[asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asteriskt38.com
Hi, fax gateway isn't just a packet bridging. It does the mediation between T30 (voice) - T38 (fax over ip) protocols. It does work for asterisk 1.4, asterisk 1.6, asterisk svn head. If it doesn't please send me a bug report and I'm going to fix it. Best regards Daniel. On Mon, Oct 6, 2008 at 7:04 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote: That isn't real T.38 support, it's just Packet2Packet bridging that works correctly. Still need to use a Cisco gateway to support sending the faxes somewhere on the PSTN. But it does work and it is reliable, I use it every day. On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote: Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive. Hopefully it works. The one in CallWeaver doesn't. On Mon, Oct 6, 2008 at 8:12 AM, Daniel Ferenci [EMAIL PROTECTED] wrote: and there is a new application called fax gateway (http://bugs.digium.com/view.php?id=13405) that can do gatewaying between T30 and T38 and vice versa. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Klaverstyn, David C schrieb: Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. Hello, maybe you should install mysql-client too ;) best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Yes, unfortunately, VOIP wiki did not mention about installing mysql-client which it should have been. Without mysql-client, you cannot change passwords, grants, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stefan Schmidt Sent: Tuesday, 7 October 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons- 1.6.0 Klaverstyn, David C schrieb: Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. Hello, maybe you should install mysql-client too ;) best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 ???
Hi, I agree with Gordon. We are still using Asterisk 1.2 because we are waiting for Asterisk 1.4 features to work as for Asterisk 1.2 (it seems to us that parking and queues have some problems... so not good enough for production). Giorgio Incantalupo Gordon Henderson wrote: On Mon, 6 Oct 2008, Alejandro Facultad wrote: Dear all, I know there are two actual versions of Asterisk: 1.4 and 1.6. There is also 1.2. It may not be supported but there are 1000's of people out there (myself included) who are still using it. My scenario is: SIP server with 100-150 SIP users, voice mail and maybe IVR. I will use GSM audio codec. Maybe in the future I'll connect a E1 line to the PSTN. What Asterisk version is better to me and why ??? The answer you are looking for is that you should be using a supported, stable version, and right now, 1.4 is the only one that fits. If I were starting today, I'd go with 1.4. But I have to ask: Why GSM? If everything is in-house on the same LAN, then why not G711a? E1 is G711a, so you'd have to get the box to transcode to G711, which depending on the number of calls and CPU, might be an issue... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
Hi! I have a different approach. I wrote a small application which simply starts an audio player. You can write a very small script to answer fast or just use telnet like this: telnet localhost 8642 At the moment everything is hardcoded, but can be changed in any case. I use 15s ring-time, telnet-port 8642 and mplayer as the audio-player. Short note: You don't have to submit anything over telnet, just connect. If you're interested in this solution I'll upload the code and give you a dialplan example (it's based in the return code of the program. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk, phpagi and singleton
2008/10/6 Steve Edwards [EMAIL PROTECTED] On Mon, 6 Oct 2008, Alex Balashov wrote: Giedrius Augys wrote: What tools and programming (scripting) language do you use for FastAGI? Whatever languages FastAGI APIs are available for. You are pretty much limited to languages whose interpreter lends itself to invocation as a standalone daemon, which may or may not exclude PHP and other languages designed to be web scripting languages and whose state is expected to be determined in terms of serial HTTP requests. I use Perl, personally: While not an interpreted scripting language, I would use C :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Steve, Do you use CAGI for fast AGI ? -- Pagarbiai / Best Regards, Giedrius Augys ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automatic call pickup
Hi, Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user wants to pick up a call within his/her pickup group, *8 must be dialed (or whatever you define in features.conf). However, these users were used to another behavior when they had a commercial PBX (Bosch). When a phone of the pickup group rang all the user had to do to grab the call was to pick his/her phone up (no dialing required). I tried to explain the advantages of deciding if you want to pick up the call or not by pressing *8 but I understand that in a very busy environment, dialing *8 (or whatever) makes you lose a second or two... Making the phones ring at the same time (for the same call) isn't an option (Dial(SIP/101SIP/102)). Extensions within a pickup group must ring one at a time. They are all using standard analog phones connected to a multi-port ATA. I was thinking of configuring some sort of auto speed dial of the pickup code (*8) whenever the user picks the phone up but it seems that these phones don't support that. I know this is a weird question but has anyone dealt with this issue in Asterisk 1.2/1.4/1.6? Or has anyone come up with a custom solution? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
Jim Boykin wrote: I know about those packages. Questions is how do we use those packages to build our own RPM. We use asterisk SVN trunk. asterisk usually comes with asterisk.spec and make target rpm. With some slight modifications on the spec file you can pretty much build whatever you need into the package. BR -- Dobry Thanks Jim On Mon, Sep 29, 2008 at 4:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 29, 2008 at 03:51:35PM +0530, Jim Boykin wrote: We use RHEL5, FC6, CentOS5. I will be happy to hear your inputs for any distribution you know. Fedora 9 has a package, but I think it is asterisk 1.6.0-rc9. Some SRPMs of lesser quality for Centos 5: http://yum.trixbox.org/centos/5/SRPMS/ http://yum.trixbox.org/centos/5/SRPMS/repodata/repoview/A.group.html http://yum.trixbox.org/centos/5/SRPMS/repodata/repoview/asterisk-0-1.4.21.2-2.html http://repo.elastix.org/centos/5/updates/SRPMS/repodata/ http://repo.elastix.org/centos/5/updates/SRPMS/repodata/repoview/A.group.html http://repo.elastix.org/centos/5/updates/SRPMS/repodata/repoview/asterisk-1-1.4.21.2-3.html (Elastix's developers have this funny habbit of making the path leading to that directory non-indexed) The lesser quality shows e.g. in the fact that the changelog is not always updated. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL and swap from macros to contexts
Atis Lezdins wrote:
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c line:2521 func: check_pval_item Warning: file
/etc/asterisk/extensions.ael, line 36-36: application call to Gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!
Hi,
In definition use:
macro set_record(A,B) {
// do something
}
And for calling:
set_record(${CALLERID(NUM)},${EXTEN});
It will automatically be translated to GoSub in 1.6, but will remain
as Macro in 1.4.
yes, I know, but I hear on IRC, that macros will be deprecated and
suggestion was to move to contexts,
personaly I would like also move away from macros, because macros have
some limitations, eg. variable number of arguments isn't possible with
classic macros,
macros also require variable to be defined in macro definition (that is
needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
so I definitively agree with moving from macros to contexts, only one
bad thing is compiler warning, when I try to Gosub to context (as macro
replacement)
PJ
Pavel--
Yes, you can ignore the warnings and go ahead and hardcoded gosub calls
into your source. I didn't upgrade 1.4 to use gosub-instead-of-macro
because
the key element ended up being calling gosub with arguments, which
didn't
make it into 1.4.
Someday, when you upgrade from 1.4 to 1.6, you will have to change
all your gosub's to use the argument passing feature, if you hardcode
gosubs now. Or, you can backport the gosub-with-arguments feature to
1.4,
and use 1.6 AEL to compile... which will give you some future
portability
when you do move to 1.6...
Sorry to make simple things sound so complicated!
murf
murf, thank you for clear answer,
currently, I'm using asterisk trunk (and 1.6 also),
do you plan to remove quite confusing AEL warnings, that appears, when I
try to hardcode Gosub with arguments into ael dialplan?
Why would you still want to hardcode them?
because I would like to move completely away from using classic macros,
because it have some limitations, as I said, variable number of
arguments passed to macro is example,
so I moving from macros to contexts that do the same functionality and
haven't limitations that macros have
and if I will have only contexts in ael dialplan I must call it with
Gosub (I can't call context using )
PJ
Please see above sample, you can use prefixing with and ().
Regards,
Atis.
PJ
Regards,
Atis
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Re: [asterisk-users] PoE switch recommendations?
Hi Ken, we are quite satisfied with Linksys SRW248G4P. 48 port PoE, 4 GB uplinks and 2 GBIC slots. VLAN, QoS and all the like is on board. Around US$600 I guess. Only drawback in my opinion is that they are loud like a starting airplane. You definately don't want them next to your desk. ;-) Christian Ken D'Ambrosio schrieb: Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- victormedia jahnstraße 105 40215 düsseldorf germany fon +49 211 5833434 fax +49 211 5833435 sip [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ldap usage in 1.6.0
Brendan Martens wrote: Having thought some more about my issue I think I can perhaps ask my question more succinctly: is it possible to get dynamic (or realtime) data from ldap within the various .conf files? If there is not a convenient function for getting this in the .conf files, what if I somehow specified a global variable within the res_ldap.conf and referenced that value inside the other .conf files? Is this possible? Sorry if these are very basic questions, I just haven't been able to find answers to them. : ( Here, I've written a perl script that rewrites the actual sip.conf itself (as well as generates a custom myexten.conf file, which is included in the main extensions.conf file.) The perl script can read from whatever datasource you setup - right now, I read from a MySQL database of users, but I know perl can read from an LDAP directory as well. This way, perl sits between Asterisk and the database/directory and does the mapping/translation required, giving more complete control over the sync process. Of course, I wrote this script in the pre-1.2 days of Asterisk, but its still running fine on 1.4.* that I've got in production right now. Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 ???
Brendan Martens [EMAIL PROTECTED] writes: On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote: The answer you are looking for is that you should be using a supported, stable version, and right now, 1.4 is the only one that fits. If I were starting today, I'd go with 1.4. 1.6.0 has just been released. Personally I'd start with that because then you don't stuck with generation old features, and as you are just starting you aren't locked into any feature sets or syntax issues, etc. I completely agree with Brendan here. 1.6 is new and undoubtedly buggy (although we haven't been hit by anything serious yet...), but the code quality is higher overall. Also, the code is fresh in the minds of the developers, so they can fix bugs faster. 1.4 isn't new, but it still has its share of bugs -- look at Mantis. The advantage of 1.4 and especially 1.2 is that you can look the bugs up in Mantis before you hit them. For your first deployment you will probably end up spending several months in testing. During those months 1.6 will stabilize a lot. If you hit a really bad bug, you can switch to 1.4 reasonably quickly -- they aren't THAT different. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ldap usage in 1.6.0
Here, I've written a perl script that rewrites the actual sip.conf itself (as well as generates a custom myexten.conf file, which is included in the main extensions.conf file.) I was hoping to keep it all native to asterisk, but I would be willing to give that a try. Where can I get this script? Brendan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Hi, in tftp server I have the followings files: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads ..and on 7906g in status menu I have: load file: sccp11.8-3-2s app load id: jar11sccp.8-3-1-22.sbn jvm load id: cvm11sccp.8-3-1-22.sbn os load id: cnu11.8-3-1-22.sbn boot load id: tnp06.3-0-1-31.bin dsp load id: dsp11.8-3-1-22.sbn I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? Thanks in advance. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 07, 2008 1:04 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Are you sure you have set the 7960 to SIP? By default they use SCCP, so you need to go through the process of changing them over, which ideally would just be done with the edits you have already in the load files but generally means going back to an early version of the SIP code then working upwards from there. You can check the current hardware in the status, if its SIP it will be something like POS-0806... (I haven't got a phone handy to check) but there is a reasonable amount of info on voipinfo about the process Cheers Duncan Sasa wrote: Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: loadInformationSIP11.8-0-4SR1S/loadInformation ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving CTLSEPmacaddress.tlv to 192.168.0.155:49152 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving SEPmacaddress.cnf.xml to 192.168.0.155:49153 ..and in asterisk CLI I have: -- Starting Skinny session from 192.168.0.155 Device SEPmacaddress is attempting to register Now when 7906G started is loaded: load file: sccp11.8-3-2s boot load id: tnp06.3-0-1-31.bin ..why isn't loaded sip firmware ?? Thanks in advance. -- Salvatore. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL and swap from macros to contexts
On Tue, Oct 7, 2008 at 2:20 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c line:2521 func: check_pval_item Warning: file
/etc/asterisk/extensions.ael, line 36-36: application call to Gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!
Hi,
In definition use:
macro set_record(A,B) {
// do something
}
And for calling:
set_record(${CALLERID(NUM)},${EXTEN});
It will automatically be translated to GoSub in 1.6, but will remain
as Macro in 1.4.
yes, I know, but I hear on IRC, that macros will be deprecated and
suggestion was to move to contexts,
personaly I would like also move away from macros, because macros have
some limitations, eg. variable number of arguments isn't possible with
classic macros,
macros also require variable to be defined in macro definition (that is
needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
so I definitively agree with moving from macros to contexts, only one
bad thing is compiler warning, when I try to Gosub to context (as macro
replacement)
PJ
Pavel--
Yes, you can ignore the warnings and go ahead and hardcoded gosub calls
into your source. I didn't upgrade 1.4 to use gosub-instead-of-macro
because
the key element ended up being calling gosub with arguments, which
didn't
make it into 1.4.
Someday, when you upgrade from 1.4 to 1.6, you will have to change
all your gosub's to use the argument passing feature, if you hardcode
gosubs now. Or, you can backport the gosub-with-arguments feature to
1.4,
and use 1.6 AEL to compile... which will give you some future
portability
when you do move to 1.6...
Sorry to make simple things sound so complicated!
murf
murf, thank you for clear answer,
currently, I'm using asterisk trunk (and 1.6 also),
do you plan to remove quite confusing AEL warnings, that appears, when I
try to hardcode Gosub with arguments into ael dialplan?
Why would you still want to hardcode them?
because I would like to move completely away from using classic macros,
because it have some limitations, as I said, variable number of
arguments passed to macro is example,
so I moving from macros to contexts that do the same functionality and
haven't limitations that macros have
and if I will have only contexts in ael dialplan I must call it with
Gosub (I can't call context using )
I think you didn't understood, that declaring macro x and calling it
with x() would make AEL parser to do it for you. They are called
macro just in AEL, but internally they are GoSub's. Additionally you
will be ready for any other future changes.
For example You can use
$aelparse -d -n -w -q extensions.ael
and take a look at generated .conf file. In 1.6.0 it would be:
[set-record]
exten = s,1,Set(LOCAL(A)=${ARG1})
exten = s,2,Set(LOCAL(B)=${ARG2})
...
exten = s,20,Return()
And call to it:
Gosub(set_record,s,1(${CALLERID(num)},${EXTEN}))
Regards,
Atis
PJ
Please see above sample, you can use prefixing with and ().
Regards,
Atis.
PJ
Regards,
Atis
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Re: [asterisk-users] Creating Asterisk Binary Package
Jim Boykin wrote: I know about those packages. Questions is how do we use those packages to build our own RPM. We use asterisk SVN trunk. asterisk usually comes with asterisk.spec and make target rpm. With some slight modifications on the spec file you can pretty much build whatever you need into the package. You can try checkinstall. It makes a package (it supports a few kinds, rpm being one of them) out of the software you compiled. Basically instead of finishing with make install you just do checkinstall and it will make a package and then use your packaging system to install it. I use this often for Debian and it works very well there. You're distribution very likely has checkinstall available in it's main repository. If not the website is here for more info: http://www.asic-linux.com.mx/~izto/checkinstall/ Brendan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Efax from Agi script
Hi all, i wrote a script agi, sking for a code, after that it sends an email now i need to send a fax... any hints or tips for that? Ty in advance. -- Riccardo Cupardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
Are you dialing a 1 before every number? That is required unless you make another pattern match. exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) Then it becomes 10-digit dialing without the need to dial a 1. If that doesn't work open up the asterisk console and attempt to make a call and reply with any error messages. I was not adding the 1 before the number but that didn't help. I opened the console 'asterisk -r' but when attempting to call out nothing happened. Is there some type of logging level that needs to be turned up? When I call in which does still work I do get the following errors and of course voicemail doesn't work.: Oct 7 09:38:08 WARNING[6146]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for '102' Oct 7 09:38:18 WARNING[6146]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Thanks again for the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] include in the DAHDI system.conf file and chan_dahdi.conf
Are includes supported in the file /etc/dahdi/system.conf link you can include in say a sip.conf What about in chan_dahdi.conf? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] include in the DAHDI system.conf file and chan_dahdi.conf
On Tue, Oct 07, 2008 at 10:08:25AM -0400, Jerry Geis wrote: Are includes supported in the file /etc/dahdi/system.conf no. Note that '#' begins a comment in system.conf . link you can include in say a sip.conf What about in chan_dahdi.conf? Yes, just like any Asterisk configuration file. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: text/plain
SIP [EMAIL PROTECTED] writes: The truth is there are plenty of email clients that CAN decode Hotmail messages, and if you choose one that can't, you can't blame anyone but yourself. The truth is that there are no Netcom^WAOL^WHotmail users who write anything worth reading. I had just forgotten to add hotmail to my killfile for Asterisk, because so few use it these days. The problem has now been rectified. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
I have just confirmed that they may be having a problem with double NAT. They have two ATAs, and they have two different DSL connections. One set-up goes from the first DSL Modem (NAT Wirless are disabled on the DSL Modems) to a Linksys WRT110 and then there is a WRT54G hooked in to the 110 that has the ATA plugged into it. The other ATA is configured from a DSL Modem (again, I was told NAT Wireless were disabled on the modem) to a WRT600N and the ATA is plugged in there. I have the same issues on both ATAs. I have no idea why their network is as poorly designed as it is, the bad part is I have to make sure the phones work there and try to troubleshoot from 3000 miles away. Any work arounds for a problem because of double NAT? A quick and dirty solution for them to get their phones working right? Steve Anness On 10/7/08 2:12 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] regcontext
On Tue, 2008-10-07 at 12:03 +0800, Nhadie wrote: just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent Linksys/SPA942-5.2.8 for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext It shouldn't be added when it's *called*, it should be added when the phone *registers*. What happens is that if regcontext and regexten are set for a particular SIP peer, Asterisk automatically creates an extension (the name controlled by the regexten setting) in the proper context (controlled by the regcontext setting) with a priority of 1 and NoOp() as the application. If this is happening at the time the phone is called and not at the time the phone registers, then this would appear to me to be a bug. Based on the other comment in your log file above (the Saved useragent messsage), I'm guessing that it really is happening at registration time. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: headsets
On Sun, Oct 5, 2008 at 3:30 PM, Bill Michaelson [EMAIL PROTECTED] wrote: The IP330 has a subminiature jack for headset/mic combos. Are there quality headsets anyone would recommend for in-office use for heavy users with these phones? Using any wiring path? I've tried a cell phone earphone/mic, and it sounds OK, but it's flimsy for this application. We've started switching out our Plantronics M-series amplifiers and headsets with headsets from Jabra. Some of our agents have a problem with their M-series amps where they get no audio when they pick up using the phone's headset button, and they have to quickly go on-hook and off-hook again with the amp. We also had some S11 sets, with which we experienced horrible echo. The main upshot for us is that the Jabra headsets don't require an external amp, so they're simpler to install and cost less. We're using them on Polycom 330, 550, and 650 sets, and the audio quality is great. The have the usual quick-disconnect, so they're appropriate for a call center (though the connector is not compatible with Plantronics). They have adapters for both 2.5mm and RJ-8 modular plugs, so they can be used on any Polycom IP phone with a dedicated headset jack. Feel free to contact me off-list if you'd like the part numbers we're using. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
I recently purchased a few SRW208P switches. They work fine. If you run Windows. Granted a lot of people run windows instead of Mac or Linux, but be aware (to those looking) that the SRW line of switches REQUIRE Internet Explorer on Windows. The support site says it is recommended, but even the login page does not work properly on anything but IE on Windows. For me, as a Mac user, it is enough to not buy any more of those ever again. That's very strange, I've used FF2 and 3 under Linux plenty of times to configure the SRW224P units. I'd have thought the web interfaces would be pretty similar between the models. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I believe. Just to make sure, you are doing a 'module reload' each time you make changes to configuration files right? Cool I've got voicemail :-). I am reloading it and have increased the logging level. When dialing out I'm seeing: -- Executing Dial(SIP/101-08183018, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-0818b178 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Oct 7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Think it's a problem with vitelity? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
Brendan Martens wrote: Jim Boykin wrote: I know about those packages. Questions is how do we use those packages to build our own RPM. We use asterisk SVN trunk. asterisk usually comes with asterisk.spec and make target rpm. With some slight modifications on the spec file you can pretty much build whatever you need into the package. You can try checkinstall. It makes a package (it supports a few kinds, rpm being one of them) out of the software you compiled. Basically instead of finishing with make install you just do checkinstall and it will make a package and then use your packaging system to install it. I use this often for Debian and it works very well there. You're distribution very likely has checkinstall available in it's main repository. If not the website is here for more info: http://www.asic-linux.com.mx/~izto/checkinstall/ Hey thanks, didn't knew about that, it worked right out of the box! BR Dobry Brendan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL and swap from macros to contexts
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c line:2521 func: check_pval_item Warning: file
/etc/asterisk/extensions.ael, line 36-36: application call to Gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!
Hi,
In definition use:
macro set_record(A,B) {
// do something
}
And for calling:
set_record(${CALLERID(NUM)},${EXTEN});
It will automatically be translated to GoSub in 1.6, but will remain
as Macro in 1.4.
yes, I know, but I hear on IRC, that macros will be deprecated and
suggestion was to move to contexts,
personaly I would like also move away from macros, because macros have
some limitations, eg. variable number of arguments isn't possible with
classic macros,
macros also require variable to be defined in macro definition (that is
needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
so I definitively agree with moving from macros to contexts, only one
bad thing is compiler warning, when I try to Gosub to context (as macro
replacement)
PJ
Pavel--
Yes, you can ignore the warnings and go ahead and hardcoded gosub calls
into your source. I didn't upgrade 1.4 to use gosub-instead-of-macro
because
the key element ended up being calling gosub with arguments, which
didn't
make it into 1.4.
Someday, when you upgrade from 1.4 to 1.6, you will have to change
all your gosub's to use the argument passing feature, if you hardcode
gosubs now. Or, you can backport the gosub-with-arguments feature to
1.4,
and use 1.6 AEL to compile... which will give you some future
portability
when you do move to 1.6...
Sorry to make simple things sound so complicated!
murf
murf, thank you for clear answer,
currently, I'm using asterisk trunk (and 1.6 also),
do you plan to remove quite confusing AEL warnings, that appears, when I
try to hardcode Gosub with arguments into ael dialplan?
Why would you still want to hardcode them?
Please see above sample, you can use prefixing with and ().
Regards,
Atis.
PJ
Regards,
Atis
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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
Julien, I would love to see this solution so please upload the code. Thank you very much. robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julien Claassen Sent: Tuesday, October 07, 2008 4:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging Hi! I have a different approach. I wrote a small application which simply starts an audio player. You can write a very small script to answer fast or just use telnet like this: telnet localhost 8642 At the moment everything is hardcoded, but can be changed in any case. I use 15s ring-time, telnet-port 8642 and mplayer as the audio-player. Short note: You don't have to submit anything over telnet, just connect. If you're interested in this solution I'll upload the code and give you a dialplan example (it's based in the return code of the program. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call center
I would like to make asterisk call center , I have a ET410P card what I need to install like packages ? where i can get a best documents to doing it ? thank you for advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. The fact that they work initially is probably a clear indication that the NAT bindings are closing up after a few minutes. In some cases it does not matter that you have qualify=yes, since the router only keeps bindings open if the traffic is being generated from the inside-outside. Your solution would be to enable the keep-alive settings on the PAP2 and set it low to something like 15 seconds. The setting is under the tab of line 1 and line 2 and its called NAT Keep Alive Enable. Andres http://www.neuroredes.com do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
Well, after very quickly making a test call it's not Vitelity. It could be something with your account? Might want to try opening a support ticket. If you want, create a sub account and e-mail me off list the username and password and I'll test it with my box or vice versa. On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I believe. Just to make sure, you are doing a 'module reload' each time you make changes to configuration files right? Cool I've got voicemail :-). I am reloading it and have increased the logging level. When dialing out I'm seeing: -- Executing Dial(SIP/101-08183018, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-0818b178 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Oct 7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Think it's a problem with vitelity? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Callerid Help Needed
Hi All, I need some help to about override callerid, if i get blocked callerid and also having privacy=full. i am trying to override callerid on that call, but the callerid is not changed The sip trace is given below INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-23a4ba1;rport From: Anonymous sip:[EMAIL PROTECTED];tag=89cc6491fcf8ae21o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: sip:[EMAIL PROTECTED];screen=yes;privacy=full;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: Anonymous sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 308 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, x-sipura Content-Type: application/sdp can any body help me to over ride the callerid? Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
On Oct 7, 2008, at 4:19 AM, Chris Bagnall wrote: I recently purchased a few SRW208P switches. They work fine. If you run Windows. Granted a lot of people run windows instead of Mac or Linux, but be aware (to those looking) that the SRW line of switches REQUIRE Internet Explorer on Windows. The support site says it is recommended, but even the login page does not work properly on anything but IE on Windows. For me, as a Mac user, it is enough to not buy any more of those ever again. That's very strange, I've used FF2 and 3 under Linux plenty of times to configure the SRW224P units. I'd have thought the web interfaces would be pretty similar between the models. I have not personally tried using FF under Linux with these, though I ran across a number of posts that say it doesn't work. I know FF2 and the latest FF3 don't work under Mac (don't work for the SRW that is) and I know they don't work on Windows. (Linksys' official statement is to use that ietab plugin that embeds IE in a firefox tab). I would expect FF to behave the same as far as what works and doesn't in all 3 environments, but maybe not. I'll install FF3 on my Linux server and try as that would be more convenient than firing up Parallels everytime I need to change a config option in the switch. On Win/Mac it lets you log in but the main menu screen is blank, nothing to click on, just the background template. *shrug* Seeing as we already need more than the 8 ports I think I'll stick to the 24/48 port versions anyway. Regards, Chris Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automatic call pickup
Am Dienstag, den 07.10.2008, 01:42 -0700 schrieb Vieri: Hi, Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user wants to pick up a call within his/her pickup group, *8 must be dialed (or whatever you define in features.conf). [...] I was thinking of configuring some sort of auto speed dial of the pickup code (*8) whenever the user picks the phone up but it seems that these phones don't support that. Hi Vieri, regarding your combination of analog phones and ATAs I would look for the auto-dial functionality in the ATA. I am pretty sure I saw it in one web-interface or the other, but surely not all vendors implement that kind of functionality. In your place I would also think about using a one-press pickup code, like #. I know this code is often in use for transfer or the like, but if pickup is the 95%+ action then transfer doing *# instead of # (or whatever) might be reasonable. This would reduce pickup to lifting the handset and pressing the bottom left-most key, which can be done without looking at the keypad. One last idea: Perhaps your multi port ATA supports different kind of ring codes (once short, twice short, no idea whatever) one of which will _not_ ring the phones (which could interpret that signal meant as a short ring as line noise or the like). Perhaps they even support silent ringing, not sending the ring signal at all, but nevertheless answering the line if hook-up happens. Best regards Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automatic call pickup
Vieri wrote: Hi, Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user wants to pick up a call within his/her pickup group, *8 must be dialed (or whatever you define in features.conf). However, these users were used to another behavior when they had a commercial PBX (Bosch). When a phone of the pickup group rang all the user had to do to grab the call was to pick his/her phone up (no dialing required). I tried to explain the advantages of deciding if you want to pick up the call or not by pressing *8 but I understand that in a very busy environment, dialing *8 (or whatever) makes you lose a second or two... Making the phones ring at the same time (for the same call) isn't an option (Dial(SIP/101SIP/102)). Extensions within a pickup group must ring one at a time. They are all using standard analog phones connected to a multi-port ATA. If your ata support custom ringtones /distinctive ring feature, you can try to dial all phones in group, but let only one phone ring normaly, and on remaining phones set some mute ringtone via dialplan using SIPAddHeader(Alert-Info: something) PJ I was thinking of configuring some sort of auto speed dial of the pickup code (*8) whenever the user picks the phone up but it seems that these phones don't support that. I know this is a weird question but has anyone dealt with this issue in Asterisk 1.2/1.4/1.6? Or has anyone come up with a custom solution? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Are you sure you have set the 7960 to SIP? By default they use SCCP, so you need to go through the process of changing them over, which ideally would just be done with the edits you have already in the load files but generally means going back to an early version of the SIP code then working upwards from there. You can check the current hardware in the status, if its SIP it will be something like POS-0806... (I haven't got a phone handy to check) but there is a reasonable amount of info on voipinfo about the process Cheers Duncan Sasa wrote: Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: loadInformationSIP11.8-0-4SR1S/loadInformation ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving CTLSEPmacaddress.tlv to 192.168.0.155:49152 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving SEPmacaddress.cnf.xml to 192.168.0.155:49153 ..and in asterisk CLI I have: -- Starting Skinny session from 192.168.0.155 Device SEPmacaddress is attempting to register Now when 7906G started is loaded: load file: sccp11.8-3-2s boot load id: tnp06.3-0-1-31.bin ..why isn't loaded sip firmware ?? Thanks in advance. -- Salvatore. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7906g SIP
Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: loadInformationSIP11.8-0-4SR1S/loadInformation ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving CTLSEPmacaddress.tlv to 192.168.0.155:49152 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving SEPmacaddress.cnf.xml to 192.168.0.155:49153 ..and in asterisk CLI I have: -- Starting Skinny session from 192.168.0.155 Device SEPmacaddress is attempting to register Now when 7906G started is loaded: load file: sccp11.8-3-2s boot load id: tnp06.3-0-1-31.bin ..why isn't loaded sip firmware ?? Thanks in advance. -- Salvatore. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automatic call pickup
--- On Tue, 10/7/08, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: regarding your combination of analog phones and ATAs I would look for the auto-dial functionality in the ATA. I am pretty sure I saw it in one web-interface or the other Thanks! I actually found the option. I'm using Grandstream's GXW4008. The option is Offhook Auto-Dial and I set that to *8. It seems to work fine. There's just one drawback: if I don't need to pick up a call but just place one then I need to press the R(Flash) key to get dial tone. Otherwise, *8 leaves me with a hung up tone and I can't dial out. This behavior may be even worse... so I may have to look for another solution. Thanks anyway. Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad Destinations
Very new to Asterisk, on my console it says there are 47 bad destinations...What is the best way to track these down and resolve them ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
Darren Severino wrote: Well, after very quickly making a test call it's not Vitelity. It could be something with your account? Might want to try opening a support ticket. If you want, create a sub account and e-mail me off list the username and password and I'll test it with my box or vice versa. You might also want to just check your settings at Vitelity. Over the last six months they have changed the server I'm support to connect to two or three times so my * box was not connecting to them. Therefor no service. I've I'd had it up for more than testing, and been testing, I'd have notices if there was any rime or reason for the changes. No notifications even. Rod -- On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I believe. Just to make sure, you are doing a 'module reload' each time you make changes to configuration files right? Cool I've got voicemail :-). I am reloading it and have increased the logging level. When dialing out I'm seeing: -- Executing Dial(SIP/101-08183018, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-0818b178 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Oct 7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Think it's a problem with vitelity? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
Interesting, I've been using them since April and haven't had a problem. I know they changed their server settings a while back but didn't notice anything recently. On Tue, Oct 7, 2008 at 11:47 AM, Roderick A. Anderson [EMAIL PROTECTED]wrote: Darren Severino wrote: Well, after very quickly making a test call it's not Vitelity. It could be something with your account? Might want to try opening a support ticket. If you want, create a sub account and e-mail me off list the username and password and I'll test it with my box or vice versa. You might also want to just check your settings at Vitelity. Over the last six months they have changed the server I'm support to connect to two or three times so my * box was not connecting to them. Therefor no service. I've I'd had it up for more than testing, and been testing, I'd have notices if there was any rime or reason for the changes. No notifications even. Rod -- On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I believe. Just to make sure, you are doing a 'module reload' each time you make changes to configuration files right? Cool I've got voicemail :-). I am reloading it and have increased the logging level. When dialing out I'm seeing: -- Executing Dial(SIP/101-08183018, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-0818b178 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Oct 7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Think it's a problem with vitelity? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
Hi! I just uploaded a small tarball of my ast_picker application with a few extras and an example_dialplan. You can find it here; http://juliencoder.de/ast_picker-0.1.tar.bz2 As I said before: It's still early stage and not too customiseable, but you can manage. If you need help, just tell me and I'll help as best I can. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conneting Asterisk to Swyx pri
I don't mean to be a pain, but i could really do with a heads up on this... does anyone have ANY ideas? I've trawled through google and come up with nothing except for questions with no answers... Cheers Geraint 2008/10/6 Geraint Lee [EMAIL PROTECTED] Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT - Swyx The above setup works fine... what i'm trying to achieve is BT SIP Trunks - Asterisk - Swyx I have connected to our BT (2 x ISDN30 UK) with asterisk and have no errors and can make and receive calls and it never dies... the problem comes when i try and connect asterisk to swyx... I can make calls from asterisk to the swyx system with no problems or errors, but... when i try and place a call from Swyx to asterisk i receive the following error: [Oct 6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !! Unexpected Channel selection 3 The call does complete as normal but after about 2 or 3 hours of calls passing through this setup i start receiving errors like the following: [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on bad channel 0/30 on span 3 [Oct 6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use And eventually no more calls can be placed from swyx to asterisk... time for some configs... and before anyone says something about wanpipe3 and 4 having dchan=0, i tried with dchan=16 and no calls can be placed... I hope someone can point me in the right direction as we're trying to get rid of swyx since we're tied down by limiting software and excessive licensing costs. Thanks! Geraint pri show spans shows all spans as up and active. zap show status shows all as ok wanrouter status shows all as connected wanpipe1 and 2: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF= NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES wanpipe3 and 4: [devices] wanpipe3 = WAN_AFT_TE1, Comment [interfaces] w3g1 = wanpipe3, , TDM_VOICE, Comment [wanpipe3] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 3 TE_CLOCK= MASTER TE_REF_CLOCK= 1 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 3 TDMV_DCHAN = 0 TDMV_HW_DTMF= NO [w3g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES zaptel.conf: loadzone=uk defaultzone=uk #Sangoma A104 port 1 [slot:1 bus:16 span:1] wanpipe1 span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 hardhdlc=16 #Sangoma A104 port 2 [slot:1 bus:16 span:2] wanpipe2 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 hardhdlc=47 #Sangoma A104 port 3 [slot:1 bus:16 span:3] wanpipe3 span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 #Sangoma A104 port 4 [slot:1 bus:16 span:4] wanpipe4 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 I have also tried with hardhdlc=109 and have the same problem. zapata.conf: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes musiconhold=default rxgain=0.0 txgain=0.0 immediate=no ; BT switchtype=euroisdn group=1 context=from-bt signalling=pri_cpe ; Port 1 - BT channel = 1-15,17-31 ; Port 2 - BT channel = 32-46,48-62 ; Swyx overlapdial=yes group=2 context=from-swyx signalling=pri_net ; Port 3 - Swyx channel = 63-77,79-93 ; Port 4 - Swyx channel = 94-108,110-124 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Setting up Asterisk
I posted this previously but didn't get a response. I have been working through the tutorial in the Van Meggelen book and can't get a registered SIP phone. I'm using 1.6, on Fedora 9 and a SPA941. I also tried an Engenius Wi-Fi SIP phone. Both can be pinged from the Linux Computer but neither give indication of being registered to Asterisk. This was supposed to be a no-brainer, and I don't know what to do to narrow the problem down. The Linux firewall is off. SELinux is in permissive mode. What sort of Asterisk commands should I be doing to test this? Or what other things? Ethereal snooping? Thanks, Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Asterisk
_ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilton Helm Sent: Tuesday, October 07, 2008 12:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Setting up Asterisk I posted this previously but didn't get a response. I have been working through the tutorial in the Van Meggelen book and can't get a registered SIP phone. I'm using 1.6, on Fedora 9 and a SPA941. I also tried an Engenius Wi-Fi SIP phone. Both can be pinged from the Linux Computer but neither give indication of being registered to Asterisk. This was supposed to be a no-brainer, and I don't know what to do to narrow the problem down. The Linux firewall is off. SELinux is in permissive mode. What sort of Asterisk commands should I be doing to test this? Or what other things? Ethereal snooping? Thanks, Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Asterisk
run asterisk in verbosed mode #asterisk -cg and try to register your WiFi phone or Xlite softephone ... and watch log on linux terminal One more option this command will extract packect contain 5060 port... #tcpdum -i eth0 port 5060 #if any issue in WiFi phone then try to register xlite or other softphone which you have.. Thanks satish patel _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilton Helm Sent: Tuesday, October 07, 2008 12:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Setting up Asterisk I posted this previously but didn't get a response. I have been working through the tutorial in the Van Meggelen book and can't get a registered SIP phone. I'm using 1.6, on Fedora 9 and a SPA941. I also tried an Engenius Wi-Fi SIP phone. Both can be pinged from the Linux Computer but neither give indication of being registered to Asterisk. This was supposed to be a no-brainer, and I don't know what to do to narrow the problem down. The Linux firewall is off. SELinux is in permissive mode. What sort of Asterisk commands should I be doing to test this? Or what other things? Ethereal snooping? Thanks, Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
On Monday 06 October 2008 14:58:09 Karl Fife wrote: In several places online, and in the Asterisk F.O.T. book, there is a warning against using '_.' saying: [it] should probably never be used. However, the need often arises act on numeric extensions that begin with *'s and #'s, and '+', and of course _X. does not match I have tried exten = _[0-9*#+]. but that seems to be the functional equivalent to _X. ignoring the addition of +,* and #. Can someone suggest the best way to deal with this without resoring to a highly repetitive/iterative dialplan? Leif and I discussed something like this at Astricon 2008, and we came up with this patch: http://bugs.digium.com/view.php?id=13632 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cisco phones getting SIP 401 unauthorized
I have a handful of cisco phones that has been working. Today they started showing X's. looking at sip debug I see the 401 unauthorized. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP IP:52110;branch=z9hG4bK29694d4a;received=IP From: sip:[EMAIL PROTECTED];user=phone To: sip:[EMAIL PROTECTED];user=phone;tag=as3155786a Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=03362846 Any idea what happened? Or how to get pasted the 401? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Jerry Geis wrote: I have a handful of cisco phones that has been working. Today they started showing X's. looking at sip debug I see the 401 unauthorized. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP IP:52110;branch=z9hG4bK29694d4a;received=IP From: sip:[EMAIL PROTECTED];user=phone To: sip:[EMAIL PROTECTED];user=phone;tag=as3155786a Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=03362846 Any idea what happened? Or how to get pasted the 401? Jerry This setup is actually running 1.2.14 asterisk. from way back - but till today had been working fine. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
Doug, I have your example working but how do I get this to work with a ring group? One more problem I have is poor quality of sound when the call file is played. I do not have this problem when moh is played or when console/dsp is used for live voice? What could be the problem? Do you know where I can find a ringing file? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, October 06, 2008 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging Robert Augustyn wrote: Ok then how do you make that an night_bell as your extension? We have an after hours IVR, press 1 if you know the party that you're trying to reach, press 2 for Dial By Directory and press 3 for the night bell. [incoming] ; ;* Check if call is within office hours, ;* if so, jump to the office-hours context ;* If not, continue on in the incoming ;* context. ; exten = s,1,GotoIfTime(07:59-16:59|mon-fri|*|*?office-hours,s,1) exten = s,n,Answer() exten = s,n,Wait(1) ;** ;* If after hours then play the 'Welcome' ;* and office hours message Press 1 if you know ;* the extension or 2 for dial by name directory ;** exten = s,n,Background(local/welcome) exten = s,n,Background(local/business-hours) exten = s,n,Background(local/8am-5pm) exten = s,n,Background(local/press1-extension) exten = s,n,Background(local/press2-directory) exten = s,n,Background(local/press3-night-bell) ;* ;* Set timeouts ;* exten = s,11,Set(TIMEOUT(response)=15) exten = s,12,Set(TIMEOUT(digit)=2) ;* ;* If 1 is pressed, go to Dial by extension ;* exten = 1,1,Goto(dial-by-extension,s,1) ; ;* If 2 is pressed, go to Dial by name ; exten = 2,1,Goto(directory,s,1) ; ;* If 3 is pressed, go to Night Bell ; exten = 3,1,Goto(night_bell,4173,1) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Did the server reboot or lose communication? This happens with our 7970's sometimes if there's been a hiccup, usually dialing voicemail registers them back up - occasionally we've had to do the soft reboot from the screen. 401 unauth - looks like it may be md5secret issue, or nat traversal over a firewall - are the phones inside the lan or on the net? Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Tuesday, October 07, 2008 3:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized Jerry Geis wrote: I have a handful of cisco phones that has been working. Today they started showing X's. looking at sip debug I see the 401 unauthorized. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP IP:52110;branch=z9hG4bK29694d4a;received=IP From: sip:[EMAIL PROTECTED];user=phone To: sip:[EMAIL PROTECTED];user=phone;tag=as3155786a Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=03362846 Any idea what happened? Or how to get pasted the 401? Jerry This setup is actually running 1.2.14 asterisk. from way back - but till today had been working fine. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
Hello Robert! I don'texactly know, what you need for a ringing file. but if it is the matter of just some announcement sound, I could make you one. It's easy. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asteriskt38.com
Since when is there a T.38 Gateway in Asterisk 1.4? On Tue, Oct 7, 2008 at 3:01 AM, Daniel Ferenci [EMAIL PROTECTED] wrote: Hi, fax gateway isn't just a packet bridging. It does the mediation between T30 (voice) - T38 (fax over ip) protocols. It does work for asterisk 1.4, asterisk 1.6, asterisk svn head. If it doesn't please send me a bug report and I'm going to fix it. Best regards Daniel. On Mon, Oct 6, 2008 at 7:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: That isn't real T.38 support, it's just Packet2Packet bridging that works correctly. Still need to use a Cisco gateway to support sending the faxes somewhere on the PSTN. But it does work and it is reliable, I use it every day. On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote: Actually it exists. 1.4 had passtrough mode ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
Julien, Thank you, I need a file which when played sounds like a phone ringing ... :) robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julien Claassen Sent: Tuesday, October 07, 2008 3:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging Hello Robert! I don'texactly know, what you need for a ringing file. but if it is the matter of just some announcement sound, I could make you one. It's easy. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asteriskt38.com
Hi, http://bugs.digium.com/view.php?id=13405 was posted on 30/08/2008. I'm looking forward to seeing your feedback or bug report. Thank you in advance. Best regards Daniel. On Tue, Oct 7, 2008 at 10:01 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote: Since when is there a T.38 Gateway in Asterisk 1.4? On Tue, Oct 7, 2008 at 3:01 AM, Daniel Ferenci [EMAIL PROTECTED] wrote: Hi, fax gateway isn't just a packet bridging. It does the mediation between T30 (voice) - T38 (fax over ip) protocols. It does work for asterisk 1.4, asterisk 1.6, asterisk svn head. If it doesn't please send me a bug report and I'm going to fix it. Best regards Daniel. On Mon, Oct 6, 2008 at 7:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: That isn't real T.38 support, it's just Packet2Packet bridging that works correctly. Still need to use a Cisco gateway to support sending the faxes somewhere on the PSTN. But it does work and it is reliable, I use it every day. On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote: Actually it exists. 1.4 had passtrough mode ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
No problem... I'll whomp something up. I'll upload a tarball tomorrow or thrusday morning at the latest. Quality: desired samplingrate, bit-depth, channel number? Any particular needs, or will CD quality just be fine for you? Kindest regards Julien P.S.: Did you get to my application? Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on screening calls / Question about the Dial g option
Hi, I have a simple desire to be able to screen people before being onnected to them. I`ve seen plenty of examples on the web and I`ve figured it out. There is only one case in where it doesnt act as I want it to: if I hang up the phone, I don`t want the caller to be disconnected but (for the sake of making this example simpler) I want him to hear the person you are trying to reach is unavailable. Problem is, in my screen macro (the one called using the M() option in the dial command), I can set MACRO_RESULT but it`s all lost when I hang up. If I use the g() option in dial, the MACRO_RESULT variable is lost. Actually, as far as I can tell (which is also what I understand about channel variables) as soon as the I hang up the dial, more commands will be executed but all variables will be lost. So, I guess my question is: how do I set a variable that ISN`T lost when the call initiated using the Dial g option is hung up ? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on screening calls / Question about the Dial g option
Mike wrote: So, I guess my question is: how do I set a variable that ISN`T lost when the call initiated using the Dial g option is hung up ? You can use the internal database for that: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging
Anything what can be played through the console/dsp will work for me. Yes, I received your application and hope to play with it tonight or tomorrow. Thank you very much. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julien Claassen Sent: Tuesday, October 07, 2008 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging No problem... I'll whomp something up. I'll upload a tarball tomorrow or thrusday morning at the latest. Quality: desired samplingrate, bit-depth, channel number? Any particular needs, or will CD quality just be fine for you? Kindest regards Julien P.S.: Did you get to my application? Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on screening calls / Question about the Dial g option
Doug, Thanks for the quick answer. How does that help me though, since this is a per channel variable and not a global variable? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, October 07, 2008 16:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on screening calls / Question about the Dial g option Mike wrote: So, I guess my question is: how do I set a variable that ISN`T lost when the call initiated using the Dial g option is hung up ? You can use the internal database for that: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Did the server reboot or lose communication? This happens with our 7970's sometimes if there's been a hiccup, usually dialing voicemail registers them back up - occasionally we've had to do the soft reboot from the screen. 401 unauth - looks like it may be md5secret issue, or nat traversal over a firewall - are the phones inside the lan or on the net? Thanks, Matt G Matt, The phones are inside the LAN. what is the md5secret? I dont know what that is? can I disable it? Or how do I set it up? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Tuesday, October 07, 2008 5:42 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized Matt, The phones are inside the LAN. what is the md5secret? I dont know what that is? can I disable it? Or how do I set it up? Jerry Hi Jerry, Hm, okay. We had to use md5secret (instead of secret) in the sip.conf for our 7970's to get them to successfully register with asterisk. However, if you had them working before then I doubt this is the issue. You can try anyway though, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret We use both secret= and md5secret= with the same password in each, one encrypted and one not encrypted - this seemed to let our 7970 register. HTH, Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
THANK YOU!!! This appears to have worked. I am assuming we can do the same thing on our SPA-962s that we send to make sure they work with no problems. Thank you to everyone here for your help. This is an excellent group to have access to for questions. I hope to learn and be able to help others. Steve Anness On 10/7/08 9:53 AM, Andres [EMAIL PROTECTED] wrote: Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. The fact that they work initially is probably a clear indication that the NAT bindings are closing up after a few minutes. In some cases it does not matter that you have qualify=yes, since the router only keeps bindings open if the traffic is being generated from the inside-outside. Your solution would be to enable the keep-alive settings on the PAP2 and set it low to something like 15 seconds. The setting is under the tab of line 1 and line 2 and its called NAT Keep Alive Enable. Andres http://www.neuroredes.com do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Mysql for CentOS 5.2 is the mysql client tools. mysql.i386 : MySQL client programs and shared libraries. Does anyone have any other suggestions? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Tuesday, 7 October 2008 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0 Yes, unfortunately, VOIP wiki did not mention about installing mysql-client which it should have been. Without mysql-client, you cannot change passwords, grants, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stefan Schmidt Sent: Tuesday, 7 October 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons- 1.6.0 Klaverstyn, David C schrieb: Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. Hello, maybe you should install mysql-client too ;) best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on screening calls / Question about the Dial g option
Mike wrote: Doug, Thanks for the quick answer. How does that help me though, since this is a per channel variable and not a global variable? Make sure your key in the database is specific to only that call. Time, date, caller-id number or even a combination of all. Can't you save your channel variable within the db and purge it when you're done with it? Save your MACRO_RESULTS as the value and the key being your channel? Since I don't know what your macro is actually doing, I'm just throwing things out there. If you need a persistant variable, save it to your database with enough info to identify what you're looking for. Once done, purge the database of that entry. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help no ring on caller side
Hi, Got this weird problem that the caller does not hear a ring. The issue is it's specific to the local telco: Using telco 1 (mobile), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. Using telco 1 (landline), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. But using Telco 2, my phone is ringing, caller does not hear a ring on his side, and i dont answer call hangs up instead of going to voicemail where should i start tracing the problem? TIA regards nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] requested special control 20 ??
I'm using Teliax, and every incoming call has: Executing [EMAIL PROTECTED]:2] Answer(IAX2/usrname-14376, ) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(IAX2/usrname-14376, DAHDI/1,60) in new stack -- Called 1 -- DAHDI/1-1 is ringing -- IAX2/usrname-14376 requested special control 20, passing it to DAHDI/1-1 -- IAX2/usrname-14376 requested special control 20, passing it to DAHDI/1-1 -- DAHDI/1-1 is ringing It all seems to work OK, but what's requesting special control 20 all about? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] changing passwords
I have a question about changing passwords. When I change the secret field in sip.conf for a Grandstream phone, and then use the browser to change the Authenticate ID field of the phone to match what's in the sip.conf file, I can no longer make calls on the phone. Any ideas? Thanks for any help, Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Yes, try perl -MCPAN -e install DBD::mysql Then do a make clean, ./bootstrap, ./configure, make menuselect Worked for me, not sure all the above is required but configure. Thanks, Steve Totaro On Tue, Oct 7, 2008 at 7:19 PM, Klaverstyn, David C [EMAIL PROTECTED] wrote: Mysql for CentOS 5.2 is the mysql client tools. mysql.i386 : MySQL client programs and shared libraries. Does anyone have any other suggestions? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Tuesday, 7 October 2008 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0 Yes, unfortunately, VOIP wiki did not mention about installing mysql-client which it should have been. Without mysql-client, you cannot change passwords, grants, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stefan Schmidt Sent: Tuesday, 7 October 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons- 1.6.0 Klaverstyn, David C schrieb: Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. Hello, maybe you should install mysql-client too ;) best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Klaverstyn, David C schrieb: Mysql for CentOS 5.2 is the mysql client tools. mysql.i386 : MySQL client programs and shared libraries. Does anyone have any other suggestions? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stefan Schmidt Klaverstyn, David C schrieb: I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. maybe you should install mysql-client too ;) http://www.centos.org/modules/newbb/viewtopic.php?topic_id=13097 Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Philipp Kempgen schrieb: Klaverstyn, David C schrieb: Mysql for CentOS 5.2 is the mysql client tools. mysql.i386 : MySQL client programs and shared libraries. Does anyone have any other suggestions? http://www.centos.org/modules/newbb/viewtopic.php?topic_id=13097 Or just download Debian at http://www.debian.org/ :-) SCNR Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing passwords
Ken Zarifes schrieb: When I change the secret field in sip.conf for a Grandstream phone, and then use the browser to change the Authenticate ID field of the phone to match what's in the sip.conf file, I can no longer make calls on the phone. Any ideas? Go to the Asterisk CLI, core set verbose 3, watch the output for messages about failed authentication attempts or something. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing passwords
On Tue, Oct 7, 2008 at 8:30 PM, Ken Zarifes [EMAIL PROTECTED] wrote: I have a question about changing passwords. When I change the secret field in sip.conf for a Grandstream phone, and then use the browser to change the Authenticate ID field of the phone to match what's in the sip.conf file, I can no longer make calls on the phone. Any ideas? Thanks for any help, Ken It has been a long time since I touched a GS phone but I think the field on the browser is password, not Authenticate ID. Don't forget to reload sip.conf or asterisk between changes. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help no ring on caller side
On Tue, Oct 7, 2008 at 8:04 PM, Nhadie [EMAIL PROTECTED] wrote: Hi, Got this weird problem that the caller does not hear a ring. The issue is it's specific to the local telco: Using telco 1 (mobile), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. Using telco 1 (landline), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. But using Telco 2, my phone is ringing, caller does not hear a ring on his side, and i dont answer call hangs up instead of going to voicemail where should i start tracing the problem? TIA regards nhadie Try answering first. [telco2] exten = s,1,Answer() -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on screening calls / Question about the Dial g option
On Tue, Oct 7, 2008 at 4:36 PM, Mike [EMAIL PROTECTED] wrote: Hi, I have a simple desire to be able to screen people before being onnected to them. I`ve seen plenty of examples on the web and I`ve figured it out. There is only one case in where it doesn't act as I want it to: if I hang up the phone, I don`t want the caller to be disconnected but (for the sake of making this example simpler) I want him to hear the person you are trying to reach is unavailable. Problem is, in my screen macro (the one called using the M() option in the dial command), I can set MACRO_RESULT but it`s all lost when I hang up. If I use the g() option in dial, the MACRO_RESULT variable is lost. Actually, as far as I can tell (which is also what I understand about channel variables) as soon as the I hang up the dial, more commands will be executed but all variables will be lost. So, I guess my question is: how do I set a variable that ISN`T lost when the call initiated using the Dial g option is hung up ? Regards,** * * *Mike* What variables do you need to play the person you are trying to reach is unavailable? Just use the 'h' extension to play the file. BTW, channel variables ARE available to the h extension to a degree, at least to the first priority of the h exten, they seem to get lost after the first priority even if you use setvar or set with single or double underscores. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Efax from Agi script
I recently did something similar using fax1.com. If you can send an email you can send a fax that way. On Tue, Oct 7, 2008 at 9:19 AM, Riccardo Cupardo [EMAIL PROTECTED] wrote: Hi all, i wrote a script agi, sking for a code, after that it sends an email now i need to send a fax... any hints or tips for that? Ty in advance. -- Riccardo Cupardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing passwords
I got this: [Oct 7 18:26:17] NOTICE[6309] chan_sip.c: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.163.134' - Username/auth name mismatch -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Tuesday, October 07, 2008 6:04 PM To: Asterisk Users Subject: Re: [asterisk-users] changing passwords Ken Zarifes schrieb: When I change the secret field in sip.conf for a Grandstream phone, and then use the browser to change the Authenticate ID field of the phone to match what's in the sip.conf file, I can no longer make calls on the phone. Any ideas? Go to the Asterisk CLI, core set verbose 3, watch the output for messages about failed authentication attempts or something. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing passwords
It has been a long time since I touched a GS phone but I think the field on the browser is password, not Authenticate ID. Don't forget to reload sip.conf or asterisk between changes. You were absolutely right. Thanks! Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing passwords
The value is not Authenticate ID; From the config file: # Authenticate ID P36 = 8000 # Authenticate password P34 = If you look at the HTML source of the webconfig the form field you need to edit will be marked P34. On Tue, Oct 7, 2008 at 5:30 PM, Ken Zarifes [EMAIL PROTECTED] wrote: I have a question about changing passwords. When I change the secret field in sip.conf for a Grandstream phone, and then use the browser to change the Authenticate ID field of the phone to match what's in the sip.conf file, I can no longer make calls on the phone. Any ideas? Thanks for any help, Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
Load the firmware of www.dd-wrt.com on that WRT54G and then put all the VoIP devices directly behind it. It MIGHT work to set the first NAT router to have the 2nd NAT router in the 1st's DMZ... but I prefer to do things The Right Way. On Tue, Oct 7, 2008 at 7:24 AM, Steve Anness [EMAIL PROTECTED] wrote: I have just confirmed that they may be having a problem with double NAT. They have two ATAs, and they have two different DSL connections. One set-up goes from the first DSL Modem (NAT Wirless are disabled on the DSL Modems) to a Linksys WRT110 and then there is a WRT54G hooked in to the 110 that has the ATA plugged into it. The other ATA is configured from a DSL Modem (again, I was told NAT Wireless were disabled on the modem) to a WRT600N and the ATA is plugged in there. I have the same issues on both ATAs. I have no idea why their network is as poorly designed as it is, the bad part is I have to make sure the phones work there and try to troubleshoot from 3000 miles away. Any work arounds for a problem because of double NAT? A quick and dirty solution for them to get their phones working right? Steve Anness On 10/7/08 2:12 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
On Tue, Oct 7, 2008 at 6:00 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Philipp Kempgen schrieb: Klaverstyn, David C schrieb: Mysql for CentOS 5.2 is the mysql client tools. mysql.i386 : MySQL client programs and shared libraries. Does anyone have any other suggestions? http://www.centos.org/modules/newbb/viewtopic.php?topic_id=13097 Or just download Debian at http://www.debian.org/ :-) SCNR Or SuSE at http://software.opensuse.org/ ... IMO the best package management of any distro. ...You would think PNAELV or Cent would have developed a better tool by now... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help no ring on caller side
Try making sure you use the r option in your dialstring. You should *NOT* be answering a ringing channel, as Steve suggested, FWIW (if it doesn't work any other way that is another story) On Tue, Oct 7, 2008 at 5:04 PM, Nhadie [EMAIL PROTECTED] wrote: Hi, Got this weird problem that the caller does not hear a ring. The issue is it's specific to the local telco: Using telco 1 (mobile), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. Using telco 1 (landline), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. But using Telco 2, my phone is ringing, caller does not hear a ring on his side, and i dont answer call hangs up instead of going to voicemail where should i start tracing the problem? TIA regards nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
Leif and I discussed something like this at Astricon 2008, and we came up with this patch: http://bugs.digium.com/view.php?id=13632 -- Tilghman That's a great idea. Good work. Also, nice work with the new CDR stuff in 1.6! So that leaves only one question: exten = ? What extension the following: '3129842314' '*989' '+13129842314' BUT does not match: 'i' 'james' is this possible? Thanks for your input! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automatic call pickup
I am not sure if it is possible to somehow invoke a function to pick up the call via dialplan, if it is a combination of that function and DISA should do what you need. On Tue, Oct 7, 2008 at 8:37 AM, Vieri [EMAIL PROTECTED] wrote: --- On Tue, 10/7/08, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: regarding your combination of analog phones and ATAs I would look for the auto-dial functionality in the ATA. I am pretty sure I saw it in one web-interface or the other Thanks! I actually found the option. I'm using Grandstream's GXW4008. The option is Offhook Auto-Dial and I set that to *8. It seems to work fine. There's just one drawback: if I don't need to pick up a call but just place one then I need to press the R(Flash) key to get dial tone. Otherwise, *8 leaves me with a hung up tone and I can't dial out. This behavior may be even worse... so I may have to look for another solution. Thanks anyway. Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help no ring on caller side
Andrew Joakimsen wrote: Try making sure you use the r option in your dialstring. You should *NOT* be answering a ringing channel, as Steve suggested, FWIW (if it doesn't work any other way that is another story) Thanks, tried the 'r' and it works. And even the voicemail worked after that. I guess telco needed the ring to hung up properly as well. Thanks again. Nhadie On Tue, Oct 7, 2008 at 5:04 PM, Nhadie [EMAIL PROTECTED] wrote: Hi, Got this weird problem that the caller does not hear a ring. The issue is it's specific to the local telco: Using telco 1 (mobile), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. Using telco 1 (landline), calls in to my DID, caller hears a ring and gets forwarded to voicemail if i did not answer. But using Telco 2, my phone is ringing, caller does not hear a ring on his side, and i dont answer call hangs up instead of going to voicemail where should i start tracing the problem? TIA regards nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] registration limit
Hi, Is there a way to limit only one registration for each user at a time? meaning if a user tries to register, but that user is already registered. i will deny? or is it possible to for a single user at the same time, and when someone calls that user, it will ring both phones? Just want something whereby a user can assign his extension on an IP phone in the office, and assign the same thing maybe to a softphone on his laptop or maybe a sip client on a mobile phone. so that whenever he leaves the office he can still be reach on his extension via the sotphone. thank you. regards, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Destinations
What do you do to get that message? On Tue, Oct 7, 2008 at 8:45 AM, Mr surfit [EMAIL PROTECTED] wrote: Very new to Asterisk, on my console it says there are 47 bad destinations...What is the best way to track these down and resolve them ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
