[Asterisk-Users] SPA 841 form SIPURA
Hello, How good is :SPA 841 form SIPURA. Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't compile asterisk-oh323 on Mandrake 10
Hi All: I am trying to compile asterisk with oh323 but I can't compile it. I am using instruction provided at http://www.oinko.net/astrecipes/index.php?from=1q=astrecipes/compiling+asterisk+with+oh323. The compile error I am getting is as follows. Quite a few other people are getting exactly same error but no one has posted a fix for this error yet. Any help would greatly be appreciated. gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/include/asteri\sk -I../wrapper -g -c -o chan_oh323.o chan_oh323.cIn file included from /usr/include/string.h:33, from chan_oh323.c:34:/usr/lib/gcc-lib/i586-mandrake-linux-gnu/3.3.2/include/stddef.h:213: error: syntax error before "typedef"In file included from chan_oh323.c:34:/usr/include/string.h:38: error: syntax error before "extern"/usr/include/string.h:39: error: parse error before "__THROW"/usr/include/string.h:43: error: parse error before "__THROW"/usr/include/string.h:56: error: parse error before "__BEGIN_NAMESPACE_STD"/usr/include/string.h:58: error: syntax error before "extern" My gcc version is 3.3.2 Thanks Karim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk rpms (was: Does anyone run Asterisk on FC4? with Digium's TDM40B cards)
Hi, On Sat, Aug 06, 2005 at 09:05:51AM -0400, Zachary Whitley wrote: On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote: Kumara Jayaweera wrote: Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Yes, Without problems. Can u install RH9 on ur box? I'm assuming that Madhawa is suggesting that you install RH9. I've installed Asterisk on FC4 with very few problems. Start with a standard FC4 installation then install the following rpms from atrpms.net: asterisk-addons asterisk-sounds zaptel zaptel-devices One little problem. Maybe it's been fixed but last time I checked it wasn't. In the /etc/init.d/zaptel the path to ztcfg is incorrect. Find all references to ztcfg and change them to = /usr/sbin/ztcfg You can copy the sample configs from /usr/share/doc/asterisk-1.0.9/configs/ to get you going. Running asterisk -c -vvv will let you know which ones you need. The rest is going to be specific to your hardware and setup. Good luck. Thanks for the comments on the packages. I'm looking for more feedback and improvements on the asterisk and friends rpms at ATrpms. Red Hat Linux, Fedora Core and RHEL (and clones) are supported. There are already some bug reports at bugzilla.atrpms.net on enhancements and bugs in the packages, see http://bugzilla.atrpms.net/buglist.cgi?query_format=advancedshort_desc_type=allwordssubstrshort_desc=long_desc_type=substringlong_desc=asteriskbug_file_loc_type=allwordssubstrbug_file_loc=bug_status=NEWbug_status=ASSIGNEDbug_status=REOPENEDemailassigned_to1=1emailtype1=substringemail1=emailassigned_to2=1emailreporter2=1emailcc2=1emailtype2=substringemail2=bugidtype=includebug_id=votes=chfieldfrom=chfieldto=Nowchfieldvalue=cmdtype=doitorder=Reuse+same+sort+as+last+timefield0-0-0=nooptype0-0-0=noopvalue0-0-0= Thanks! -- Axel.Thimm at ATrpms.net pgph3UTL1jE08.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom 301 phone advice
I have two 300s and 4 500s. The 300s talk the same language, but have a lousy screen. The other thing to consider is, while it does have the 'monitor only' speaker, the volume is horrible. Cranked up to its highest setting, you can't hear voicemail with ANY background sound. Go for the 501 and sleep at night ;) Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Jim Duda [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, August 06, 2005 7:50 AM Subject: [Asterisk-Users] polycom 301 phone advice Can anyone tell me if the CallerID information is automatically displayed on the LCD screen of the 301? Can asterisk manipulate the LCD screen for the purposes of displaying callerid? Is this a good quality phone? Or, is the 501 worth the added expense? I believe the only real differences between 301 and 501 are that the 501 has one additional line (total of 3) and has speaker phone capability. Thanks, Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Planet sip phone and asterisk
Does anyone know how to connect the planet VIP-152T phone to asterisk because I can't get it to work at all. What configuration is needed? I have a [EMAIL PROTECTED] setup and works perfect with the soft phones but not with the planet... I 'm a newby so help me pleaseee Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialplan mapping for multiple outbound providers to determine best rates
[EMAIL PROTECTED] wrote: This is a similar idea to LCR (least cost routing) on normal pbx systems. Any advice would be nice, since I'm sure those users who use asterisk for more commercial purposes have figured our a way to do this... Jump to the LCR section on this page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20billing Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and firmware 4.0 problem
On Sat, Aug 06, 2005 at 08:50:58AM +0200, Christian Stredicke wrote: Please take a look at http://www.snom.com/howto40.html. We tried to make the upgrade procedure as smooth as possible, if you are having problems please tell us and we will try to make it more simple. For example, if you have a batch of phones give us an email and we will send you the files in one go. New phones dont need that upgrade procedure. It is only necessary when you are crossing the 4.0 version border. For example, all 320 already have the certificate installed already, so for 320 there is no need to go throught the procedure. For release notes for 4.0, please check out http://www.snom.com/snom360_release_notes.html. I found the license issue info on the website. So my question boils down to this: Does this explain why they would not register, or do I have to worry that there is some new setting which caused the problem? I do not want to go through the pain of upgrading the customer's phones (probably with a site visit) only to find that I have to downgrade them and go though it again with 4.1. Also, is there any way to tell the phone, *before a reboot* that I want it to update the firmware? I do most of my maintenance remotely, and I can tell the phones where to find new firmware and clicking Load will start the reboot process. However, I need a person there to press the Check button so that it will really update the firmware. Is there any way around this so that I can update the phone after-hours and remotely? Thank you... Not just for this answer, but for all the answers I get from this list! I've been working with asterisk for a bit over a year, though I do not know near as much as many of you. I try to chime in with answers when I can, but I have received much more info from this list than I have contributed. This list is a true shining example of how Open Source Software can work! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Friday, August 05, 2005 8:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Snom 360 and firmware 4.0 problem I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the error occurs on that one, too. The distributor's latest suggestion was to go to the newest firmware, 4.0. I did that on the new 360 (from the RMA) and with the same account settings as the one it was replacing, it could not register with *. Since I was in a pinch, I updated the firmware down to the latest below 4.0 and the phone works just fine. Does anyone with more knowledge than I know what might be going on? Maybe a new default setting in 4.0 that's breaking things? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc HFC-S in nt mode but no dial tone after pickup
I have installed a HFC-S card in nt mode according to the documentation on voip-info.org and it works quite well except for two problems: 1. When I pick up the phone I no not get a dial tone indicating that I can start dialing but asterisk seems to jump automatially to extension s. 2. When I try to create an extension starting with * (like _*XX) asterisk also goes straight into s instead of executing _*XX. Can anybody help me with theseß Cheers, Arik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA 841 form SIPURA
On Sun, August 7, 2005 2:07 am, [EMAIL PROTECTED] said: How good is :SPA 841 form SIPURA. Not good if voice quality is a requirement. Talking on the handset sounds to the caller and callee like you're on one of those really old speakerphones that clips the beginning of each phrase after a pause. There's a ramp-down of white-noise at the end of each phrase. The speakerphone is totally useless as the user is completely inaudable unless yelling with their face directly in front of the unit. I've fiddled with a batch of 7 of them, with little useful support from Sipura by-the-way, for a few months and finally gave up last week. I ordered a batch of Polycom IP501 units to replace them. I should also mention that Sipura's support systems appear to have recently changed. I used to get a real person with a useful response or valid request for more info within a day, usually within a few hours. I just had a SPA-3000 fail with a hardware problem and spent over a week going back and forth with a tech support person who was obviously just cutting and pasting responses from a crib sheet. He knew almost nothing about the product and didn't understand wat I was writing. Tech support has gone from good to intolerable IMHO. Their SIP implementations seem fine. I've not had any trouble getting them to work with Asterisk. I like their ATAs and the SPA-3000 is awesome IMHO. Their 841 is a unmentinable corpse-strewn horror. Their support is going to the dogs with their recent acquisition by Linksys/Cisco. My $0.02, Paul -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC [EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk registered in ser proxy
In you sip.conf what if you change: register = 7771::[EMAIL PROTECTED]/7771 to register = 7771:[EMAIL PROTECTED]/7771 PB Jenna Cole wrote: im using iptel.org SER proxy. the proxy is working without authentication. the problem is that the Asterisk is not sending a REGISTER sip message. --- Juan Salas [EMAIL PROTECTED] escribió: Which SIP proxy are you using? Check the authentication parameters (user-id, auth-id, password)? Post the sip debug peer 10.0.0.115 logs. Saludos. jsalas -Mensaje original- De: Jenna Cole [mailto:[EMAIL PROTECTED] Enviado el: Friday, August 05, 2005 12:58 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] asterisk registered in ser proxy if i remove that line, asterisk stop sendind the OPTIONS message to the SIP PROXY, but it's still NOT sending the REGISTER message. i would alse need to register more than one number --- Eric Wieling aka ManxPower [EMAIL PROTECTED] escribió: Jenna Cole wrote: thanx for the reply. i tried it, and now asterisk is doing something. but the problem is that instead of sendind a REGISTER message to the SIP PROXY, it is sendind an OPTIONS message, and the PROXY responds with 404 NOT FOUND ihave in my sip.conf file: register = 7771::[EMAIL PROTECTED]/7771 [10.0.0.115] type=peer context=default secret= username=7771 fromdomain=10.0.0.115 canreinvite=yes dtmfmode=RFC2833 qualify=yes host=10.0.0.115 insecure=very fromuser=7771 Remove the qualify=yes and Asterisk will stop sending the options packets. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Only terrorists use the r option to Dial. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and firmware 4.0 problem
Does this explain why they would not register, or do I have to worry that there is some new setting which caused the problem? I do not want to go through the pain of upgrading the customer's phones (probably with a site visit) only to find that I have to downgrade them and go though it again with 4.1. I guess you haven't read my earlier post to your question, it would have answered your question Also, is there any way to tell the phone, *before a reboot* that I want it to update the firmware? I do most of my maintenance remotely, and I can tell the phones where to find new firmware and clicking Load will start the reboot process. However, I need a person there to press the Check button so that it will really update the firmware. Is there any way around this so that I can update the phone after-hours and remotely? Thank you... Not just for this answer, but for all the answers I get from If you would have read the manual of the phone or took 5 seconds to browse through the html interface of the phone itself you would have found the parameter to get the behaviour you want of the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse DTMF Problems Anyone?
I too have been having inbound dtmf problems with VP Connect using iax2 for inbound. When I switched to sip, and added the relaxdtmf=yes, all 10 inbound test calls I did seemed to work fine for dtmf. I'm going to leave my config set up to use sip for inbound VP Connect calls for a while and see how if functions. thanks for the relaxdtmf tip Umair. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to configure/install ISDN Card
Hello, I'm a newbie trying to figure out how to install an ISDN Card. I downloaded the latest Asterisk iso and did install it (actually version 1.3). ISDN Card details: Eicon Diva Server BRI-2M/-2F I did listen something that capi 2.0 is needed and some drivers etc. but I can't figure out how it works. Has some of you maybe experience in that? Or does someone of you know an easy to understand how to? The purpose of that ISDN Card is to use it as backup line eg. When the main line goes down the call should go out trough the ISDN. Could that be implemented with LCR Leased Cost Routing ? Sorry for such basic stuff questions. Any help very appreciated Cheers migmig -- GMX DSL = Maximale Leistung zum minimalen Preis! 2000 MB nur 2,99, Flatrate ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC web can't connect to DB
I noticed another user on this had the same problem that i'm having.
Through the webpage I can't connect to database I get unavailable
database message. In my http logs I get this:
[Sun Aug 07 00:13:38 2005] [error] [client 172.25.25.30] DBI
connect('database=astcc ; your astcc database name;host=127.0.0.1 ; astc
c host name','astccadmin ; your MySQL user name that can access
ASTCC',...) failed: Access denied for user: 'astccadmin ; your MySQL
user [EMAIL PROTECTED]' (Using password: YES) at
/var/www/cgi-bin/astcc-admin/astcc-admin.cgi line 67, referer:
http://172.25.2 5.8/cgi-bin/astcc-admin/astcc-admin.cgi?mode=Configure
Any ideas??
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Re: [Asterisk-Users] asterisk rpms (was: Does anyone run Asterisk on FC4? with Digium's TDM40B cards)
There are already some bug reports at bugzilla.atrpms.net on enhancements and bugs in the packages, see http://bugzilla.atrpms.net/buglist.cgi?query_format=advancedshort_desc_type=allwordssubstrshort_desc=long_desc_type=substringlong_desc=asteriskbug_file_loc_type=allwordssubstrbug_file_loc=bug_status=NEWbug_status=ASSIGNEDbug_status=REOPENEDemailassigned_to1=1emailtype1=substringemail1=emailassigned_to2=1emailreporter2=1emailcc2=1emailtype2=substringemail2=bugidtype=includebug_id=votes=chfieldfrom=chfieldto=Nowchfieldvalue=cmdtype=doitorder=Reuse+same+sort+as+last+timefield0-0-0=nooptype0-0-0=noopvalue0-0-0= Thanks! ___ I didn't know that there was a bugzilla site setup for atrpms. Thanks for the great repo. I'll make sure to post anything that I find. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to configure * for Net2phone using innomedia settings
Hello all: I need help configuring * to register with Net2phone using the credentials provided with an Innomedia MTA 3328-2r fxs device. In using ethereal I see where the user agent string includes the MAC address of the device. Net2phone also is using MD5 authentication. If the mac address is in the user agent string does that prevent me from registering with 2 different Innomedia Accounts on the same * box? Is there a way for asterisk send a variable user agent string depending on the account? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom 301 phone advice
Chris Coulthurst wrote: I have two 300s and 4 500s. The 300s talk the same language, but have a lousy screen. The other thing to consider is, while it does have the 'monitor only' speaker, the volume is horrible. Cranked up to its highest setting, you can't hear voicemail with ANY background sound. Go for the 501 and sleep at night ;) There are several gain options in the Polycom config files, the *might* even be model specific, but I don't recall for sure. Search the sample config files for the string gain. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls from Asterisk to CallManager 3.0 how?
Hello all We succesfully added a H323 Gateway to our CallManager 3.0 that resides in Mexico and were/are able to make calls from CallManager SCCP phones to the Asterisk Server phones in the U.S.; however, we have not been able to call from Asterisk server in U.S. to CallManager phones in Mexico Here is what we tried: 1. Adding a Gatekeeper into CallManager and then have Asterisk (and also stand alone softphone) send calls thru the gatekeeper. That didn't work. The gatekeeper was never found. I am sure we configured something wrong, but not sure what. So we need help with this is this the correct/one of the approaches? 2. Using the same gateway that is currently working to make calls from CallManager to Asterisk. That didn't work either. I found severals threads on lists.digium.com and elsewhere that talked about succesfully doing the same thing I am trying to do, but no details as how to doit. So, if anybody can give me a clue as how to do this it would very much be appreciated. I also found information on cisco website detailing how to add h323 gateways and gatekeepeers to CallManager but again no luck. By the way, the calls that are made from CallManager phones in Mexico to Asterisk server in U.S. were so far of optimal quality, even though the test calls made were from from phones in the U.S. connected to the CallManager Server in Mexico. I was surprised how good they sounded. I even routed some calls to the PSTN using the XP100 clone and they sounded excellent. Since I seem to be clueless as how to approach this problem, I am not sure what other information I need to provide in order for anybody to help. I would post results back later regardless of outcome, but would like to make this work thanks to all in advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
Hi all, I'm new to the forum. Oh nonewbie question coming, I hear you all cry! I'm playing around with [EMAIL PROTECTED] and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've posted my config files in Adobe pdf format at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've spent at least a couple of weeks trying to sort it out and am now seeking your good advice. Asterisk pc is attached to a small network which connects to the internet via a 3COM firewall broadband router. The Asterisk has an IP on the network off DHCP and it's IP is cleared through the firewall by DMZ setting. I'm signed up with sipgate.co.uk Any advice of sorting out incomming calls would be gratefully received. Thanks Brian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P - All extensions have same CallerID
As you can see, the channels are set properly. One thing I did notice is all of the ;; in front of the [ext] sections. Does that seem correct? I removed them and it didn't change anything. Other files that you would like to look at? Thanks, Mike Looks a bit more complicated than it needs to be but I don't think there's anything wrong with the zapata.conf (and friends). Anyone know if modifying configuration files by hand breaks AMP? I'm assuming that that AMP is going to expect a particular setup. The ';' is just a comment character. Everything after a ; is a comment including the other ;'s. the [xxx] I'm guessing is there just to let you know what extension AMP is using for that channel. I think that it is very confusing to use the configuration file syntax in comments. It makes it hard to see what is a comment and what isn't but that's just my opinion. I think the next thing to look at is the extensions.conf file. head wrapped around all of this. The good thing is once I know how to do it, I don't need to ask again. Give a man a configuration and he'll make calls for a day. Teach a man how to configure and he'll make calls for a lifetime ;) I tried usinig AAH first too but found that it got you going with something that sort of works quickly but then trying to work backwards from a complex system was too difficult. There were too many confounding variables. Is it A, is it B is it A and B I've found it easier to start with rpms. You don't have to worry about compiling and installing the system and you get to start from a simple state and work your way up. There are many great sources of info but here are a few I've found helpful: This list google voip-info.org asteriskdocs.org VoIP Telephone with Asterisk by Paul Mahler The sample config files included with asterisk. (Search for theConfigFileYouWantToCheckOut.sample) Please feel free to add to this list if you know any good sources of documentation. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA 841 form SIPURA
-Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Paul Dugas Envoyé : dimanche 7 août 2005 16:11 À : Asterisk Mailing List Objet : Re: [Asterisk-Users] SPA 841 form SIPURA On Sun, August 7, 2005 2:07 am, [EMAIL PROTECTED] said: How good is :SPA 841 form SIPURA. Not good if voice quality is a requirement. Talking on the handset sounds to the caller and callee like you're on one of those really old speakerphones that clips the beginning of each phrase after a pause. There's a ramp-down of white-noise at the end of each phrase. This is not true You have to switch to last firmware and/or disable silent suppression The speakerphone is totally useless as the user is completely inaudable unless yelling with their face directly in front of the unit. Have a look at last firmware and user setup Best Reagrds Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can call from iax extn but cannot call it - unable to cteate channel iax
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid=some name 300
auth=md5
Then in my extensions.conf I have:
exten = 300,1,Dial(IAX/${EXTEN},20)
exten = 300,2,Hangup
I can dial from iaxComm (a soft IAX client) and that works fine. But when I
try to dial 300 get:
WARNING[22077]: channel.c1970 ast_request: No channel type registered for
'IAX'
NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of type
'IAX'
I have restarted Asterisk after config change.
What have I not done. I am just testing the iaxComm program.
Angus
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Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
I've posted my config files in Adobe pdf format at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I think you're either going to get complaints about the pdf files or people are simply going to ignore your question. Is there any reason you chose to post pdf's instead of just posting the ASCII files? And you're really going to hear it when people follow your link and find the file isn't there. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BudgeTone 100 Woes
I knew about that one. I have Silence Suppression set to NO. Jim Tony Mountifield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote: -=-=-=-=-=- -=-=-=-=-=- I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my voice is find on the other end, but my receiving end gets the cutouts. I find it rather annoying and tend to always use the Sipura phones, which work great. I believe it's a configuration issue on the BudgeTone. I've followed all the examples and notes I could find on the subject on voip-info.com. Has anyone else had this experience with the BudgeTone? In general, I like the phone, wish it worked better. Turn OFF Silence Suppression in the Budgetone configuration. If SS is enabled, the phone stops sending RTP when you are silent. Asterisk relies on the incoming RTP stream being continuous, using it to generate the timing for the outgoing RTP. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it - unable to cteate channel iax
Angus Comber wrote: I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]: channel.c1970 ast_request: No channel type registered for 'IAX' NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of type 'IAX' Try IAX2 -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice prompt repository
I was wondering if there would be any interest or support out there for an IVR voice prompt repository, a la atrpms but for voice prompts instead of rpms. I was thinking of something that collected the meta data such as spoken text, gender, file size, speaker ID, language, duration, encoding, MD5, etc. prompts could also be organized into collections almost like IVR themes where a complete set of standard base prompts are collected so you could make one change in your configuration file and all prompts are changed to the new speaker. There could also be a rating for quality of recordings and links to professional services if you needed better quality or specific recordings, etc. It could be like pod casting for IVR. Suggestions, comments, questions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it - unableto cteate channel iax
On 8/7/2005, Angus Comber [EMAIL PROTECTED] wrote:
Then in my extensions.conf I have:
exten = 300,1,Dial(IAX/${EXTEN},20)
exten = 300,2,Hangup
I can dial from iaxComm (a soft IAX client) and that works fine. But
when I try to dial 300 get:
WARNING[22077]: channel.c1970 ast_request: No channel type registered
for 'IAX'
NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of
type 'IAX'
I have restarted Asterisk after config change.
What have I not done. I am just testing the iaxComm program.
You have not used a correct Technology in your dial command.
The 'show application dial' says:
Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL])
Technology is the chan_.so file loaded and resource is the defined in
the configuration file for the technology.
So either you need a chan_iax.so - OR - you need to READ the
extension.conf
file. I have never seen a dial command like yours. One that is VERY
close
is one like Dial(IAX2/${EXTEN},20)
I am not just picking on you Angus. I do tend to read almost every
message
coming through the list and I get tired of reading all the questions that
5 minutes of reading the configuration files or searching the wiki (as out
of date as it is) or even typing 'help' at the CLI prompt can remedy.
Guess I'm just getting old...
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Re: [Asterisk-Users] voice prompt repository
I think that's a good idea, something that I'd have use of atleast :) That be if there does not exist a site like this already, but non that I have heard of. Johan There are two major products that come out of Berkeley: LSD and UNIX. We don't believe this to be a coincidence. -- Jeremy S. Anderson Zachary Whitley skrev: I was wondering if there would be any interest or support out there for an IVR voice prompt repository, a la atrpms but for voice prompts instead of rpms. I was thinking of something that collected the meta data such as spoken text, gender, file size, speaker ID, language, duration, encoding, MD5, etc. prompts could also be organized into collections almost like IVR themes where a complete set of standard base prompts are collected so you could make one change in your configuration file and all prompts are changed to the new speaker. There could also be a rating for quality of recordings and links to professional services if you needed better quality or specific recordings, etc. It could be like pod casting for IVR. Suggestions, comments, questions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it - unableto cteate channel iax
[EMAIL PROTECTED] wrote: coming through the list and I get tired of reading all the questions that 5 minutes of reading the configuration files or searching the wiki (as out of date as it is) or even typing 'help' at the CLI prompt can remedy. And some of us are getting tired of people complaining about the wiki being outdated/incorrect, when they have just as much ability to fix it as anyone else does. It's especially annoying when someone posts a comment attached to a wiki page saying this is wrong here's the correct info' when they could have just edited the page in the first place (which requires the same amount of time). If you find a wiki page that is incorrect, incomplete or needs any other editing, do it! The rest of the community will be thankful for your help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BudgeTone 100 Woes
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote: I knew about that one. I have Silence Suppression set to NO. Ah, ok. Puzzling then. If you'd like to post the full budgetone config page(s), one of us might be able to spot something. What revision of budgetone firmware are you using? Is the budgetone talking to an Asterisk box of yours, or directly to an external provider? Cheers Tony Jim Tony Mountifield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote: -=-=-=-=-=- -=-=-=-=-=- I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my voice is find on the other end, but my receiving end gets the cutouts. I find it rather annoying and tend to always use the Sipura phones, which work great. I believe it's a configuration issue on the BudgeTone. I've followed all the examples and notes I could find on the subject on voip-info.com. Has anyone else had this experience with the BudgeTone? In general, I like the phone, wish it worked better. Turn OFF Silence Suppression in the Budgetone configuration. If SS is enabled, the phone stops sending RTP when you are silent. Asterisk relies on the incoming RTP stream being continuous, using it to generate the timing for the outgoing RTP. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] z-machine + asterisk = fun!
I was tinkering with Asterisk and the Festival text-to-speech engine, and wrote some short Asterisk::AGI scripts to read back live weather reports. After that, I thought I needed something more interactive to work with... Then I had a flashback to 1996, first year university, standing in the C O club at the University of Waterloo, where someone had just pulled out their US Robotics Palm Pilot and started up Zork. A couple of hours later, after a quick trip to the campus computer store, I was playing Zork in the palm of my hand! Now Zork is back! Listen as the eerie voice of Festival takes you into the Underground Empire, and marvel as you explore this world with your dial pad, unlocking the secrets within! Note that some more commands need to be implemented before you can actually -enter- the underground empire. For now you can just futz around on the surface. See $dtmf_translation in Asterisk/Games/Zork/ZIO_Asterisk.pm for number-to-phrase translations. I've posted the proof-of-concept at http://uc.org/read/Zasterisk Feedback is welcomed ;-) Cheers, Simon P. Ditner | The Toronto Asterisk Users Group -- http://taug.ca | Join by sending email to [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it -unableto cteate channel iax
On 8/7/2005, Kevin P. Fleming [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: coming through the list and I get tired of reading all the questions that 5 minutes of reading the configuration files or searching the wiki (as out of date as it is) or even typing 'help' at the CLI prompt can remedy. And some of us are getting tired of people complaining about the wiki being outdated/incorrect, when they have just as much ability to fix it as anyone else does. At least I used a personal pronoun... or are you speaking for Digium? 8-) It's especially annoying when someone posts a comment attached to a wiki page saying this is wrong here's the correct info' when they could have just edited the page in the first place (which requires the same amount of time). I don't do that - go find those that do if you're annoyed. If you find a wiki page that is incorrect, incomplete or needs any other editing, do it! The rest of the community will be thankful for your help. Well - here on the list I have seen people still using 1.0.5. Maybe that info is correct for that version. I don't know. I'm not an expert. I tend to use the documentation supplied with and inside asterisk. And nothing you have said invalidates my statement. Most questions could easily be answered by reading... the config files, the help screens, or the wiki. I am sure the wiki will have massive updates after the 1.2 freeze hits. I am glad we have both had the chance to vent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
Here's my kernel info: Linux asterisk.hulber.com 2.6.9-11.EL #1 Fri May 20 18:17:57 EDT 2005 i686 i686 i386 GNU/Linux And my Asterisk version: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-05 21:21:13 UTC Kumara Jayaweera wrote: Hi MARK, Thanks a lot for the reply. my box is Intel based. and there is no USB conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may be the place to see. Thank you Kumara - Original Message - From: MF Hulber [EMAIL PROTECTED] To: Kumara Jayaweera [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 06, 2005 4:02 PM Subject: Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards I don't use Fedora but I do use RHEL AS 4 without any problem. Do you have any USB conflicts? MARK. Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
And card status: asterisk*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 Kumara Jayaweera wrote: Hi MARK, Thanks a lot for the reply. my box is Intel based. and there is no USB conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may be the place to see. Thank you Kumara - Original Message - From: MF Hulber [EMAIL PROTECTED] To: Kumara Jayaweera [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 06, 2005 4:02 PM Subject: Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards I don't use Fedora but I do use RHEL AS 4 without any problem. Do you have any USB conflicts? MARK. Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using * and 3rd party GW together
Dear folks, Actually this is my first post here, so sorry for any inconvenience. Im planning for a solution a bit larger in scale than ususal. I'm goin to use * as a PSTN gateway with E1 links and use two other 3rd party Gateways for FXO lines. I should be able to switch from every incoming channel to any outgoing one and also to some SIP softphones. I planned to use SER as a sip server but really dont know were I should enforce my call routing mechanisms. Is SER applicable of doing that or should i write any application on the SER to do so ro is there any need for a softswitch at all? Any help and hints would be highly appreciated, M. Shokuie Nia.Don't just search. Find. MSN Search Check out the new MSN Search! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it - unableto cteate channel iax
[EMAIL PROTECTED] wrote: snip I am not just picking on you Angus. I do tend to read almost every message coming through the list and I get tired of reading all the questions that 5 minutes of reading the configuration files or searching the wiki (as out of date as it is) or even typing 'help' at the CLI prompt can remedy. Guess I'm just getting old... Guess we all are, but that is better than the other choice. Many of us read all the posts, and some of us really get tired of a small number who continue to complain and write paragraphs on those who aren't able to either find the information they need or understand it when they do. There is a large disparity between the beginner and those who have lived with this camel ( Asterisk ) for months to years. Help from the CLI leaves a LOT to be desired. The Wiki is either correct, outdated, or wrong depending on which of the 1000 flavors of Asterisk one happens to have settled on, usually because it mostly works I could go on, but you get the idea. If you can help, do so, as someone else did for this fellow in just a few characters If all you can do is berate someone else for not reading and UNDERSTANDING, or assume they didn't find the answer in the bushels of poorly organized sometimes correct information, then just move on to the next post. John Novack ( old fart ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] z-machine + asterisk = fun!
Wow! Not sure what else to say. This ranks right up there with my ability to open my garage door from asterisk... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 07, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] z-machine + asterisk = fun! I was tinkering with Asterisk and the Festival text-to-speech engine, and wrote some short Asterisk::AGI scripts to read back live weather reports. After that, I thought I needed something more interactive to work with... Then I had a flashback to 1996, first year university, standing in the C O club at the University of Waterloo, where someone had just pulled out their US Robotics Palm Pilot and started up Zork. A couple of hours later, after a quick trip to the campus computer store, I was playing Zork in the palm of my hand! Now Zork is back! Listen as the eerie voice of Festival takes you into the Underground Empire, and marvel as you explore this world with your dial pad, unlocking the secrets within! Note that some more commands need to be implemented before you can actually -enter- the underground empire. For now you can just futz around on the surface. See $dtmf_translation in Asterisk/Games/Zork/ZIO_Asterisk.pm for number-to-phrase translations. I've posted the proof-of-concept at http://uc.org/read/Zasterisk Feedback is welcomed ;-) Cheers, Simon P. Ditner | The Toronto Asterisk Users Group -- http://taug.ca | Join by sending email to [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BudgeTone 100 Woes
Thanks for the assistance. I'm running version 1.0.6.7 of the software, tftp updated a few weeks ago. I'm interfacing with an Asterisk box on my local lan. My sip.conf is as follows: [100] type=friend context=home callerid=Jim 100 secret=mysecret host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 mailbox=100 disallow=all allow=ulaw allow=gsm Can you recommend a method to which I can post the configuration from the grandstream bt100 device? Jim Tony Mountifield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote: I knew about that one. I have Silence Suppression set to NO. Ah, ok. Puzzling then. If you'd like to post the full budgetone config page(s), one of us might be able to spot something. What revision of budgetone firmware are you using? Is the budgetone talking to an Asterisk box of yours, or directly to an external provider? Cheers Tony Jim Tony Mountifield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote: -=-=-=-=-=- -=-=-=-=-=- I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my voice is find on the other end, but my receiving end gets the cutouts. I find it rather annoying and tend to always use the Sipura phones, which work great. I believe it's a configuration issue on the BudgeTone. I've followed all the examples and notes I could find on the subject on voip-info.com. Has anyone else had this experience with the BudgeTone? In general, I like the phone, wish it worked better. Turn OFF Silence Suppression in the Budgetone configuration. If SS is enabled, the phone stops sending RTP when you are silent. Asterisk relies on the incoming RTP stream being continuous, using it to generate the timing for the outgoing RTP. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it -unableto cteate channel iax
[EMAIL PROTECTED] wrote: At least I used a personal pronoun... or are you speaking for Digium? 8-) Nope, just Sunday afternoon venting... no official position should be inferred from my comments G Well - here on the list I have seen people still using 1.0.5. Maybe that info is correct for that version. I don't know. I'm not an expert. I tend to use the documentation supplied with and inside asterisk. Well, we all know that is incomplete, and that the wiki is a valuable source of additional documentation, hints, examples and such. There is also nothing wrong with editing a page there saying this procedure/process/configuration has changed as of -XX-XX (or version 1.0.9), here is the updated information... And nothing you have said invalidates my statement. Most questions could easily be answered by reading... the config files, the help screens, or the wiki. I am sure the wiki will have massive updates after the 1.2 freeze hits. Agreed 100%. I was not in any way disagreeing with your comments to the OP, only reacting to the 'wiki is outdated' comment. I am glad we have both had the chance to vent And now back to your regularly scheduled newbie [EMAIL PROTECTED] questions :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA 841 form SIPURA
[EMAIL PROTECTED] wrote: How good is :SPA 841 form SIPURA. I won't be ordering any more of them... One of the units we ordered had problems with the dial pad not registering the correct key press. i.e. when pressing the line 1 button line 2 would activate and dialing 5 would dial 5 and 8. I would also agree with Paul that the voice quality was poor... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it - unableto cteate channel iax
If you find a wiki page that is incorrect, incomplete or needs any other editing, do it! The rest of the community will be thankful for your help. I don't want to get in the middle of this but what wiki are we referring to? voip-info.org/wiki-asterisk ?? I would be willing to contribute if I knew were to go. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it -unableto cteate channel iax
On 8/7/2005, John Novack [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: snip I am not just picking on you Angus. I do tend to read almost every message coming through the list and I get tired of reading all the questions that 5 minutes of reading the configuration files or searching the wiki (as out of date as it is) or even typing 'help' at the CLI prompt can remedy. Guess I'm just getting old... Guess we all are, but that is better than the other choice. Many of us read all the posts, and some of us really get tired of a small number who continue to complain and write paragraphs on those who aren't able to either find the information they need or understand it when they do. Well - the part snipped was the training part... 8-) There is a large disparity between the beginner and those who have lived with this camel ( Asterisk ) for months to years. Help from the CLI leaves a LOT to be desired. The Wiki is either correct, outdated, or wrong depending on which of the 1000 flavors of Asterisk one happens to have settled on, usually because it mostly works I could go on, but you get the idea. I think everyone gets that idea. If you can help, do so, as someone else did for this fellow in just a few characters If all you can do is berate someone else for not reading and UNDERSTANDING, or assume they didn't find the answer in the bushels of poorly organized sometimes correct information, then just move on to the next post. John Novack ( old fart ) Well - us old farts have to stand together (no one will stand near anyway). But I don't think just saying 'use IAX2' is going to enlighten anyone. That's why I included the whole Dial(Technology/resource) definition. There has to be a channel driver for the technology and the technology has to have a configuration file to assign the resources. I put that all in there so he could UNDERSTAND. And maybe others reading the list could understand it as well. And I guess IAX2 vs. IAX is a real bad example. LOL ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] z-machine + asterisk = fun!
On Sun, 2005-08-07 at 14:59 -0500, Tim Connolly wrote: Wow! Not sure what else to say. This ranks right up there with my ability to open my garage door from asterisk... Sarcasm or serious? Sounds cool to me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can call from iax extn but cannot call it -unableto cteate channel iax
On 8/7/2005, Zachary Whitley [EMAIL PROTECTED] wrote: If you find a wiki page that is incorrect, incomplete or needs any other editing, do it! The rest of the community will be thankful for your help. I don't want to get in the middle of this but what wiki are we referring to? voip-info.org/wiki-asterisk ?? I would be willing to contribute if I knew were to go. Thanks. Yup - that's it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
I've re-uploaded the config files in NON pdf Any help welcomed. Regards - Original Message - From: Brian McCarey To: asterisk-users@lists.digium.com Sent: Sunday, August 07, 2005 5:55 PM Subject: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate. Hi all, I'm new to the forum. Oh nonewbie question coming, I hear you all cry! I'm playing around with [EMAIL PROTECTED] and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've posted my config files in at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've spent at least a couple of weeks trying to sort it out and am now seeking your good advice. Asterisk pc is attached to a small network which connects to the internet via a 3COM firewall broadband router. The Asterisk has an IP on the network off DHCP and it's IP is cleared through the firewall by DMZ setting. I'm signed up with sipgate.co.uk Any advice of sorting out incomming calls would be gratefully received. Thanks Brian. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] list of T.38 providers on wiki: please contribute
Please excuse my ignorance but doesn't the VOIP/PSTN gateway (Broadvoice,VP Connect) have to support T38 in order for an T38 supported ATA to do any good ? Yes. BroadVox supports it. I've started a wiki page to track providers who do, mainly because such providers are so hard to find, and the available information out there can be so misleading and unreliable: http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers+T.38 Currently even providers that *do* offer T.38 rarely advertise that fact on their website (for example, I can't find anything about T.38 on BroadVox's site, and we support faxing is meaningless). I've heard rumors that Packet8 and SunRocket support T.38, but I'd like confirmation (and, specifically, confirmation that it can be made to work with some freely-available software package that I can script into some form of crude cooperation with asterisk). I'd greatly appreciate any other entries; I'm still dreaming of a pay-as-you-go IAX + T.38 provider (I'm guessing the T.38 would have to run over H.323 or SIP, which is no problem). First one to offer this with a reasonable level of professionalism and reliability gets my business and word-of-mouth advertising to all my clients. What would be really great (hint, hint) would be if somebody would just resell BroadVox this way -- they do PAYG if you buy carrier level amounts of service. Seems like easy money to me, especially if you can get SIP reinvites working so you're not in the media path. - a ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] request for clarification on Asterisk T.38 bounty
The bounty stands at $5,500. I'm seriously considering taking a shot at it if I can find a decent T.38 provider to test with (I'm still hoping for reliable PAYG T.38). It looks like a lot of very smart people have done a lot of very hard work (t38modem, spandsp) that would go towards getting this working. At this point it appears to be mostly a matter of integration (libspandsp+asterisk), encapsulating T.38 inside IAX2 (not too hard), and testing (tedious and time-consuming). Basically the easier but less-fun part of the big-picture task. My main question is this: how is the bounty divided? Does the person who does this grunt work get the whole $5,500, or does part of it go to the authors of t38modem/spandsp (which would surely be a large part of any solution)? I guess on one hand it would be unjust *not* to divide the bounty with them, but on the other hand, if the bounty is to be divided, I think the uncertainty about exactly how that would happen might be a factor in why the bounty has gone unclaimed for so long. - a -- PGP/GPG: 5C9F F366 C9CF 2145 E770 B1B8 EFB1 462D A146 C380 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BudgeTone 100 Woes
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote: Thanks for the assistance. I'm running version 1.0.6.7 of the software, tftp updated a few weeks ago. I'm interfacing with an Asterisk box on my local lan. My sip.conf is as follows: [100] type=friend context=home callerid=Jim 100 secret=mysecret host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 mailbox=100 disallow=all allow=ulaw allow=gsm Can you recommend a method to which I can post the configuration from the grandstream bt100 device? I think the easiest way would be to save each page of parameters as a HTML file using File/Save As... in your broswer. Then put them all in a zip file and either post it here (does the list accept small attachments?) or on a web page somewhere and post a link. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA 841 form SIPURA
On Sun, August 7, 2005 1:15 pm, Thierry Wehr said: This is not true You have to switch to last firmware and/or disable silent suppression I believe Thierry is not alone in having success with these units. I cannot explain it but my guess is that there are some inconsistencies in their hardware or something in the myriad of configuration setting is awry. I've been contacting Sipura for help with the problems but have received nothing other than try this updated firmware. I've been religiously checking for, and installing, any upgrades posted on their support site. None have addressed the handset voice quality or inaudible speaker-phone problems I am getting. I intend to keep one of the 7 I have as a test unit but will be ebay'ing the other 6; gently used, original packaging if anybody's interested ;) Cheers! -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC [EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] z-machine + asterisk = fun!
On Sunday 07 August 2005 14:45, [EMAIL PROTECTED] wrote: Now Zork is back! Listen as the eerie voice of Festival takes you into the Underground Empire, and marvel as you explore this world with your dial pad, unlocking the secrets within! Haha! Hehe, very cool. READ ALL ABOUT IT! How old text games got revitalized through VoIP! Zork author is making new fortunes as a whole new generation discovers the fun you can have on your phone! Pen and paper not included. -- List Manager Network Voice Communications, Inc. netwvcom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to connect to FWD
Hi, My asterisk server is behind firewall and i am trying to connect to FWD. i hv configured as mentioned in this link http://www.freeworlddialup.com/advanced/iax. i am able to register my server with FWD. But when i dial 393612, i always get 'No one is available to answer this time, try again later'. I hv portforwarded tcp 4569 and 5060 from my firewall to my asterisk server. Any idea what else is missing. Debug info -- Called fwd/393393612 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00015ms SCall: 02703 DCall: 2 [65.39.205.121:4569] AUTHMETHODS : 3 CHALLENGE : 207142319 USERNAME: 686928 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00098ms SCall: 2 DCall: 02703 [65.39.205.121:4569] MD5 RESULT : 8785af398932159114985608249d26ce Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00093ms SCall: 02703 DCall: 2 [65.39.205.121:4569] FORMAT : 4 -- Call accepted by 65.39.205.121 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00093ms SCall: 2 DCall: 02703 [65.39.205.121:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: HANGUP Timestamp: 01848ms SCall: 02703 DCall: 2 [65.39.205.121:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01848ms SCall: 2 DCall: 02703 [65.39.205.121:4569] -- Hungup 'IAX2/fwd/2' == No one is available to answer at this time -- Executing Goto(SIP/200-d345, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) -- Executing NoOp(SIP/200-d345, Dial failed due to NOA thanks, -B __ Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NT1 devices with analog ports on HFC based ISDN BRI cards in NT mode and asterisk (chan_mISDN)
Hi all! I got information that NT1 devices with analog ports (ab ports like on Siemens Santis or Intracom NetMod or etc.), can be used like ISDN BRI/Analog converters (i.e. for connecting analog phones or faxes to a HFC based ISDN cards in NT mode). Yesterday, I tried to do so with Intracom NetMod NT1, HFC-S PCI card, mISDN drivers and chan_mISDN in *, but without success. chan_mISDN tells that configuration is OK (NT mode PTmP) but L1 is constantly down and no dialtone on alaog phone connected to ab port of the Intracom NT1. Cable (cross ISDN cable) between NT1 and HFC card is corrcect also (I know because, when I connect other * with AVM - Fritz in TE mode to the same Intracom NT1 (with a straight ISDN cable), chan_mISDN reports that L1 on both * goes to UP). Did anyone managed to make this configuration work ? If so, any help would be apritiated. Thank you very much. Regards, Nenad Radosavljevic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BudgeTone 100 Woes
I've attached my zip file. Thanks for the help. Jim Tony Mountifield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote: Thanks for the assistance. I'm running version 1.0.6.7 of the software, tftp updated a few weeks ago. I'm interfacing with an Asterisk box on my local lan. My sip.conf is as follows: [100] type=friend context=home callerid=Jim 100 secret=mysecret host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 mailbox=100 disallow=all allow=ulaw allow=gsm Can you recommend a method to which I can post the configuration from the grandstream bt100 device? I think the easiest way would be to save each page of parameters as a HTML file using File/Save As... in your broswer. Then put them all in a zip file and either post it here (does the list accept small attachments?) or on a web page somewhere and post a link. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin 666 bt100.zip M4$L#!!0(`.=!S.U_O E\PX``'%```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`,QE8$*+H*1 MJ# KGV8-+1R\):,FT#?PTYML,4T)G.1K.G:8B!%NE=5;C.\%0YXM^/[EMAIL PROTECTED] M:'HWFSV,/V2K252V2[+K*2XTV7T?CF[C8C,6L)H.3WJ+$M7TBLF;WTW M:2EZ02]%21[EMAIL PROTECTED]D+*2G[J9FFL.K2PG=5;[EMAIL PROTECTED],6V-MVNB1 MZO]**[I$66YY8_VW//8N3-L)_*W#=+;%)61K:WK_J%(OHWVUL+VD:K3D ML*^TY+UVPG7F!YU1=C:()OXR# ]Q](VV*!T44U@/7]#?(8F/F3-[FX5Z MV[8TX-LOA78/CRB*7:IDTYAPG)0P`B1J1[N\?Z;D#_I[TT^IE91S[BI$ M=(N1IBZC4X?1KKC4[6CF9OO3S'%DJ?Z43B[VO!W 1X;X^#/P8.A1Y=\ MW1P,W\'\\D19E=5F!W0D@)-/U],VH4Y2)\CXH$4Y'Z2/TPIOZ_7K)-RM [EMAIL PROTECTED]Z[$Q_A5$`6CB:ZR,[EMAIL PROTECTED]-PIHM;9OZC1IRL+' M*5X*7OHX)5?XES5*9QSCA!F;[EMAIL PROTECTED] 7^S2[]#)6N49A. 2HGD4* MY8X0GCX1S%I9X.VQ==]0#$:.Q.3BFC;[,@UB,SO*_6:N#Z1`K26$X M0$BSSE'H!'\C*QO=$GRD;DH3*)HV33C;4_AM!,G][W^[2/- ,DZ,7S=8I M$SXSM;R(AV:S):Z!A-3%(CA?080T\I$FQ]AU:1N_$)T8+'V9[\NZXPF MI;I^?19EE.^#14IZ53A7WU%8%C7**(=K2_T.X0L:W3D[AU4DDA)2(R)KS? MD/8LYF(\[/LTCT6FAR0.`' ]WGQZ*B;Q53.LBL.]DJ(WAAT9/;3641+GR M4DJ?%1N,FN\KE9.'H$E(GTH,JI49- ?FA_?WR3*O'G#8JIH*(S.LC =!I MV\K8JR=V+EEJI4J'EL7%J+#8T:L(:,QXG!0S[EMAIL PROTECTED]#_=H' ESD/MS M*0-%RGW_'ROAM D8)'XF KEX,#00ZVZW'P`V7,B3E87HK/0)2WST'RXMQ M%B/@^T\#G9E00XJL2.LR,'!_RD'R\OM,5DT)^ @YPBI0SLY#)0D021E=3: M?AT+J!A)4^,[EMAIL PROTECTED]'%1%O6GKA?LWM!D8;\.3A8G8$B]7Q; M#O;R. @S2+MD ;U([EMAIL PROTECTED];STBZ=Q,!Q6_O,E2 MKCSPTD+4RZ6#; KC$P?_8RAL\[#VXQXZ])_;(L/$S_;91PH-O_XU\[[ M2O*2?*#=SI!P^RI09@/A $BC6?S1X+3.^EYH]=S3#)SGD-HAEDDKK#]C M`XIR^=EMM)[GCHPK1-8DF8MB B9_AH3TF*SNE'6Z-IT 4(06+FTDK?H M'(CKH.JRJOO8HPQN)%8Q!!2=#S=$V%M$,ULHZN77,P;Q[EMAIL PROTECTED] M7)CH7$\FV/_%6,;#;J(]GDJ+US9.[EMAIL PROTECTED]+M![VT1/5AF)JT=0]3P/' M@[EMAIL PROTECTED])R$6)]85*1/+#Q+GWWA_\#%W*D/VQ6T^ZATWG(P2Z:_;T. M;*GUL'A;ALJA3ZN0/H$#ZZ**W%3E9DMM=MML;?!#3U:,!*%/ZTF\5TI; ( MHWN(MN!Y(7;W\I0^09_[JUH`UKH=S(]*UX=T6[7]S;N(#=FHO%!RG M@7D1Q?KV#]@,DL[N9+RLJKU91OB^K#7WYI/HS::4MNDIT'4$?9`918 MTN(`Q)6:3.2C70XYJ$K3SX96:U.\)-HEK,?FB=YHU?7/XY%T.YZBZ2+ MJER_9,BM[C3\S]59G^E_N_XB,%PA-SHK8-G!AYN31- +_R;JV27 MCV [EMAIL PROTECTED],\-+T_' /KB!X^1H*;%:9;B!6H(1-Q8K)R7DF,T M8#W9+H.*TNZS36;$^(?#%A?J8#0#Z2B7^G@:P2037W5?'K$^P5,Q5=_Z[ MNW/@HN[;M-':M ,?[EMAIL PROTECTED]Y!VAUD:!OO)*'W3G.M#;HU-:L.!G; MJKF#3A'W7DH=-7,N2\'D6T- ET?\D1RJH[D(MW4BC^;SQ/4]U.ZW6:Y' M;.]8FYPRZ *NZ-'2;,BC%N;[EMAIL PROTECTED]:=:-!U+EV\(2;):DO+Y*XA6K(F(^MSM M/))F-2$-'GFKY!M6%WH;1[1I)3X[EMAIL PROTECTED]):CFR5(5[WWE M#1U*.Z:'=6(;7FK$V8Z^/=' $W*NGY#FL6X)D-0!(=5WFJE.%6M8\;Q6+K MV)'O:!6/3V'$D_2C5MRTRV21-4AZW#TMD$7[P%$3TCL/QH;P*P MFQ:2$' 4==!/[7;?#NECAQ%!U$*1K2#DB?I=*ND]EQ=T-$KK*[0)=N=HJ MLMJ^Z'8ZK4[X4#.T36'1B9V8D*/@[EMAIL PROTECTED];A727Z.%PJRTO#1=1+;()%9 [EMAIL PROTECTED],S5SM!;O$;*([EMAIL PROTECTED];XB(E].BLJ'[QQ-9T]CZK'@@!%U M_Y5!O:]*A%;X.)[EMAIL PROTECTED]6+D$2I\FOWG[EMAIL PROTECTED]@9,)QO)[EMAIL PROTECTED] M['8L.MDTGL!6Y6SX?K((([EMAIL PROTECTED]'B8YHJL%'(E!D$I.1@`=;!NKDPL7. M6'(7G[J!#(:FQ@:U(2XYO3V$:WI]]KR1$DP.S)V,'_#$PWO910#Z9XE0H M74K\.%,65HPHK7Q8?7IB^8X6',!UI3(Y[EV3,]#-^.%[TUV_:K3)V=) M75ZM59OTB$M1N6,[XR_F!J:W^BHYB\1THLJRKQP1EV]_O_-N)_A9_8O9J
[Asterisk-Users] VoicePulse Connect down Sunday evening?
It appears that incoming calls (IAX) through voicepulse are being rejected... anyone else experiencing this? -Trent Tuggle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse DTMF Problems Anyone?
http://bugs.digium.com/view.php?id=4631 FYI I am experiencing the same, but due to lack of cooperation from ITSP am not able to proceed with debugging it. Feel free to pursue... regards, mark On 8/8/05, Justin Richards [EMAIL PROTECTED] wrote: I too have been having inbound dtmf problems with VP Connect using iax2 for inbound. When I switched to sip, and added the relaxdtmf=yes, all 10 inbound test calls I did seemed to work fine for dtmf. I'm going to leave my config set up to use sip for inbound VP Connect calls for a while and see how if functions. thanks for the relaxdtmf tip Umair. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- regards, Mark P. Edwards FWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] http://www.voip-info.org/ front page taken out by spammer
Today the front page of http://www.voip-info.org/ was taken out by a spammer. It also seem the history page for http://www.voip-info.org/ was also nuked. I've restored the best I could using google cache, but still missing some information. Who is an admin on http://www.voip-info.org/ and can fix it? PB ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: BudgeTone 100 Woes
Jim Duda wrote: I've attached my zip file. Thanks for the help. Jim Tony Mountifield [EMAIL PROTECTED] wrote in message Try using the IP address of the server instead of the name. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] request for clarification on Asterisk T.38 bounty
Adam Megacz wrote: The bounty stands at $5,500. I'm seriously considering taking a shot at it if I can find a decent T.38 provider to test with (I'm still hoping for reliable PAYG T.38). It looks like a lot of very smart people have done a lot of very hard work (t38modem, spandsp) that would go towards getting this working. At this point it appears to be mostly a matter of integration (libspandsp+asterisk), encapsulating T.38 inside IAX2 (not too hard), and testing (tedious and time-consuming). Basically the easier but less-fun part of the big-picture task. t38modem is of little use for this. It is purely a terminating program. spandsp is, of course, applicable as its modems are a core requirement. Doing a quick botch up of T.38 isn't too hard. A solid reliable implementation takes considerably more effort. Some real R as well as D is needed to do it properly. The bounties give no indication of criteria for judging completeness. My main question is this: how is the bounty divided? Does the person who does this grunt work get the whole $5,500, or does part of it go to the authors of t38modem/spandsp (which would surely be a large part of any solution)? I think you should forget these bounties. There is nobody administering them, so I think the chances of a payout are minimal. I guess on one hand it would be unjust *not* to divide the bounty with them, but on the other hand, if the bounty is to be divided, I think the uncertainty about exactly how that would happen might be a factor in why the bounty has gone unclaimed for so long. It has gone unclaimed for so long because the problem is not trivial, and I have been too busy with other things to complete my implementation. It has been sitting here half finished since the beginning of the year. Passthrough is simple, but the interesting things are termination, and PSTN gateway operation. The code I have, tidied up, would provide UDPTL-to-UDPTL passthrough operation for SIP, which many would find useful. Maybe I should tidy and commit it as an interm step. It implements the UDPTL transport, with full FEC handling, and offer some simple botches to sip.c to make it udptl and T.38 aware. I have most of a gateway and termination implementation, too, but it isn't close to being ready to commit. I find sip.c is currently too messy to produce anything more than a botch for it. A couple of people have said they are reworking sip.c to make the addition of new codecs, transports, etc. and their renegotiation function smoothly. I haven't seen any results so far. I did only minimal work on sip.c in the hope that one those efforts would bear fruit in * 1.2. As with many things in *, the licencing forced me to do rather more work than necessary. If * were GPL, I could have used some GPL'ed ASN.1 code I found. To make code that could be committed to CVS I had to spend quite some time rolling my own routines. The final result is faster, but it took a lot more effort. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] request for clarification on Asterisk T.38 bounty
Steve Underwood wrote: produce anything more than a botch for it. A couple of people have said they are reworking sip.c to make the addition of new codecs, transports, etc. and their renegotiation function smoothly. I haven't seen any results so far. I did only minimal work on sip.c in the hope that one those efforts would bear fruit in * 1.2. This is a confirmed situation; once the 1.2-rc has been rolled out (meaning the 1.2 CVS branch has been created), Olle and I will be starting a new version of chan_sip, taking into account all that has been learned in the last 12+ months regarding protocol conformance, interoperability, modularity and other issues. One of the big items on that list is to make chan_sip have _zero_ assumptions about what sort (and what number) of media streams will be associated with a session, so that adding support for alternative media streams will be significantly more likely (still not trivial, obviously). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
Thank you very much kumara - Original Message - From: MF Hulber [EMAIL PROTECTED] To: Kumara Jayaweera [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 07, 2005 10:24 PM Subject: Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards Here's my kernel info: Linux asterisk.hulber.com 2.6.9-11.EL #1 Fri May 20 18:17:57 EDT 2005 i686 i686 i386 GNU/Linux And my Asterisk version: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-05 21:21:13 UTC Kumara Jayaweera wrote: Hi MARK, Thanks a lot for the reply. my box is Intel based. and there is no USB conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may be the place to see. Thank you Kumara - Original Message - From: MF Hulber [EMAIL PROTECTED] To: Kumara Jayaweera [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 06, 2005 4:02 PM Subject: Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards I don't use Fedora but I do use RHEL AS 4 without any problem. Do you have any USB conflicts? MARK. Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list of T.38 providers on wiki: please contribute
I have a NY 212 packet8 service if you would like to work with me to set this up on my [EMAIL PROTECTED] service, I'm happy to test this with you. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Megacz Sent: Sunday, 7 August 2005 5:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] list of T.38 providers on wiki: please contribute Please excuse my ignorance but doesn't the VOIP/PSTN gateway (Broadvoice,VP Connect) have to support T38 in order for an T38 supported ATA to do any good ? Yes. BroadVox supports it. I've started a wiki page to track providers who do, mainly because such providers are so hard to find, and the available information out there can be so misleading and unreliable: http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers+T.38 Currently even providers that *do* offer T.38 rarely advertise that fact on their website (for example, I can't find anything about T.38 on BroadVox's site, and we support faxing is meaningless). I've heard rumors that Packet8 and SunRocket support T.38, but I'd like confirmation (and, specifically, confirmation that it can be made to work with some freely-available software package that I can script into some form of crude cooperation with asterisk). I'd greatly appreciate any other entries; I'm still dreaming of a pay-as-you-go IAX + T.38 provider (I'm guessing the T.38 would have to run over H.323 or SIP, which is no problem). First one to offer this with a reasonable level of professionalism and reliability gets my business and word-of-mouth advertising to all my clients. What would be really great (hint, hint) would be if somebody would just resell BroadVox this way -- they do PAYG if you buy carrier level amounts of service. Seems like easy money to me, especially if you can get SIP reinvites working so you're not in the media path. - a ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to connect to FWD
Correction. i hv port forwarded udp 4569 and 5060. -B --- Balaji NJL [EMAIL PROTECTED] wrote: Hi, My asterisk server is behind firewall and i am trying to connect to FWD. i hv configured as mentioned in this link http://www.freeworlddialup.com/advanced/iax. i am able to register my server with FWD. But when i dial 393612, i always get 'No one is available to answer this time, try again later'. I hv portforwarded tcp 4569 and 5060 from my firewall to my asterisk server. Any idea what else is missing. Debug info -- Called fwd/393393612 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00015ms SCall: 02703 DCall: 2 [65.39.205.121:4569] AUTHMETHODS : 3 CHALLENGE : 207142319 USERNAME: 686928 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00098ms SCall: 2 DCall: 02703 [65.39.205.121:4569] MD5 RESULT : 8785af398932159114985608249d26ce Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00093ms SCall: 02703 DCall: 2 [65.39.205.121:4569] FORMAT : 4 -- Call accepted by 65.39.205.121 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00093ms SCall: 2 DCall: 02703 [65.39.205.121:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: HANGUP Timestamp: 01848ms SCall: 02703 DCall: 2 [65.39.205.121:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01848ms SCall: 2 DCall: 02703 [65.39.205.121:4569] -- Hungup 'IAX2/fwd/2' == No one is available to answer at this time -- Executing Goto(SIP/200-d345, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) -- Executing NoOp(SIP/200-d345, Dial failed due to NOA thanks, -B __ Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to connect to FWD
Balaji NJL wrote: -- Call accepted by 65.39.205.121 (format ulaw) -- Format for call is ulaw It's working fine.. == No one is available to answer at this time -- Executing Goto(SIP/200-d345, s-NOANSWER|1) in new stack ..but you're not answering the phone, or it's offline. Try looking at your SIP debug. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] function declaration isn't a prototype
hi, thank you vary much for the updates, i jsut got the latest from cvs and the error is fixed, however, i got this new error, when running make, /usr/local/sparc-sun-solaris2.8/bin/ld: cannot find -lncurses collect2: ld returned 1 exit status make: *** [asterisk] Error 1 pls advise. much thnks. chris. - Original Message - From: Steve Drach [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 06, 2005 12:04 AM Subject: Re: [Asterisk-Users] function declaration isn't a prototype In file included from include/asterisk/utils.h:26, from term.c:32: include/asterisk/strings.h:232: parse error before `va_list' include/asterisk/strings.h:232: warning: function declaration isn't a prototype make: *** [term.o] Error 1 pls advise on how i can fix this, It's fixed in CVS head now, Do an update. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
