[Asterisk-Users] Re: SoftFax question
Thanks Steve and you other guys for your help. I hope the quality of the other stuff I did yesterday is better than my search for 'fax extension' (a bit embarrassed) Freddi On Wed, 12 Nov 2003, Freddi Hansen wrote: Hi, I am looking at using the softfax that Steve Underwood has developed. It's very straight forward when you assign an extension for the fax. A function that several pbx's has is that they listen for the 'faxtone' for 5 seconds after 'answer' in the menu where you can enter your local extension number, it's normally done in parallel with the DTMF detection. I think that snip You want a fax extension: exten=fax,1,Blah() A google for 'fax extension' turns up the announcement of this feature here: http://lists.digium.com/pipermail/asterisk-users/2002-October/005414.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Soft fax (rxfax) 8 byte output problem resolved?
Hi, On Wed, 12 Nov 2003 at 13:53, David Carr wrote: I have read all the mailing list posts regarding rxfax receiving a fax and outputing an 8 byte tif file (tif header only). This is the problem I can't seem to get past. I think this is not necessarily a single problem. As I understood Steve the 8 byte tiff file just means, that there was no fax received, which can have various reasons. Has anyone out there also had this problem and found some workaround for it? I don't have a workaround so far, but I'll try to give as much detail as possible of what's happening on my installation, in the hope that it helps Steve to find out what's going wrong. The Asterisk machine is a K6/333 (too slow?), connected to the internal ISDN line of a small ISDN PBX (chan_capi). I've tried to receive faxes from a real fax machine and from a fax modem with no success. Both devices are connected to analog lines of the PBX. I run Asterisk in verbose mode, and try sending a Fax from the modem with sendfax (http://alpha.greenie.net/mgetty). After some time sendfax prints out the following message and exits: sendfax: FAILED to transmit 'f1.g3'. Transmission error: +FHNG:20 (Unspecified Transmit Phase B error) Asterisk's log looks like this: -- Executing RxFAX(CAPI[contr1/98]/0, /tmp/fax.tif) in new stack Changed from phase 0 to 1 Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 TSI: 43 20 20 20 20 20 37 39 36 39 36 33 32 20 31 31 39 20 39 34 2b TSI without final frame tag Remote fax gave TSI as: +49 911 2369697 DCS: 83 00 46 f0 00 DCS with final frame tag In state 9 DCS: Store and forward Internet fax: no Real-time Internet fax: no Can receive fax Data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at V.29 Changed from phase 3 to 5 Fast carrier up Fast carrier down Changed from phase 5 to 4 0 bad bits in trainability test Start rx document - compression 1 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Equalizer state: -7 (0.0, 0.0) - 0.0 -6 (0.0, 0.0) - 0.0 -5 (0.0, 0.0) - 0.0 -4 (0.0, 0.0) - 0.0 -3 (0.0, 0.0) - 0.0 -2 ( -0.08332,-0.68161) - 0.47154 -1 (0.66136, 0.74688) - 0.99522 0 (0.83336, 2.86588) - 8.90777 1 (0.66136, 0.74688) - 0.99522 2 ( -0.08332,-0.68161) - 0.47154 3 (0.0, 0.0) - 0.0 4 (0.0, 0.0) - 0.0 5 (0.0, 0.0) - 0.0 6 (0.0, 0.0) - 0.0 7 (0.0, 0.0) - 0.0 Equalizer state: -7 (0.24872,-0.01973) - 0.06225 -6 (0.06295,-0.59891) - 0.36265 -5 (0.04198,-0.41768) - 0.17622 -4 (0.12922, 0.34088) - 0.13290 -3 (0.29121, 0.41776) - 0.25932 -2 (0.05779,-1.18501) - 1.40758 -1 (0.67094,-0.35069) - 0.57314 0 (0.60670, 2.23125) - 5.34655 1 (0.34490, 1.36992) - 1.99564 2 ( -0.31233, 0.09159) - 0.10594 3 ( -0.00825, 0.00787) - 0.00013 4 (0.02226,-0.47177) - 0.22306 5 ( -0.17953, 0.14528) - 0.05334 6 ( -0.25318, 0.57800) - 0.39819 7 ( -0.10465, 0.05659) - 0.01415 Equalizer state: -7 (0.10674, 0.25606) - 0.07696 -6 ( -0.05000,-0.10084) - 0.01267 -5 ( -0.04246,-0.24755) - 0.06308 -4 (0.01990, 0.30283) - 0.09210 -3 ( -0.04673, 0.31177) - 0.09939 -2 ( -0.20575,-0.84531) - 0.75687 -1 (0.53295, 0.18096) - 0.31678 0 (0.84089,
Re: [Asterisk-Users] OT: Document Control System?
Chris Albertson wrote: The new OpenOffice works very well now and is completley cross platform. It also allows one to save in any of a serval file formats. I've been using it to produce HTML, PDF and plain text format copies of documentation. and I can run this same Open Office suite on Solaris, Linux and Windows. I will give it a shot. It's been a little bit since using it. Thanks, -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1-700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LCR for i4l (least cost routing)?
Hi, -Original Message- Returning to the original question of this thread: Have you ever looked at an LCR implementation instead of building your own Db? I know that i4l is not so popular around here, but still this might be of interest: Yes, this feature is now being supported by isdnlog. What it does is that it allows isdnlog to choose your telephone provider when placing a call through your ISDN card, depending on the time of day and the current rate information. Since isdnlog 4.16 an external script is called (if configured) to change various ISP settings (e.g. DNS lookup, proxy setup,...). This is cute, but it does require you to: - maintain current rates in a configuration list that I4L knows about - the telco won't tell you - be limited only to LCR with carrier-select codes, so VoIP is not an option. So a setup to do it from Asterisk most certainly has its use. There may be things to learn from the I4L sources in aspect of matching algorithms perhaps (I don't know, haven't looked at them yet). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
Mariam, I have added these lines and still no transfert menu on my IP10S Regards, Daniel Pavel Litvinenko wrote: Marian Danisek wrote: Daniel ANDRE wrote: I have the MGCP Firmware and call transfer doesn't work in my configuration. this is my mgcp.conf with working call transfer: [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no add the next: callgroup=0 cancallforward=yes transfer=yes ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Open Source Linux PBX!
and dangnabbit, don't cross post to the three lists -wasim On Thu, 13 Nov 2003, Sergio Serrano Revuelto wrote: try to cvs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source Linux PBX!
Title: Mensaje Try this guide: http://www.automated.it/guidetoasterisk.htm Miklos - Original Message - From: Sergio Serrano Revuelto To: [EMAIL PROTECTED] ; [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 8:02 AM Subject: RE: [Asterisk-Users] Open Source Linux PBX! try to cvs srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Quan Le TrungEnviado el: jueves, 13 de noviembre de 2003 10:43Para: [EMAIL PROTECTED]CC: [EMAIL PROTECTED]; [EMAIL PROTECTED]Asunto: [Asterisk-Users] Open Source Linux PBX! Hi! I have just bought the Wildcard TDM400P 4-port FXS PCI Card, and Wildcard X100P is a single-port FXO PCI Card to install on my computer to implement the PBX (Private Packet Exchange). However, I cannot download the corresponding softwares (asterisk, libpri and zaptel) at the following address: ftp://ftp.asterisk.org/pub/telephony . If anyone has already downloaded these softwares, please kindly send them to me via the following e-mail: [EMAIL PROTECTED] . Thanks in advance! P.S Please kindly send files in separate e-mails to me because of limited size of received e-mails. Best regards, Quan L. T.
RE: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
Hi, -Original Message- I have added these lines and still no transfert menu on my IP10S Can you try what happens if you program one of the function keys to send 'Flash' and use that ? I have not tried this, but I can imagine it might work that way.. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LCR for i4l (least cost routing)?
Hi! - maintain current rates in a configuration list that I4L knows about - the telco won't tell you The trick that this only needs to be done once per country. From what I found there seems to be a LCR update maintained for Germany. Unfortunately I didn't manage to find that for any other countries (I was looking for Belgium in particular) - on the other hand there are a couple of (often free) services that one could use to extract that data and put them into the required i4l LCR format. - be limited only to LCR with carrier-select codes, so VoIP is not an option. True. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems setting up E100P E1 germany
On Fri, 2003-10-24 at 17:46, Andreas Otto wrote: Hello list, Hi Andreas, ich habe Deine Mail auf der Liste gesehen, aber leider keine Antworten darauf. Hast Du mitlerweile eine Lösung gefunden, denn ich plane auch ein System mit der Zaptel E1 aufzubauen, und daher wäre ich stark daran interesiert, ob sowas in Deutschland überhaupt funktioniert. Würde mich freuen von Dir was dazu zu hören... Danke vielmals.. und sorry, dass ich hier auf die Frage keine Lösung sondern eine eigene Frage bringe :)) Gruss Ernst i've got some problems getting a E1 line with a E100P up and running (germany). # cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS/CRC4 YELLOW RED YELLOW RED sounds not so good. When launching asterisk, enabling pri debug on that span, i see outgoing attempts: ;-- snip [00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended ;-- snip Thats all. No reply from the net. I thought the problem was in the telco's side, but they double-checked everything and assure me that everything would be okay on their side. Anyone with some hints? Thanks a lot! - Andreas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bye Ernst - Ernst Lehmann Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems setting up E100P E1 germany
hmmm 2 fragen: 1. did you checked the wiring ? 2. are you using latest cvs ? regards Michael Bielicki PS: Andio are you in cologne over christmas ? On Thursday 13 of November 2003 11:46, Ernst Lehmann wrote: On Fri, 2003-10-24 at 17:46, Andreas Otto wrote: Hello list, Hi Andreas, ich habe Deine Mail auf der Liste gesehen, aber leider keine Antworten darauf. Hast Du mitlerweile eine Lösung gefunden, denn ich plane auch ein System mit der Zaptel E1 aufzubauen, und daher wäre ich stark daran interesiert, ob sowas in Deutschland überhaupt funktioniert. Würde mich freuen von Dir was dazu zu hören... Danke vielmals.. und sorry, dass ich hier auf die Frage keine Lösung sondern eine eigene Frage bringe :)) Gruss Ernst i've got some problems getting a E1 line with a E100P up and running (germany). # cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS/CRC4 YELLOW RED YELLOW RED sounds not so good. When launching asterisk, enabling pri debug on that span, i see outgoing attempts: ;-- snip [00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended ;-- snip Thats all. No reply from the net. I thought the problem was in the telco's side, but they double-checked everything and assure me that everything would be okay on their side. Anyone with some hints? Thanks a lot! - Andreas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LCR for i4l (least cost routing)?
Hi -Original Message- - maintain current rates in a configuration list that I4L knows about - the telco won't tell you The trick that this only needs to be done once per country. From what I found there seems to be a LCR update maintained for Germany. Unfortunately I didn't manage to find that for any other countries (I was looking for Belgium in particular) - on the other hand there are a couple of (often free) services that one could use to extract that data and put them into the required i4l LCR format. Really ? Can you point me there ? Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 based software client ..pls help
Hi, I am very closed to implement the IAX2 version in DIAX, but still some issues which I don't know how to handle, maybe someone from this list can help me. Trying to register with the * server as in version 1, I get the following in the * console: NOTICE[1150495040]: File chan_iax2.c, Line 2919 (register_verify): Inappropriate authentication received and in the client: Registration rejected There is something to be changed in iax.conf file regarding the user definition? there is an iax2.conf file too? I ask this because in the iax.conf file there is a line in the general section: port=5036 which is specific to IAX1 Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Assignement of extension to Netmeeting with dynamic IP address
Hello everybody, I wonder if anyone can help me in something I am trying to do but have no clue on how to do it: I have an Asterisk installation and I would like to be able to assign certain extensions to people with NetMeetings that take dynamic IP address. Does any one know how I can get the IP of an incoming channel in order to be able to dial back to that channel after the call is hangup? Does registering with a gatekeeper has anything to do with this? Any clues welcome Thanks, Anna Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Assignement of extension to Netmeeting with dynamic IP address
Registering with a gatekeeper is a MUST in this case... When your netmeeting is started , it is registred in the gatekeeper with a number (let's say 1234).. Now , when you call form * 1234 , the gatekeeper routes you to the netmeeting's IP. Dynamic IP's are a big problem with Netmeeting when you do authentification authorization , but i don't think that you use that... Hope this helps.. Regards Alex - Original Message - From: Panagidou Anna [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 1:19 PM Subject: [Asterisk-Users] Assignement of extension to Netmeeting with dynamic IP address Hello everybody, I wonder if anyone can help me in something I am trying to do but have no clue on how to do it: I have an Asterisk installation and I would like to be able to assign certain extensions to people with NetMeetings that take dynamic IP address. Does any one know how I can get the IP of an incoming channel in order to be able to dial back to that channel after the call is hangup? Does registering with a gatekeeper has anything to do with this? Any clues welcome Thanks, Anna Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Couple of Questions for Australian Users!
Just a couple of questions for Aussie users/resellers! I have only just started to look at asterisk a couple of weeks ago and have found some very intresting discussions and some useful info on what can and can´t be done with it and technology. The questions i have is, Are there people using it is Aussie?( I would say yes so prob answered my own question ) :) Hardware, is the digium hardware OK in australia? or does it require changes/mod´s or is just a pain in the butt? Alternate hardward for Aussie if above is an issues? Resellers/Importers of the Digium hardware? I have looked at the website and see that there are aussie resellers but they all seem to have their own agenda and the one I spoke to was not intrested in selling me hardware alone they wanted to do me a whole service deal! Also IP phones? What type/recomendation? I am going to be using Phone Software as well on laptops but I need an IP phone option to do phones over WAN. I really only know of the cisco and avaya types, I have not had any contact with any other types of phones. Intrested in Aussie suppliers for what ever phone you would recomened unless they are great phones then I would have to find an USA reseller. OK think thats all! Thanks Guys doing some great work with what looks to be some very GREAT software! Thanks in advance for your help! David Uzzell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Assignement of extension to Netmeeting with dynamic IP address
Panagidou Anna wrote: Hello everybody, I wonder if anyone can help me in something I am trying to do but have no clue on how to do it: I have an Asterisk installation and I would like to be able to assign certain extensions to people with NetMeetings that take dynamic IP address. Does any one know how I can get the IP of an incoming channel in order to be able to dial back to that channel after the call is hangup? You, certainly, don't want to do something like that. Does registering with a gatekeeper has anything to do with this? Yeap, that's the right solution. Just instruct all Netmeeting users to register with a gatekeeper. Each user will register with a unique number/ID, known to you. You can use these numbers/IDs to connect extenion numbers with real IP phones (in extensions.conf). Any clues welcome Thanks, Anna Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Hi to ALL my name is Dimitri and im a CEO of startup Company in Italy focused on Internation call traffic i usualy use Asterisk (very good app :-) ) for switching call. I ask now to Asterisk User of Telecom Company if is possible to cooperate in creation of network International POP call Termination through Voip Tunnel from us. What we think about? Thanks to all Dimitri Bellini ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Rgds, Adam -Original Message- From: reseaux [mailto:[EMAIL PROTECTED] Sent: 13 November 2003 13:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Hi to ALL my name is Dimitri and im a CEO of startup Company in Italy focused on Internation call traffic i usualy use Asterisk (very good app :-) ) for switching call. I ask now to Asterisk User of Telecom Company if is possible to cooperate in creation of network International POP call Termination through Voip Tunnel from us. What we think about? Thanks to all Dimitri Bellini ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LCR for i4l (least cost routing)?
Hi! Really ? Can you point me there ? Let's see if I can dig up again what I found... The trick that this only needs to be done once per country. From what I found there seems to be a LCR update maintained for Germany. Look here (German telco reates only, but I saw some Austrian project as well): http://sourceforge.net/projects/rates4linux/ http://rates4linux.sourceforge.net/r4l/index.php3 Quote: This page is about the rates4linux project. The projects goal is to provide a database with current phone rates for use within Linux. Currently we only provide German, Austrian Swiss, Dutch and some French providers, and this database is only supported by ISDN4Linux. Feel free to use the mailing-list to contact us, or write an email to one of us if you're from another country or have problems using this database. English and German language is welcome! looking for Belgium in particular) - on the other hand there are a couple of (often free) services that one could use to extract that data and put them into the required i4l LCR format. Free LCR data for Germany: http://www.telefonsparbuch.de/tmpl/calc/telephone/lcr/index.htm General info: http://www.linux-magazin.de/Artikel/ausgabe/2001/05/voip/voip.html Der Least Cost Router (LCR) Isdnrate ist Bestandteil von Isdn4linux. Damit der LCR korrekte Daten liefert, ist es wichtig, dass das ISDN- System korrekt konfiguriert ist. Vor allem muss in /etc/isdn/isdn.conf unter AREACODE die eigene Vorwahl eingetragen sein. Ob der LCR korrekt konfiguriert ist, lsst sich feststellen: isdnrate -b 0 -H 04181 Eine 153 Sekunden lange Verbindung von +49 40/ Hamburg nach +49 4181/ Buchholz kostet am Mon Jan 22 10:18:46 2001 Hierbei sollte statt Hamburg der Ort angezeigt werden, in dem sich das Gateway befindet. Ist der LCR richtig konfiguriert, kann er als Daemon gestartet werden: isdnrate -D2 Wenn der Daemon korrekt leuft, funktioniert folgender Befehl ohne Probleme: isdnrate -C -b 1 04181 01013_1:Tele 2 Preselection DEM0.153 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I hate to do this but..
I hate to bring this thread back to life, but... it may be possible to get it supported, do you think the price point is remotely competitive with Digium hardware? Also as I am not about to divulge my information to them to look in the downloads section, what is the licensing of their SDK? What is the licensing of the driver? Steven On Tue, 2002-11-26 at 14:52, Jamin W. Collins wrote: Is there any current/planned support for Aculab hardware? http://www.aculab.com Looks like they have Linux drivers and an SDK. Has any advancement taken place in this? Has someone developed a working channel driver for this product? I have one, and would like to see if it would be a possibility to get working... Thanks -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limit timeout of outgoing calls??
In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart
Re: [Asterisk-Users] I hate to do this but..
Josh Roberson wrote: On Tue, 2002-11-26 at 14:52, Jamin W. Collins wrote: Is there any current/planned support for Aculab hardware? http://www.aculab.com Looks like they have Linux drivers and an SDK. Has any advancement taken place in this? Has someone developed a working channel driver for this product? I have one, and would like to see if it would be a possibility to get working... Flebay it and use the proceeds to support Digium by purchasing Zaptel hardware. http://www.digium.com/ Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Limit timeout of outgoing calls??
AGI has set autohangap command. That should do it. Ta SJ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, November 13, 2003 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Limit timeout of outgoing calls?? In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart
Re: [Asterisk-Users] Limit timeout of outgoing calls??
Bartosz Jozwiak wrote: In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart d'oh switch-1*CLI show application AbsoluteTimeout -= Info about application 'AbsoluteTimeout' =- [Synopsis]: Set absolute maximum time of call [Description]: AbsoluteTimeout(seconds): Set the absolute maximum amount of time permitted for a call. A setting of 0 disables the timeout. Always returns 0. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Limit timeout of outgoing calls??
At 12:27 13/11/03 +, you wrote: AGI has set autohangap command. That should do it. Ta SJ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, November 13, 2003 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Limit timeout of outgoing calls?? In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart Use Absolute timeout application Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] recommendation?
Hello, I am completely new to this topic. Therefore I would like to know: Is there a list of recommended ip-phones for use with asterisk and/or a list of recommended resellers availalbel? What shipping/handling costs for a single phone are to be expected, when a reseller is located in the US or in Asia and the recipient in Europe/Germany? Any experiences? Regards gh __ Gesendet von Yahoo! Mail - http://mail.yahoo.de Logos und Klingeltöne fürs Handy bei http://sms.yahoo.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New User - Auto Attend / IVR
Hi All, Could anyone please point me in the direction of any information on using * in auto attend / IVR mode? Sorry if this has being asked before but I have search and can find no information. And YES I will be supporting the mission and purchasing Digium by purchasing Zaptel hardware! Regards to you all Kind regards Nick From: Nick Grindley Position: Managing Director / CEO Company:Intelligent Television and Video Limited Country:United Kingdom E-Mail: mailto:[EMAIL PROTECTED] www:www.itvv.co.uk Tel:+44 1723 500767 - Extension 527 Fax:+44 1723 501208 This e-mail, and any attachment, is private and confidential. If you have received it in error, please delete it from your system, do not use or disclose the information in any way, and notify us immediately. The contents of this message may contain personal views which are not necessarily the views of Intelligent Television and Video Limited, unless they are specifically stated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] speex 8kbit/s
Hi, is it possible to configure speex 8 kbit/s between asterisk servers? David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
we offer IAX/IAX2 a-z termination and would be interested in any exchange anytime. Would be very interested in talking to you, just email me your number and we can have a chat :) cheers Michael On Thursday 13 of November 2003 13:52, reseaux wrote: Hi to ALL my name is Dimitri and im a CEO of startup Company in Italy focused on Internation call traffic i usualy use Asterisk (very good app :-) ) for switching call. I ask now to Asterisk User of Telecom Company if is possible to cooperate in creation of network International POP call Termination through Voip Tunnel from us. What we think about? Thanks to all Dimitri Bellini ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exit the Directory Application?
How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Thanks, Marcus
[Asterisk-Users] Graphical Interface
Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David
[Asterisk-Users] 2 AGI questions..
Question 1.. Do the say number and say digits commands in AGI scritps work? If I use EXEC SayNumber 123 it works but is I try say number 123 it doesn't.. I think I have the syntax right becaasue thats how its shown when typing show agi on a console and also on the agi pages I have looked at.. Question 2.. Can an AGI script be executed on the h extension so that is will run at the end of each call? This is basically the script now so there can't be a problem with the script.. #/usr/bin/php -q ?php ob_implicit_flush(true); set_time_limit(15); exit(); ? Thats it.. But I get this.. -- Executing AGI(SIP/2010-a826, end1.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/end1.php == Spawn extension (local, h, 1) exited non-zero on 'SIP/2010-a826' My other script ends like this.. -- AGI Script start1.php completed, returning 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to interconnect gnugk and asterisk?
Hello folks. We are trying to interconnect an asterisk installation with a gnugk 2.0.5 installation to become able to use some H323 hardware that needs a gatekeeper (particulary an Ericsson WebSwitch 100). We have managed asterisk to dial H323 endpoints successfully (although calls are interrupted immediately after connection with spawn extension exited non-zero), but we could not manage gnugk to dial to asterisk. What I am confused most about is, what asterisk is for the gatekeeper: An Endpoint? A Gateway? A Neighbour? Further, asterisk seems to use root as H323 ID and I could not find out how to change that. Any help appreciated, working h323.conf/gnugk.ini would be perfect. kr Christoph Lechleitner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface
David Winkler wrote: Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David There are a few projects on the grow.. mostly web based.. Nothing complete yet.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Graphical Interface
Hello, I have developed a graphical interface using Perl/TK that has the following features: I'm still cleaning up the client code, but it will be released before the end of the month on Sourceforge. Here are some of the things I have added to the code: - Recording of any Zap channel by extension they are connected to at the click of a button- A refreshing list of active Zap channels- dialing a number by entering in a number or selecting from a list of recently dialed numbers and clicking a DIAL button- Asterisk based conference-calling of up to 6 external channels(even on single-line phone)- Admin section that allows you to Hangup any Zap channel at the click of a button- Call Parking and retrieval from specific extensions- Runs on Linux and Windows On the server side you will need a MySQL server, a couple AGI scripts and some custom dialplan extensions, but the Asterisk code itself is unaltered. On the Client side you just need to have perl and Tk/tcl modules installed on Linux and on windows you just need Activestate perl, you also need to make sure you have the Net:Telnet and Net::MySQL perl modules loaded on both(these are easy to get and have no prerequisites). MATT--- -Original Message-From: David Winkler [mailto:[EMAIL PROTECTED]Sent: Thursday, November 13, 2003 8:42 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Graphical Interface Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David
Re: [Asterisk-Users] pick up ringing exten
Hello, So the callgroup=2, pickupgroup=2 i have to add to my sip.conf for a SIP channel right ? And then do I have to but something in extensions for *8 ? Bart - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 7:41 AM Subject: Re: [Asterisk-Users] pick up ringing exten Is it possible with Asterisk to pick up ringing extension from other extension? So I do not have to run to other desk to pick up the phone. Sure, just add callgroup=2 pickupgroup=2 to each extension definition in sip.conf as an example. Dial *8 to pick up that ringing extn. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I hate to do this but..
Date: Thu, 13 Nov 2003 07:19:59 -0500 From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] I hate to do this but.. Reply-To: [EMAIL PROTECTED] Josh Roberson wrote: On Tue, 2002-11-26 at 14:52, Jamin W. Collins wrote: Is there any current/planned support for Aculab hardware? http://www.aculab.com Looks like they have Linux drivers and an SDK. Has any advancement taken place in this? Has someone developed a working channel driver for this product? I have one, and would like to see if it would be a possibility to get working... Flebay it and use the proceeds to support Digium by purchasing Zaptel hardware. http://www.digium.com/ Jeremy McNamara Hi , I hope that you aren't to quick to ditch aculab project. I dont see Aculab as an competitor to Digium at all. Their cards does cost about 3 times as much as the TE410 that I use now. Why would anyone even consider using these cards then ?. I have myself been using them for 10+ years so I have a bit of experience with their cards. If it was possible to use an Aculab card and a TE410 in the same server then we would be able to dump the last of our old servers and replace them with '*' servers. The reason that we use the Aculab cards is because of their C7/SS7 ISUP support which is a 'must' in several of our installations. In addition to this they do support a large number of legacy R2 protocol and they do carry PTT approvals for using these protocols in almost any country. I think that the card they have with 4 E1's / H323 and ethernet could be considered just as a MG seen from the '*' side and that might be the most easy way to go if you dont want go all the way and make a channel driver. My 1c on that issue. Freddi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I hate to do this but..
I think that the card they have with 4 E1's / H323 and ethernet could be considered just as a MG seen from the '*' side and that might be the most easy way to go if you dont want go all the way and make a channel driver. My 1c on that issue. Freddi The Aculab VoIP card actually only supports 2E1. Also, it is possible to use an API to drive it to handle the RTP only, leaving SIP/h.323 whatever to the application. However, what you can't do with the Aculab side of things (and I have though about this myself) is use it as a replacement for the Digium cards as there is no way to get the voice onto the PCI bus (even with Prosody). I'm afraid I would consider using these cards instead of the Digium ones otherwise for two reasons: 1) Worldwide certification and approval 2) Worldwide protocol support. However, they don't work and thats all there is to it! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pick up ringing exten
No, you don't. Its preprogrammed into asterisk already. Hello, So the callgroup=2, pickupgroup=2 i have to add to my sip.conf for a SIP channel right ? And then do I have to but something in extensions for *8 ? Bart - Original Message - Is it possible with Asterisk to pick up ringing extension from other extension? So I do not have to run to other desk to pick up the phone. Sure, just add callgroup=2 pickupgroup=2 to each extension definition in sip.conf as an example. Dial *8 to pick up that ringing extn. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 based software client ..pls help
On Thu, 13 Nov 2003 13:10:52 +0200, Dan [EMAIL PROTECTED] wrote: Hi, I am very closed to implement the IAX2 version in DIAX, but still some issues which I don't know how to handle, maybe someone from this list can help me. Trying to register with the * server as in version 1, I get the following in the * console: NOTICE[1150495040]: File chan_iax2.c, Line 2919 (register_verify): Inappropriate authentication received and in the client: Registration rejected Since you are using the iaxclient_lib core, your should really just need to change IAXVER=1 to IAXVER=2 in the iaxclient/lib/Makefile Note that the .remote element of the iaxc_ev_call_state struct only holds callerid number when you switch to IAX2, rather than the name and number. (I added a remote_name element, because I use that in iaxComm) There is something to be changed in iax.conf file regarding the user definition? there is an iax2.conf file too? I ask this because in the iax.conf file there is a line in the general section: port=5036 which is specific to IAX1 I've always run with the port line commented out. You'll notice that there is a console error associated with this. I always took it to mean that IAX2 would be disabled. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit timeout of outgoing calls??
13 2003 15:35 Peter Brown : In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart Use Absolute timeout application Can it be done by per-session limits? For example, when user calling to phone in prefix , we looking up user's info and see: ok, you have money to talk with no more than 10 minutes - and set him personal session timeout to 10 min? -- , (MT6561-RIPE, 2:463/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: I hate to do this but..
I think that the card they have with 4 E1's / H323 and ethernet could be considered just as a MG seen from the '*' side and that might be the most easy way to go if you dont want go all the way and make a channel driver. My 1c on that issue. Freddi The Aculab VoIP card actually only supports 2E1. Also, it is possible to use an API to drive it to handle the RTP only, leaving SIP/h.323 whatever to the application. However, what you can't do with the Aculab side of things (and I have though about this myself) is use it as a replacement for the Digium cards as there is no way to get the voice onto the PCI bus (even with Prosody). I'm afraid I would consider using these cards instead of the Digium ones otherwise for two reasons: 1) Worldwide certification and approval 2) Worldwide protocol support. However, they don't work and thats all there is to it! Linus I think that a simple/light integration should be possible. The Aculab acting as an endpoint should be able to register at the '*' (allthough I haven't actually tried it). The voice will not be using either the S.100 or PCI bus, it would be coming via the ethernet. You can think of it just like a MG that speaks SS7 against the PTT and H323 again '*'. The program that uses the Aculab API would just be a 'dumb' bridging program. The other benefit you get is that all the G723 and G729 to ulaw can be done on the embedded DSP on the board. Freddi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to interconnect gnugk and asterisk?
13 2003 16:59 Christoph Lechleitner : We are trying to interconnect an asterisk installation with a gnugk 2.0.5 installation to become able to use some H323 hardware that needs a gatekeeper (particulary an Ericsson WebSwitch 100). We have managed asterisk to dial H323 endpoints successfully (although calls are interrupted immediately after connection with spawn extension exited non-zero), but we could not manage gnugk to dial to asterisk. What I am confused most about is, what asterisk is for the gatekeeper: An Endpoint? A Gateway? A Neighbour? Further, asterisk seems to use root as H323 ID and I could not find out how to change that. Any help appreciated, working h323.conf/gnugk.ini would be perfect. In gnugk.ini I describe a peer: [RasSrv::PermanentEndpoints] 10.253.1.253=ASTERISK;4117 In asterisk's h323.conf: disallow=all allow=alaw allow=ulaw (yes, I can't use anything different from G.711, and also if I write down allow=all here all calls fail). gatekeeper = disable AllowGKRouted = yes [harbour] type=peer context=default host=10.253.1.254 (there host is host running gnugk) and that's all! When somebody reaches gnugk with call to 4117, it is routed to * and inside * correctly routed as I wish (as I describe in extensions.conf for 4117). BUT! I can't do calls from * to gnugk with mysterious error not enough bandwidth Maybe you can share your configs too? I'm interesing in h323.conf, sip.conf and extensions.conf. -- , (MT6561-RIPE, 2:463/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pick up ringing exten
When I am trying to pickup calling extension with my Cisco ATA I got this in SIP debug: Looking for *8# in extensions-fxs SIP/2.0 404 Not Found Bart - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 11:43 AM Subject: Re: [Asterisk-Users] pick up ringing exten No, you don't. Its preprogrammed into asterisk already. Hello, So the callgroup=2, pickupgroup=2 i have to add to my sip.conf for a SIP channel right ? And then do I have to but something in extensions for *8 ? Bart - Original Message - Is it possible with Asterisk to pick up ringing extension from other extension? So I do not have to run to other desk to pick up the phone. Sure, just add callgroup=2 pickupgroup=2 to each extension definition in sip.conf as an example. Dial *8 to pick up that ringing extn. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I hate to do this but..
What about their FAX support? Looks like they support T.37 (store and forward) fax and T.30 Lee Goodman - Original Message - From: Josh Roberson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 7:07 AM Subject: [Asterisk-Users] I hate to do this but.. I hate to bring this thread back to life, but... it may be possible to get it supported, do you think the price point is remotely competitive with Digium hardware? Also as I am not about to divulge my information to them to look in the downloads section, what is the licensing of their SDK? What is the licensing of the driver? Steven On Tue, 2002-11-26 at 14:52, Jamin W. Collins wrote: Is there any current/planned support for Aculab hardware? http://www.aculab.com Looks like they have Linux drivers and an SDK. Has any advancement taken place in this? Has someone developed a working channel driver for this product? I have one, and would like to see if it would be a possibility to get working... Thanks -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Steve Sobol schrieb: Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... http://www.voipproviderslist.com ? Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... Your probably right and it soon will do, because of my work with Asterisk (and general VoIP tech) our company has agreed that as we are the second largest PSTN provider in The Netherlands (we also operate in Norway and Austria) we should leverage our large switched telephone network by providing SIP/H.323 access to it. It's a brand spanking new product, the product team are still trying to get their heads around it but as soon as they do there will be a press announcement and documents posted on our site. Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit timeout of outgoing calls??
Hello, I have tested it with Grandstream Budgetone phone. And with this phone it is working very well. But it does not want to work with CISCO ATA. Cisco ata is sending *8# Budgetone phone is sending *8 Can I change something in Cisco ATA to make it work ? Or it is a BUG of Asterisk?? Bart - Original Message - From: Max Tulyev [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 11:55 AM Subject: Re: [Asterisk-Users] Limit timeout of outgoing calls?? 13 2003 15:35 Peter Brown : In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart Use Absolute timeout application Can it be done by per-session limits? For example, when user calling to phone in prefix , we looking up user's info and see: ok, you have money to talk with no more than 10 minutes - and set him personal session timeout to 10 min? -- , (MT6561-RIPE, 2:463/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
cd /usr/src patch -p0 file.diff bkw On Thu, 13 Nov 2003, Robert Boardman wrote: me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit timeout of outgoing calls??
SORRY WRONG POST!! - Original Message - From: Bartosz Jozwiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 12:44 PM Subject: Re: [Asterisk-Users] Limit timeout of outgoing calls?? Hello, I have tested it with Grandstream Budgetone phone. And with this phone it is working very well. But it does not want to work with CISCO ATA. Cisco ata is sending *8# Budgetone phone is sending *8 Can I change something in Cisco ATA to make it work ? Or it is a BUG of Asterisk?? Bart - Original Message - From: Max Tulyev [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 11:55 AM Subject: Re: [Asterisk-Users] Limit timeout of outgoing calls?? 13 2003 15:35 Peter Brown : In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart Use Absolute timeout application Can it be done by per-session limits? For example, when user calling to phone in prefix , we looking up user's info and see: ok, you have money to talk with no more than 10 minutes - and set him personal session timeout to 10 min? -- , (MT6561-RIPE, 2:463/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem in transfer the records of user id to the MySql database
Hello all, I am using the mysql cdr module to stored cdr in MySql database. I have added some users through extensions.conf exten = 1000#,1,GoTo(callout,s,1) exten = 1001#,1,GoTo(callout,s,1) exten = 1002#,1,GoTo(callout,s,1) exten = 1003#,1,GoTo(callout,s,1) I want to store that Users id in the mysql database. Please help me if any one solved this type of problem. Thanks. Dipak _ Apply to 50,000 jobs now. http://go.msnserver.com/IN/36715.asp Post your CV on naukri.com today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multi call iconenct?
Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas?
Re: [Asterisk-Users] Exit the Directory Application?
On Thursday 13 November 2003 07:31, Marcus Adolfsson wrote: How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Just enter a new extension. For example, if you want # to exit the Directory application, program the # extension. exten = #,1,Goto(s,5) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap timeout not occurring
Some channel banks have reversal polarity detection (far end answer supervision). How * handle this signal? Jorge Don Pobanz wrote: On Wednesday, November 12, 2003 3:47 PM, Tom Weeks [SMTP:[EMAIL PROTECTED] wrote: Good day, I am trying to setup an outbound dial plan which will time out if no answer. Using a X100P with the following dial command : exten = 101,3,Dial(Zap/1/3036972357,5) ; try the desk line - fail to step 104 That is right, it does not time out and never will correctly since the X100P 'seizes' the line from the phone company immediately upon dialing so the X100P does not know whether the far end phone is ringing, is busy or if someone has answered. I believe it would take getting some type of digital interface (T1 or ISDN) in order to have far end answer supervision. Don Pobanz It dials out successfully, but never times out. I have a basic Zapata config : group = 1 context = RedRockWeb language = en signalling = fxs_ks usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = no immediate = no channel = 1-2 Suggestions? Thanks! Tom [EMAIL PROTECTED] www.tellink-corp.com 303-697-2357 303-697-3103 fax File: ATT00015.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface
On Thursday 13 November 2003 07:41, David Winkler wrote: Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. See gastman for a manager and gnophone for a client. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7910
Hi Sorry for maybe trivial question... anyone now how to setup OR have a [SEPDefault.cnf] file for Cisco 7910 to make to work with Asterisk on Skinny Regards A TeleSoft Ltd. ( DDI : - +44 20 860785347 Fax : - +44 20 86078531. E-mail: - [EMAIL PROTECTED]Web Site : -http://www.telesoft.ltd.uk
Re: [Asterisk-Users] multi call iconenct?
http://connect.voicepulse.com - Andrew Thompson - Original Message - From: Shoval Tomer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 10:33 AM Subject: [Asterisk-Users] multi call iconenct? Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas? Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
[Asterisk-Users] IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX
-extensions.conf on voip1
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
-extensions.conf on voip2
exten = 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)
voip1*CLI iax show users
Username Secret Authen Def.Context A/C
voip2 md5,plaintextIAX No
voip1*CLI iax show peers
Name/UsernameHost Mask Port Status
voip2/voip2 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip2*CLI iax show users
Username Secret Authen Def.Context A/C
voip1 md5,plaintextIAX No
voip2*CLI iax show peers
Name/UsernameHost Mask Port Status
voip1/voip1 x.x.x.x (S) 255.255.255.255 5036 UNREACHABLE
voip1*CLI iax debug
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip1*CLI
voip2*CLI iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00 Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01 Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00 Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00 Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00 Type: IAX Subclass: POKE
voip2*CLI
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multi call iconenct?
Shoval Tomer wrote: Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas? These run * and can provide iax2 terminations with multiple call presentations per pnone number www.NuFone.comHighly recommended on this list www.VoicePulse.com Sounds good but no service reviews yet There have been several others reported recently on this list which may or may not support IAX. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
I can't remember who it was, but someone on this list was aiming to compile a list. We certainly replied, offering UK rest of world IAX (and SIP) termination. If that project isnt happening, it would be a great idea if someone else wanted to take it up. Linus - Original Message - From: Steve Sobol [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 3:31 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multi call iconenct?
Try Nikotel - they have multi account feature and very good services :) http://www.nikotel.com/index.htm Shoval Tomer wrote: Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multi call iconenct?
http://www.xvoip.com Call as many as you want ;-) Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 11:57 AM Subject: Re: [Asterisk-Users] multi call iconenct? http://connect.voicepulse.com - Andrew Thompson - Original Message - From: Shoval Tomer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 10:33 AM Subject: [Asterisk-Users] multi call iconenct? Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas? Ë^®+$RÇ«²f¢-)à-+-Ë^®+$RÇ«²X¬¶Çb,+¦r?¡¶ÚþX¬¶Çb,+¦r?¿T¨¥T©ÿ-+-Swèý«-z¸¬'ë ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Linus, We started this list on Forums : http://asterisk.xvoip.com So any body can post info about services, etc which are off-topic for this list.. Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Linus Surguy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 12:08 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) I can't remember who it was, but someone on this list was aiming to compile a list. We certainly replied, offering UK rest of world IAX (and SIP) termination. If that project isnt happening, it would be a great idea if someone else wanted to take it up. Linus - Original Message - From: Steve Sobol [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 3:31 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
Thanks for your help Brian how would you come the the values required for the distincive ring? Robb Brian West wrote: cd /usr/src patch -p0 file.diff bkw On Thu, 13 Nov 2003, Robert Boardman wrote: me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
- Original Message - From: Asterisk online forums [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 5:48 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Linus, We started this list on Forums : http://asterisk.xvoip.com So any body can post info about services, etc which are off-topic for this list.. Just a suggestion, but under 'Service Providers' why not start a subject 'Other Providers'? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I hate to do this but..
Jeremy McNamara said: Flebay it and use the proceeds to support Digium by purchasing Zaptel hardware. http://www.digium.com/ I like the zaptel cards, I really do. The price per port is better than other fxo/fxs's in almost all cases and the support I've received from Digium is second to none. However, deploying several pbx's for friends (and using the X100p's), I've come to find that the echo cancellation to be sub-par when contrasted with other makers. I realize ecan is a huge and complex issue, but it's central to really convincing me to 'buy' more Digium hardware. Such is the catch-22; it seems we need people to buy/support Digium in hopes that they can improve things overall -- hoping this includes work on dsp.c to achieve better ecan. It's difficult to recommend that someone buy 4 or 6 ports of fxo knowing that they'll probably be complaining to me about excessive echo. Note: I've tried every possible echo suppressor listed in the Makefile, and so far Mark2 + aggressive suppression works best, at the sacrifice of FDX. I'd like FDX and aggressive ecan. G This isn't a jab at zaptel or Mark, this note is to highlite one issue I've had recommending Digium hardware. --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ESQL ODBC (was MySQL Licence may be changing.esql)
There are any number of ODBC libraries available for UNIX/Linux. I'd say ODBC has _better_ support under UNIX then under MS Windows There are both Open Source and comercial ODBC implementations. Go to google, freshmeat.net or source fourge and try some searches and see what turns up. The hard part will be the _time_ it takes you to evaluate all of them and select what's best. Go to the postgresql.org site and you can hunt down some ODBC stuff too. There s now an easy point and click way to build a GUI, web based ODBC application. Just get Open Office and use the html editor to build a form and you can connect it to a DBMS via ODBC. and te result will be completley cross platform. There is another cross-DBMS interface with even longer history then ODBC. That's embedded SQL. A lot of programmers will directly call the MySQL/PostgreSQL/Oracle library interface functions from their C code. This is NOT the best design. THere is an ANSI standard way to call a DBMS from C that is supported by most DBMSes called embedded SQL or ESQL and has been used for at least 10 or maybe 15 years now. esql is very clean in that the results of a query are stuffed directly into variables and the SQL querries are in-lined with the C code. esql seems to mostly just work when you change databases. All of the largescale DBMSes suport esql. Still ODBC works too. and ODBC will talk to data source that do not do SQL. So you could have a flat file or LDAP used for the back end. whereas esql is tied to the SQL DBMS. --- Robert G. Werner [EMAIL PROTECTED] wrote: Thu, 13 Nov 2003, Fresno CA, Adam Hart, spoke these words: Prehaps a novel thought but what about ODBC for asterisk? Isn't that the whole idea of standards and such, stop adding support for every db and just have odbc? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ODBC is not well supported on UNIX. You have to have the connector software, as well as the ODBC libs. -- Robert G. Werner [EMAIL PROTECTED] x5204,ICQ #311363925 Udall's Fourth Law: Any change or reform you make is going to have consequences you don't like. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Roger Schreiter wrote: Hmmm, this information should be on a website somewhere... http://www.voipproviderslist.com ? Sure. :) Was just observing that there seems to be a big influx of We are doing VoIP termination posts in the past 24 hours and was pointing out - in a roundabout way - that while this is not an inappropriate place to put that information, there are other places where it should go too. -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
DONE. Any other suggestions ? Please let me know if we need to add some additional sections. Any suggestions are welcome. Thanks, Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Linus Surguy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 1:16 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) - Original Message - From: Asterisk online forums [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 5:48 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Linus, We started this list on Forums : http://asterisk.xvoip.com So any body can post info about services, etc which are off-topic for this list.. Just a suggestion, but under 'Service Providers' why not start a subject 'Other Providers'? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange problem with * and festival
I'm trying to use festival with * and for an unknown reason , it fails..
Here is a small debug:
*CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Executing Festival(H323:20231, just a test) in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (default, 500, 1) exited non-zero on 'H323:20231'
ClearCallThread::ClearCallThread: Object initialized.
-- Hungup 'H323:20231'
WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
ClearCallThread::ClearCallThread: Object deleted.
I'm using the lastest * from CVS.
My festival is patched and compiled and it is working just fine..
* even creates the file used for cache and that file is ok too..
I've done some debugging and found out that the exit point is located in
app_festival.c near line 161.
Here is the code:
if (res 1) {
res = -1;
break;
}
Can somebody tell me what's wrong ?
Thanks a lot
Alex
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I hate to do this but..
You mentioned that when you enable the aggressive suppression support you sacrifice FDX. Forgive my ignorance, but what is FDX? By sacrificing it, what am I not getting? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton L. Kapela Sent: Thursday, November 13, 2003 10:17 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] I hate to do this but.. Jeremy McNamara said: Flebay it and use the proceeds to support Digium by purchasing Zaptel hardware. http://www.digium.com/ I like the zaptel cards, I really do. The price per port is better than other fxo/fxs's in almost all cases and the support I've received from Digium is second to none. However, deploying several pbx's for friends (and using the X100p's), I've come to find that the echo cancellation to be sub-par when contrasted with other makers. I realize ecan is a huge and complex issue, but it's central to really convincing me to 'buy' more Digium hardware. Such is the catch-22; it seems we need people to buy/support Digium in hopes that they can improve things overall -- hoping this includes work on dsp.c to achieve better ecan. It's difficult to recommend that someone buy 4 or 6 ports of fxo knowing that they'll probably be complaining to me about excessive echo. Note: I've tried every possible echo suppressor listed in the Makefile, and so far Mark2 + aggressive suppression works best, at the sacrifice of FDX. I'd like FDX and aggressive ecan. G This isn't a jab at zaptel or Mark, this note is to highlite one issue I've had recommending Digium hardware. --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] box for asterisk
Hi All, We are looking at creating our first asterisk box, what type of server requirements do we need to keep in mind? Is there any preferred server that is used for asterisk? How processor intensive is asterisk? And what is the requirements for the sound card? Thanks, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI verbose command
Using the verbose comand in an AGI you can feed information back to the console in the same way that applications do.. so by specifying different information to be returned at various verbosity levels the AGI will produce results similar to those of Asterisk applications.. Is there any standard to this?? In otherwords somthing like application execution info at verbosity level 2, Detailed progress info at level 3 and debugging info and level 4... I guess the Asterisk developers would need to answer this.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EU SIP Phone providers
Does anyone know of SIP phone providers (Grandstream in particular) who are located in Germany (or the EU) Thanks for any info. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: I hate to do this but..
Freddi Hansen wrote: I'm afraid I would consider using these cards instead of the Digium ones otherwise for two reasons: 1) Worldwide certification and approval 2) Worldwide protocol support. October 02, 2003 Extra...Extra...Read all about it. Digium wants the whole world to know that the quad span T1/E1/PRI card, the TE410P, has passed Telecom FCC, Euro, and Australian certifications. This milestone in our hardware development gives the TE410P a significant boost for deploying applications worldwide and allows it to compete with similar cards in the market. We passed the rigorous certifications last week and the paperwork must be completed to make it official, but the hard work is done and we are excited to announce this great news to our customers. EMC testing is next. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] box for asterisk
Steve Bradwell wrote: Hi All, We are looking at creating our first asterisk box, what type of server requirements do we need to keep in mind? Is there any preferred server that is used for asterisk? How processor intensive is asterisk? And what is the requirements for the sound card? I was just about to send a mail like this to the list, even though I'm a fairly longtime user of asterisk. What I would like to ask is slightly different, along these lines: for a SOHO type situation, if those of you on the list with lots of asterisk experience were to be able to choose the ideal hardware platform to put a TDM/X100P card (or two?) in, what would you use? I'm thinking about such things as plenty of PCI slots, nice behavior wrt interrupt handling, CPU speed, etc. I have gotten several requests from potential users along this line, and would love to spec out a few boxes that would represent state of the art. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with * and festival
Do you answer the channel first?
exten = s,1,Answer
exten = s,2,Festival,Asterisk rocks!!
bkw
On Thu, 13 Nov 2003, Alexandru Coseru wrote:
I'm trying to use festival with * and for an unknown reason , it fails..
Here is a small debug:
*CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Executing Festival(H323:20231, just a test) in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (default, 500, 1) exited non-zero on 'H323:20231'
ClearCallThread::ClearCallThread: Object initialized.
-- Hungup 'H323:20231'
WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
ClearCallThread::ClearCallThread: Object deleted.
I'm using the lastest * from CVS.
My festival is patched and compiled and it is working just fine..
* even creates the file used for cache and that file is ok too..
I've done some debugging and found out that the exit point is located in
app_festival.c near line 161.
Here is the code:
if (res 1) {
res = -1;
break;
}
Can somebody tell me what's wrong ?
Thanks a lot
Alex
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EU SIP Phone providers
On Thu, 2003-11-13 at 20:11, rnc Info Lists wrote: Does anyone know of SIP phone providers (Grandstream in particular) who are located in Germany (or the EU) nikotel perhaps? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
Daniel ANDRE wrote: Mariam, I have added these lines and still no transfert menu on my IP10S they must be aded before line = aaln/1 transfer works by flash event Regards, Daniel Pavel Litvinenko wrote: Marian Danisek wrote: Daniel ANDRE wrote: I have the MGCP Firmware and call transfer doesn't work in my configuration. this is my mgcp.conf with working call transfer: [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no add the next: callgroup=0 cancallforward=yes transfer=yes ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
Thats one thing that needs to be added to the patch.. I did this areound line 4471 in chan_zap.c ast_verbose( VERBOSE_PREFIX_3 DEBUG: %d\n, curRingData[counter1]); bkw On Thu, 13 Nov 2003, Robert Boardman wrote: Thanks for your help Brian how would you come the the values required for the distincive ring? Robb Brian West wrote: cd /usr/src patch -p0 file.diff bkw On Thu, 13 Nov 2003, Robert Boardman wrote: me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL PROTECTED] Sent: Thu, 13 Nov 2003 06:43:03 -0500 (EST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Distintive Ring on x100p On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface
Is there any pointers to some webapps that can be tried to configure/manage asterisk ? i know there is one on sourceforge.. but does not have any downloads attached to it. Sri WipeOut wrote: David Winkler wrote: Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David There are a few projects on the grow.. mostly web based.. Nothing complete yet.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: I hate to do this but..
Freddi Hansen wrote: I'm afraid I would consider using these cards instead of the Digium ones otherwise for two reasons: 1) Worldwide certification and approval 2) Worldwide protocol support. October 02, 2003 Extra...Extra...Read all about it. Digium wants the whole world to know that the quad span T1/E1/PRI card, the TE410P, has passed Telecom FCC, Euro, and Australian certifications. This milestone in our hardware development gives the TE410P a significant boost for deploying applications worldwide and allows it to compete with similar cards in the market. We passed the rigorous certifications last week and the paperwork must be completed to make it official, but the hard work is done and we are excited to announce this great news to our customers. EMC testing is next. Jeremy McNamara Hi Jeremy, I can see that name and comments doesn't match after your cut of the posting. We are using the TE410P but what I was writing is that we are NOT considering to replace the TE410P with boards from Aculab, we are replacing Aculab boards with Digium boards BUT we would need more Digium boards IF we could use both Digium and Aculab cards in the same server. The reason being that TE410P doesn't support SS7-ISUP so we continue using only Aculab cards in the servers that must support SS7/ISUP. Freddi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I hate to do this but..
On Thu, 2003-11-13 at 12:52, Ed Rubright wrote: You mentioned that when you enable the aggressive suppression support you sacrifice FDX. Forgive my ignorance, but what is FDX? By sacrificing it, what am I not getting? Full Duplex, or the ability to have both sides talking at once instead of using it like a walkie-talkie. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton L. Kapela Sent: Thursday, November 13, 2003 10:17 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] I hate to do this but.. Jeremy McNamara said: Flebay it and use the proceeds to support Digium by purchasing Zaptel hardware. http://www.digium.com/ I like the zaptel cards, I really do. The price per port is better than other fxo/fxs's in almost all cases and the support I've received from Digium is second to none. However, deploying several pbx's for friends (and using the X100p's), I've come to find that the echo cancellation to be sub-par when contrasted with other makers. I realize ecan is a huge and complex issue, but it's central to really convincing me to 'buy' more Digium hardware. Such is the catch-22; it seems we need people to buy/support Digium in hopes that they can improve things overall -- hoping this includes work on dsp.c to achieve better ecan. It's difficult to recommend that someone buy 4 or 6 ports of fxo knowing that they'll probably be complaining to me about excessive echo. Note: I've tried every possible echo suppressor listed in the Makefile, and so far Mark2 + aggressive suppression works best, at the sacrifice of FDX. I'd like FDX and aggressive ecan. G This isn't a jab at zaptel or Mark, this note is to highlite one issue I've had recommending Digium hardware. --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM40B X100P Call Waiting
When on a call from a TDM40B out through a X100P how do you answer a call waiting? I have tried the flash button on the phone and I have read about *0 which does not appear to work for me either. Is there some sort of configuration I have to do to make this work? My zapata config file follows. zapata.conf [channels]language = en ; TMD40Bsignalling = fxo_ksgroup = 2context = trustedchannel = 3-6 context = inbound-1include = extensionssignalling = fxs_ksgroup = 1callgroup = 1pickupgroup = 1usecallerid = yescallerid = asreceivedhidecallerid = nocallwaiting = yescallwaitingcallerid = nothreewaycalling = notransfer = noechocancel = yesechocancelwhenbridged = yesrxgain = 0.6txgain = 0.6immediate = nobusydetect = nocallprogress = nomusiconhold = randommailbox = 2000channel = 1
[Asterisk-Users] Asterisk on FreeBSD
I've tried to summarize my experiences of Asterisk on a FreeBSD system: http://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD If you have facts to add, please mail me off list or edit the Wiki page. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Graphical Interface
Phpconfig is available via cvs cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout zaptel libpri asterisk phpconfig regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sri Sent: Friday, 14 November 2003 6:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Graphical Interface Is there any pointers to some webapps that can be tried to configure/manage asterisk ? i know there is one on sourceforge.. but does not have any downloads attached to it. Sri WipeOut wrote: David Winkler wrote: Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David There are a few projects on the grow.. mostly web based.. Nothing complete yet.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.538 / Virus Database: 333 - Release Date: 10/11/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 2610 as an MGCP gateway
Title: Cisco 2610 as an MGCP gateway I'm looking for config examples to use a Cisco 2610 with an FXO card as an MGCP gateway for my Asterisk server. I would like the cisco config along with the Asterisk MGCP config. This is a test environment and this is the only equipment I have at the moment. I do have the phones also.
[Asterisk-Users] Errors in build.
I am getting the following error when I try to build asterisk. gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-11/13/03-12:48:12\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o chan_zap.c chan_zap.c: In function `zt_train_ec': chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function) chan_zap.c:1081: (Each undeclared identifier is reported only once chan_zap.c:1081: for each function it appears in.) make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 Here is the script I use to conduct the build. #!/bin/sh echo Now moving old directories. mv zapata zapata-`date +%Y%m%d` mv zaptel zaptel-`date +%Y%m%d` mv libpri libpri-`date +%Y%m%d` mv asterisk asterisk-`date +%Y%m%d` export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot echo CVS password is anoncvs cvs login cvs checkout zapata zaptel libpri asterisk cd zapata make clean ; make install cd zaptel make clean ; make install cd ../libpri make clean ; make install cd ../asterisk make clean ; make install Can anyone clue me in as to what I am doing wrong? Thank you, -Ryan -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: I hate to do this but..
I'm afraid I would consider using these cards instead of the Digium ones otherwise for two reasons: 1) Worldwide certification and approval 2) Worldwide protocol support. Extra...Extra...Read all about it. Digium wants the whole world to know that the quad span T1/E1/PRI card, the TE410P, has passed Telecom FCC, Euro, and Australian certifications. This milestone in our hardware development gives the TE410P a significant boost for deploying applications worldwide and allows it to compete with similar cards in the market. We passed the rigorous certifications last week and the paperwork must be completed to make it official, but the hard work is done and we are excited to announce this great news to our customers. EMC testing is next. Jeremy McNamara Jeremy, That was actually my posting you acredited to Freddi. I am *not* 'bashing' Digium etc., I think the cards are great, give good value for money and combined with Asterisk provide a good solution. However, we work and assist people deploying around the world and in many cases that is outside the those areas you mention, and therefore although the Digium coverage is great, in some cases we need more, both in terms of approval and protocol coverage. Aculab (and similar such as Dialogic etc.) have spent a lot of time, and a lot of money getting worldwide approval / protocol support. If for example we want to deploy a 'legal' box in for example Hong Kong - we can't use Digium's cards, so we use Aculab - this is why if someone made an Aculab driver for Asterisk this would assist a lot of people deploying 'real' 'telco approved' applications. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aculab SS7/ISUP (new subject)
Freddi Hansen wrote: with boards from Aculab, we are replacing Aculab boards with Digium boards BUT we would need more Digium boards IF we could use both Digium and Aculab cards in the same server. The reason being that TE410P doesn't support SS7-ISUP so we continue using only Aculab cards in the servers that must support SS7/ISUP. Do you use the Aculab SS7/ISUP together with Asterisk somehow? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Errors in build.
- Original Message - From: Ryan V. Howerton [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 3:23 PM Subject: [Asterisk-Users] Errors in build. I am getting the following error when I try to build asterisk. gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-11/13/03-12:48:12\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o chan_zap.c chan_zap.c: In function `zt_train_ec': chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function) chan_zap.c:1081: (Each undeclared identifier is reported only once chan_zap.c:1081: for each function it appears in.) make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 Here is the script I use to conduct the build. #!/bin/sh echo Now moving old directories. mv zapata zapata-`date +%Y%m%d` mv zaptel zaptel-`date +%Y%m%d` mv libpri libpri-`date +%Y%m%d` mv asterisk asterisk-`date +%Y%m%d` export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot echo CVS password is anoncvs cvs login cvs checkout zapata zaptel libpri asterisk cd zapata make clean ; make install cd zaptel Am I way off, or should the above line have been: cd ../zaptel make clean ; make install cd ../libpri make clean ; make install cd ../asterisk make clean ; make install Can anyone clue me in as to what I am doing wrong? Thank you, -Ryan -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - Andrew Thompson ,µêâ²E,z»j)b b²Ð,µêâ²E,z»%Ëlv(ºg(m§ÿåËlv(ºg(ùYùb²Ø§~ڲ׫É.±êì
[Asterisk-Users] Indications - ring signals etc
On request, I've updated the following page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20indications.conf with ring signals from Brazil. (And at the same time, the Brazilian signals was added to the CVS). If you have an entry in indications.conf that you want to share, a variant or configuration for a country not defined in indications.conf in the distribution - please add it to the wiki or mail me off list. A special thank you to everyone that suggests improvements and additions to the wiki by mailing me. It shows that the Wiki is used. Forgive me if I don't react instantly. Another thank you to all of you directly updating the wiki - you're important. Every addition builds on the knowledge base, making it easier to install, implement and use Asterisk. The Wiki site, voip-info.org, should now be more reachable since the DNS problems are sorted out and the backup procedure is changed. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : For the SQL gurus..
This comes to a statement of optimal. What is optimal? Optimal with respect to what??? We want something that adds as close to 0 to the time to connect a call. If your database is not in memory at the time (swapped out, whatever) then there is overhead that you have to consider. If the database is swapped out, someone should consider getting a new job. IMHO, a database admin's job isn't only to design the shite, but also specify hardware that should work with it. You won't use a Volkswagen beetle to transport 20 people at a time, would you? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background only responds to 1 digit
I have a problem where the Background application only seems to work if one digit is pressed. Extensions with multiple digits just timeout and asterisk hangs up. Below is the relevant excerpt from extensions.conf. In this example, pressing 2 will access the service menu. Then pressing 1 will do the echo test ok but pressing 8463 or 33 will cause an invalid extension message. Any ideas for a solution are appreciated. Robert [default] exten= s,1,ResponseTimeout,10 exten= s,2,Background(rnc-mainmenu) exten= 1,1,Goto(local-extensions,2001,1) exten= 2,1,Goto(services,s,1) [services] exten= s,1,ResponseTimeout,10 exten= s,2,Background(rnc-svcmenu) exten = 1,1,Answer exten = 1,2,Playback(demo-echotest) exten = 1,3,Echo() exten = 1,4,Playback(demo-echodone) exten = 1,5,Wait(1) exten = 1,6,Playback(vm-goodbye) exten = 1,7,Wait(1) exten = 1,8,Hangup exten =33,1,Answer exten =33,2,MusicOnHold(random) exten =8463,1,DateTime() exten =8463,2,Wait(2) exten =8463,3,Playback(vm-goodbye) exten =8463,4,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Needed: Digium/Asterisk Resller in Riverside, CA Area
An end user company in Riverside California needs a small Asterisk system installed. They need 2 FXO ports and 4 IP Phones with Voicemail, Conferences, etc configured. If any Digium resellers in that area are interested, please email me off list so that I can get you in touch with my contact there. Sean ___ Sean Robertson NETXUSA p. 800-289-6389 f. 864-233-4344 Ask me about Voice over IP. http://www.netxusa.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] OT : For the SQL gurus..
take off your mail list NOW or ELSE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
