechotraining=yesimmediate=no; define channelscontext=internal ; Uses the [internal] context in extensions.confsignalling=fxo_ks ; Use FXO signalling for an FXS channelchannel = 1 ; Telephone attached to port 1context=incoming ; Incoming calls go to [incoming] in extensions.confsignalling=fxs_ks ; Use FXS signalling for an FXO channelchannel = 2 ; PSTN at tached to port 2 I have also configured two soft phones (xten). I can dial from one softphone to the other and vice versa. I am using SIP. I am using O'Reilly Asterisk: The Future of Telephony. Now to the questions I hope I have provided enough information I am using the latest release of Redhat Linux and Asterisk. I have got 1 FXS and 3 FXO channels. 1. I do not get any kind of tone in my analogie phone connected to the FXS port. 2. when I run ZTTOOL I see the card but are there any other tools that give more information I do not see any activity going on. 3. when I turn on computer all the lights at the back of digium card are green and are list even though no wire is inserted. Is that normal? 4. when I insert analogue phone line from telco in FXO port what am I suppose to look at in the zttool. I do not see any kind of activity so far. 5. Are there any open source graphical tools that I can also instal to configure, monitor and troubleshoot asterisk. 6. What other books/links can be helpful in learning this interesting software. I thank you all for you help in advance. Regards, Amir Aziz __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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 11/24/2005 ___ --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] Small office with all employee's offsite
On Nov 26, 2005, at 4:01 PM, Jason Marshall wrote: I want all calls to come into the Asterisk box in the main office. This is relatively easy, but how you do it depends on where the analog POTS lines are terminated. At the central office or at the employees' remote location? (I assume that they terminate at the remote locations) You're right, I should have been clearer. The way things are now is probably suboptimal, but here it is anyway. We have one phone number, the line for which is terminated in the main office, which is where I'd like the server to be. The two employees, offsite, have seperate lines which terminate in either location. OK, then this is easy. Instal Asterisk in the central location, along with a Sipura SPA-3000. Configure that unit to answer the incoming POTS line and act as a VOIP gateway for Asterisk. Then configure two additional SPA-3000 units, one at each employee's location. Then, configure Asterisk (I recommend [EMAIL PROTECTED] for your setup, BTW) to route the incoming call to the right extension based on time of day, auto-attendant, whatever. The SPA-3000 units at each remote site will also be able to accept the employee's incoming POTS line and pass that call through to the phone they normally use without resorting to sending it to the Asterisk server and back. (It's all in the SPA-3000 setup. What we do, depending on who is on at that time, is forward the main number (which is hooked up to an old portmaster 2 via a modem, so reachable remotely) to whoever should be getting the calls. This is suboptimal for at least two reasons that I can see: 1) We're paying for a phone line which is basically never used -- the call forwarding happens at the telco's switch in the CO, so nothing ever comes in over that line; This will not change, you're still looking at three lines in the scenario I outlined above. (Unless you switch to incoming VOIP, but I do *NOT* recommend that.) 2) There's no way to record the calls, or to have a consistent voicemail prompt, nor is there any way to present the caller with any options if, for instance, the person who has the phone forwarded to him is busy, or has gone missing for whatever reason... Asterisk will indeed solve this problem. [snip] If I put one of these at each of the two remote sites, could I set them up so that the employees' phones would ring whether the call was routed to them via VOIP, OR if I call their current phone number? So if the server dies, or the DSL to the employees' locations dies, we could revert back to the lame way we're handling call routing now -- by just forwarding the main incoming line to one employee's number? Yes, on both counts. The downside of using a SPA-3000 at the remote location to answer the phone, send the incoming call to the asterisk server, and then send it back to the extension at the remote site is that you will use double the bandwidth. using SIP reinvites might help with that, though. If I understand you, this scenario would be to intercept calls to each employee's current telephone number, redirect the call via VOIP into the Asterisk server, and then direct another VOIP call back to the employee's handset. If that's what you mean, that's not what I hope to accomplish. No one knows each employee's actual telephone number. It's all hidden with the call-forwarding of the main number to each employee's number. Given that you have the one incoming line at the central location, you are good to go. Don't worry about the above. I should see if my local bookstore has a copy, to save on shipping (and delays at the border). If no one has it, I may very well take you up on your offer. Do you have a paypal seller's account? Yes! Feel free to make donations as often as you feel necessary... ;-) Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --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] wcfxo loads correclty after issuing twice the command ztcfg -vvvv !!
Bukoka Budoka wrote: Hi to all, when i issue the ztcfg command for the first time i get the message Changing signalling on channel 1 from Unused to FXS Kewlstart. When i issue it for the second time i get the normal message 1 channels configured. Has anyone any ideas of why not to have the normal behavior on the first place (i mean without passing from Unused to FXS Kewlstart and then to 1 channels configured. ??? --- [EMAIL PROTECTED] ~]# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Changing signalling on channel 1 from Unused to FXS Kewlstart [EMAIL PROTECTED] ~]# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My Zaptel is as follows: fxsks=1 loadzone=us defaultzone=us -- My zapata is as follows: [channels] context=outgoing ;switchtype=national signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ;flash=1 usecallerid=no hidecallerid=yes callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=3.0 txgain=-1.0 group=1 callgroup=1 pickupgroup=1 immediate=yes cidsignalling=bell channel = 1 The zaptel driver as well as the wcfxo driver is loaded in rc.local: modprobe zaptel modprobe wcfxo /sbin/ztcfg -v /usr/sbin/asterisk -vvv /var/log/asterisk/status-log thank you all, Budoka. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you run ztcfg first time after a new configuration, it will show first screen, specially when you change signalling configs. Afterward ztcfg will show second, until you make change. :D ___ --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] New Asterisk user - Dumb Questions
Mike McMullen wrote: Hi All, This is my first posting and if am asking dumb questions please let me apologize. I'm in no way a telephony or pbx expert. I have tried googling for answers but can't seem to find the answers I need. Probably because I'm not using the right words. I have installed [EMAIL PROTECTED] 2.0 which uses Asterisk 1.2. Everything appears to be running great with two exceptions: 1) On outbound dialing it appears that about 50% of the time the leading dial digit is lost by the phone company. I have 8 analog lines connected into two digium TDM400P cards. I think Asterisk is dialing the number before the carrier (SBC) is ready. Is there a way to create a delay in the dialing to allow SBC to be ready to take the dialing? Kind of like the w command in modem dialing. If so, what do I need to put and in which configuration file? In AMP, edit your zap trunk(s) and add a 'w' to 'Outbound Dial Prefix'. This will prepend a 'w' onto all outgoing calls on those trunks. I have the same problem as you and this solves it. 2) Is there a way to increase the volume of received vmail in email? I've tried listening to them on 2 different PCs and a laptop. Even with volume turned up all the way, they are very faint. I haven't tried this yet, but in 1.2 there is a new feature to the voicemail application. From the CLI type 'show application voicemail' and you'll see: g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). Cheers, kevin -- Optimacy Communications, LLC http://www.optimacycomm.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] New Asterisk user - Dumb Questions
From: Kevin Hanson [EMAIL PROTECTED] In AMP, edit your zap trunk(s) and add a 'w' to 'Outbound Dial Prefix'. This will prepend a 'w' onto all outgoing calls on those trunks. I have the same problem as you and this solves it. Hi Kevin! It worked like a champ! Thanks, Mike ___ --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] A rather big setup.
On Nov 27, 2005, at 10:22 PM, Vedran Dakic wrote: What worries me is the fact that when you have 100-200 offices - they're used to having 2-3 lines only for them - one for fax, two for voice, etc. So, in a way, having in mind around 200-300 outbound calls at peak time is pretty much normal. Also, when you think of the number of phones - it would only be normal to assume for people to have up to 1000 internal phone conversations peak (the less transcoding - the better, of course). If you're providing them with analog lines that they would plug faxes, phones, etc into, you should use T1/E1 cards and Analog Channel banks. This choice, of course, affects: I have a freedom of making whatever I want, so I can have a separate LAN for VoIP purposes only - a bunch of dedicated patch panels, VLANs on Cisco switches, or whatever. I'm just considering this setup way before it has to go online because of the price of traditional PBX for this kind of setup which can only make you hurl. And you know how much potential upgrades cost for a setup like this - a traditional PBX can be a nightmare :( If you are using analog channel banks instead of ATAs or SIP hardphones, then VLANS, etc are not necessary. Of course, you will then need wiring for however many lines into each office, but then again, that most likely already exists. (thus saving on investment...) Also, if these tenants are not related, then why not run more than one Asterisk server and avoid interconnecting them? Sure, you'll have multiple systems to maintain, but they will be smaller, less complex systems. Also, since each company is unrelated, there is little benefit to having them all on the same server (no need to dial between offices, etc) Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --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] New Asterisk user - Dumb Questions
On Nov 27, 2005, at 8:52 PM, John Novack wrote: Mike McMullen wrote: [snip] I have installed [EMAIL PROTECTED] 2.0 which uses Asterisk 1.2. Everything appears to be running great with two exceptions: 1) On outbound dialing it appears that about 50% of the time the leading dial digit is lost by the phone company. I have 8 analog lines connected into two digium TDM400P cards. I think Asterisk is dialing the number before the carrier (SBC) is ready. Is there a way to create a delay in the dialing to allow SBC to be ready to take the dialing? Kind of like the w command in modem dialing. If so, what do I need to put and in which configuration file? You are absolutely correct. Asterisk does NOT listen for dialtone, and no one seems able or cares to fix that problem IF, and only IF, you are dialing with DTMF, you can insert a series of w into the dial string. Search the list archives for exactly where, then you will have to struggle with [EMAIL PROTECTED] to make the change and keep it from being overwritten. If, on the other hand, one uses pulse dial, then the w in the dialstring will not work. As someone else mentioned, open AMP, configure your ZAP Trunk and put the 'w' in there. Works like a charm. 2) Is there a way to increase the volume of received vmail in email? I've tried listening to them on 2 different PCs and a laptop. Even with volume turned up all the way, they are very faint. Another chronic Asterisk problem. I am not sure there has been a fix to this either. Seems to only be there with the TDM400, or perhaps the TDM400 and the X100 cards.. You might want to consider looking at your ZAP txgain and rxgain settings. Be careful, because messing with them and setting them improperly can result in echo problems. However, it is possible that the ZAP channel rxgain needs to be boosted. You don't notice on normal telephone calls because you have your phone's volume control turned up to compensate. The Voicemail app, however, isn't able to compensate in the same way. Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --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] A rather big setup.
if this is a brand new thing you can force the phones on people and then you can to provisioning remotly of for instance Grandstream so they can change the config themself. By forcing a common set of codex you can avoid cpu overhead of translation so you only have to think of teh datashuffle. Bu doing god work at the dialpla you make shure that all the calls thats internal never hit the main pbx'es in the celler and oly use them for outgoing! Best regards jan --On Monday, November 28, 2005 04:22:09 AM +0100 Vedran Dakic [EMAIL PROTECTED] wrote: Hello, Those people currently aren't using any kind of phones, but the investment company that has this building in the works wants to deliver everything for them so they just have to - move in and do business. What worries me is the fact that when you have 100-200 offices - they're used to having 2-3 lines only for them - one for fax, two for voice, etc. So, in a way, having in mind around 200-300 outbound calls at peak time is pretty much normal. Also, when you think of the number of phones - it would only be normal to assume for people to have up to 1000 internal phone conversations peak (the less transcoding - the better, of course). I have a freedom of making whatever I want, so I can have a separate LAN for VoIP purposes only - a bunch of dedicated patch panels, VLANs on Cisco switches, or whatever. I'm just considering this setup way before it has to go online because of the price of traditional PBX for this kind of setup which can only make you hurl. And you know how much potential upgrades cost for a setup like this - a traditional PBX can be a nightmare :( Cheers, Vedran. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet Sent: Monday, November 28, 2005 12:08 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] A rather big setup. I think there is more to consider. One or two fat machines in the basement forr connecting to the PSTN is very fine. But are all the people allready using voip handsets, or old fashioned analoge handsets? If so, you need quite a large number of channelbanks. You speak of 300/1500 concurrent phone calls? If so how many handsets are you considering? Is the lan capable of handling this load? Is the lan 100% dedicated for voip, or are there a bunch of servers/workstations also using this lan? Interesting project Hans ___ --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 -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B pgpLA98p9uSVF.pgp Description: PGP signature ___ --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
