Re: [asterisk-users] Asterisk 12 questions
Looks like I figured It out! A couple of things seemed to be getting in the way: 1. Old leftovers - I had a previous version of Asterisk kicking about. I used the package manager and removed It. 2. Openssl-dev libraries. Learned that it's a dependency of chan_sip. I am embarrassed to say that I did not know! Now, I've a working version of Asterisk 12 with chan_sip . Now - to decipher the AMI... Glen Sent from my Android - if my spelling, diction or grammar is poor, please don't think that I am an idiot! On Jan 30, 2014 7:53 AM, "Daniel Jenkins" wrote: > > > > On Thu, Jan 30, 2014 at 12:48 PM, Glen Millard wrote: > >> >> Hi. I'm attempting to compile Asterisk 12, but we want to use chan_sip >> instead of pjsip. >> > > Hi Glenn, > > >> I am missing something. I assumed that chan_sip was going to be added by >> default. Apparently not. I saw it in the menuconfig. Dumb question, but >> double xx beside It..does that mean not avail/not going to be installed? >> > > Yes, if theres x's it means it can't be installed - due to lack of a > dependency, rather than it being an option which has been enabled or > disabled. > > >> Can someone point me in the proper direction? A specific area where I can >> learn to build Asterisk 12 with chan_sip? I'm needing to use the chan_sip >> for the time being until I can learn the new SIP stack. >> > > Have you tried running ./contrib/scripts/install_prereq ? this should > install everything you need if you're compiling from source. > > > http://svn.asterisk.org/svn/asterisk/branches/12/contrib/scripts/install_prereq > > I don't know what you need to be able to install chan_sip as I've never > had to have a look, but your first port of call should be the > install_prereq script. > >> Thanks - much appreciated! >> >> Glen >> > > Dan > > >> Sent from my Android - if my spelling, diction or grammar is poor, please >> don't think that I am an idiot! >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 12 questions
Hi. I'm attempting to compile Asterisk 12, but we want to use chan_sip instead of pjsip. I am missing something. I assumed that chan_sip was going to be added by default. Apparently not. I saw it in the menuconfig. Dumb question, but double xx beside It..does that mean not avail/not going to be installed? Can someone point me in the proper direction? A specific area where I can learn to build Asterisk 12 with chan_sip? I'm needing to use the chan_sip for the time being until I can learn the new SIP stack. Thanks - much appreciated! Glen Sent from my Android - if my spelling, diction or grammar is poor, please don't think that I am an idiot! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass-through Call Recording Transfer Information
> > I am currently using asterisk to record all incoming calls. My setup is as > follows, the asterisk server has a two TE120P cards one of which > sends/receives calls from the carrier and the other is connected to a > Siemens HiPath 3000. All calls that come into asterisk use MixMonitor to > record calls and this works fine, but if a call gets transferred the > transfer information is not sent back to my asterisk box (as far as I know), > therefore it is hard to track calls in a searchable database. I am using the > ETSI protocol between the asterisk box and the HiPath 3000 so that if the > asterisk box breaks down we can simply plug the incoming lines from the > asterisk box to the HiPath 3000 and keep receiving calls. Does anybody have > any idea's that will enable me to get this transfer information. > > Thanks > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pass-through Call Recording Transfer Information
Hi, I am currently using asterisk to record all incoming calls. My setup is as follows, the asterisk server has a two TE120P cards one of which sends/receives calls from the carrier and the other is connected to a Siemens HiPath 3000. All calls that come into asterisk use MixMonitor to record calls and this works fine, but if a call gets transferred the transfer information is not sent back to my asterisk box (as far as I know), therefore it is hard to track calls in a searchable database. I am using the ETSI protocol between the asterisk box and the HiPath 3000 so that if the asterisk box breaks down we can simply plug the incoming lines from the asterisk box to the HiPath 3000 and keep receiving calls. Does anybody have any idea's that will enable me to get this transfer information. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNOW sip module not installed
Hey Guys, I have installed AsteriskNOW and I've found that the SIP module is not installed (along with all the SIP configuration files) until I log into the freepbx webGUI and click the reload button. Is there a way to get all the sip configuration files and module installed without having to do a reload from the webGUI? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] web-meetme cbEnd.php not running - error
Hey, Ive installed web meetme and everything is working fine except no records are being written to the cdr and participants tables, this is because the cbEnd.php script is not running. Below is the output of the cbEnd.php when I run in manually. I am running asterisk 1.4.20.1 and web meetme 3.1.0 and the latest release's of PEAR,PHP and MySQL. ./cbEnd.php PHP Strict Standards: Assigning the return value of new by reference is deprecated in /usr/share/php5/PEAR/PEAR.php on line 569 PHP Strict Standards: Assigning the return value of new by reference is deprecated in /usr/share/php5/PEAR/PEAR.php on line 572 PHP Strict Standards: Non-static method DB::connect() should not be called statically in /usr/local/apache2/htdocs/web-meetme/lib/database.php on line 8 PHP Strict Standards: Non-static method DB::parseDSN() should not be called statically in /usr/share/php5/PEAR/DB.php on line 520 PHP Strict Standards: Non-static method PEAR::raiseError() should not be called statically in /usr/share/php5/PEAR/DB.php on line 543 PHP Strict Standards: Non-static method DB::errorMessage() should not be called statically, assuming $this from incompatible context in /usr/share/php5/PEAR/DB.php on line 965 PHP Strict Standards: Non-static method DB::isError() should not be called statically, assuming $this from incompatible context in /usr/share/php5/PEAR/DB.php on line 688 PHP Strict Standards: is_a(): Deprecated. Please use the instanceof operator in /usr/share/php5/PEAR/DB.php on line 594 PHP Strict Standards: Non-static method PEAR::getStaticProperty() should not be called statically, assuming $this from incompatible context in /usr/share/php5/PEAR/PEAR.php on line 867 PHP Strict Standards: Non-static method DB::isError() should not be called statically in /usr/local/apache2/htdocs/web-meetme/lib/database.php on line 9 PHP Strict Standards: is_a(): Deprecated. Please use the instanceof operator in /usr/share/php5/PEAR/DB.php on line 594 DB Error: not found ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] web meetme PHP undefined variable
I am hoping maybe some of you have come across these before in your experience with web meetme. Below are the messages im receiveing when I load the web meetme home page. Notice: Undefined variable: s in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 9 Notice: Undefined variable: logoff_section in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 12 Notice: Undefined variable: logoff_section in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 19 Notice: Undefined index: auth in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 29 Notice: Undefined variable: AUTH_USER in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 39 Notice: Undefined index: auth in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 45 Notice: Undefined index: auth in /usr/local/apache2/htdocs/web-meetme/lib/header.inc on line 28 Notice: Undefined variable: logoff_sel in /usr/local/apache2/htdocs/web-meetme/lib/header.inc on line 35 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Web Meetme module not loading
Matt Riddell wrote: > In the latest readme for WebMeetMe (3.1.0) it states: > > * Compile and install CBMySQL > App_cbmysql is now included in the web-meetme package, > located in ./cbmysql. To install just run make; make install > > Copy the sample cbmysql.conf to /etc/asterisk and create > a dialplan similar to the one in cb-extensions.conf.sample > Modify the settings to suit your system. The location of the > mysql.sock file is likely not correct, check /etc/my.conf for > the correct location. > > That fixed it Matt, just compiling in the wrong directory. Thanks for all your help. -Glen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Web Meetme module not loading
On Tue, Sep 1, 2009 at 3:06 PM, Matt Riddell wrote: > On 1/09/09 4:54 PM, Glen wrote: > >>>>> Parsing '/etc/asterisk/cbmysql.conf': Found > >>>>> asterisk: symbol lookup error: > /usr/lib/asterisk/modules/app_cbmysql.so: > >>>>> undefined symbol: mysql_init > >> ldd /usr/lib/asterisk/modules/app_cbmysql.so > > This is the output > > > > linux-gate.so.1 => (0xe000) > > libpthread.so.0 => /lib/libpthread.so.0 (0xb7f6) > > libc.so.6 => /lib/libc.so.6 (0xb7e2d000) > > /lib/ld-linux.so.2 (0x8000) > > Er weird - so it's not even requesting a link to the mysql library. > > Looks like the linking somehow went wrong - if it can't find mysql_init > and it doesn't look for it then surely something went wrong at the > linking stage. > > Can you post me the output of the compilation? > > -- > Cheers, > > Matt Riddell > Director > When compiling the module I simply recompiled asterisk (I was told this is the best way), below is the output of that. remote:/usr/src/asterisk-1.4.20.1 # make && make install Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CC] app_cbmysql.c -> app_cbmysql.o app_cbmysql.c:37:1: warning: "AST_MODULE" redefined : warning: this is the location of the previous definition app_cbmysql.c: In function âcheckMaxâ: app_cbmysql.c:116: warning: implicit declaration of function âast_say_numberâ app_cbmysql.c: In function âroomQueryâ: app_cbmysql.c:181: warning: unused variable âeatimeâ app_cbmysql.c:337: warning: control reaches end of non-void function [LD] app_cbmysql.o -> app_cbmysql.so make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. +- Asterisk Build Complete -+ + Asterisk has successfully been built, and + + can be installed by running: + + + + make install+ +---+ menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. if [ x`/usr/bin/id -un` = xroot ]; then CFLAGS=" -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -include /usr/src/asterisk-1.4.20.1/include/asterisk/autoconfig.h -march=i686 " sh build_tools/mkpkgconfig /usr/lib/pkgconfig; fi mkdir -p /var/lib/asterisk/static-http for x in static-http/*; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/static-http ; \ done mkdir -p /var/lib/asterisk/images for x in images/*.jpg; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/images ; \ done mkdir -p /var/lib/asterisk/agi-bin make -C sounds install make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/sounds' make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/sounds' mkdir -p /usr/lib/asterisk/modules mkdir -p /usr/sbin mkdir -p /etc/asterisk mkdir -p /usr/bin mkdir -p /var/run mkdir -p /var/spool/asterisk/voicemail mkdir -p /var/spool/asterisk/dictate mkdir -p /var/spool/asterisk/system mkdir -p /var/spool/asterisk/tmp mkdir -p /var/spool/asterisk/meetme mkdir -p /var/spool/asterisk/monitor make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/utils' for x in astman stereorize streamplayer aelparse muted; do \ if [ "$x" != "none" ]; then \ /usr/bin/install -c -m 755 $x /usr/sbin/$x; \ fi; \ done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/utils' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/agi' mkdir -p /var/lib/asterisk/agi-bin for x in agi-test.agi eagi-test eagi-sphinx-test jukebox.agi; do /usr/bin/install -c -m 755 $x /var/lib/asterisk/agi-bin ; done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/agi' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/res' for x in res_adsi.so res_a
Re: [asterisk-users] Asterisk Web Meetme module not loading
Matt Riddell wrote: > On 1/09/09 4:31 PM, Glen wrote: > >> Matt Riddell wrote: >> >>> On 31/08/09 2:33 PM, Glen wrote: >>> >>> >>>> I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also >>>> installed the latest versions of mysql and php. I followed the readme >>>> file that came with the web meetme app and everything seemed to go fine >>>> up until I realised the module wasnt being loaded. When I stop asterisk >>>> and try to start it, it errors out and does not load and I get the >>>> following message: >>>> >>>> Parsing '/etc/asterisk/cbmysql.conf': Found >>>> asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: >>>> undefined symbol: mysql_init >>>> >>>> >>> Likely you don't have mysql-devel libraries installed - though I wonder >>> how it would have compiled. >>> >>> mysql_init is a function provided by the libmysqlclient library - if you >>> didn't compile app_cbmysql.so yourself, you could type ldd >>> app_cbmysql.so to see what it links to then check your lib directory to >>> see if you have the same - you might have 64 bit when it was compiled >>> for 32 bit or something >>> \ >>> >> Hi Matt, >> >> I have the following mysql packages installed >> >> MySQL-client-community-5.1.37 >> MySQL-devel-community-5.1.37 >> MySQL-server-community-5.1.37 >> MySQL-shared-community-5.1.37 >> >> Also I get no errors when compiling app_cbmysql.so (I do compile this >> from source) >> > > What do you get if you type: > > ldd /usr/lib/asterisk/modules/app_cbmysql.so > > This is the output linux-gate.so.1 => (0xe000) libpthread.so.0 => /lib/libpthread.so.0 (0xb7f6) libc.so.6 => /lib/libc.so.6 (0xb7e2d000) /lib/ld-linux.so.2 (0x8000) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Web Meetme module not loading
Matt Riddell wrote: > On 31/08/09 2:33 PM, Glen wrote: > >> I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also >> installed the latest versions of mysql and php. I followed the readme >> file that came with the web meetme app and everything seemed to go fine >> up until I realised the module wasnt being loaded. When I stop asterisk >> and try to start it, it errors out and does not load and I get the >> following message: >> >> Parsing '/etc/asterisk/cbmysql.conf': Found >> asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: >> undefined symbol: mysql_init >> > > Likely you don't have mysql-devel libraries installed - though I wonder > how it would have compiled. > > mysql_init is a function provided by the libmysqlclient library - if you > didn't compile app_cbmysql.so yourself, you could type ldd > app_cbmysql.so to see what it links to then check your lib directory to > see if you have the same - you might have 64 bit when it was compiled > for 32 bit or something > \ Hi Matt, I have the following mysql packages installed MySQL-client-community-5.1.37 MySQL-devel-community-5.1.37 MySQL-server-community-5.1.37 MySQL-shared-community-5.1.37 Also I get no errors when compiling app_cbmysql.so (I do compile this from source) Any idea's? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Web Meetme module not loading
I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also installed the latest versions of mysql and php. I followed the readme file that came with the web meetme app and everything seemed to go fine up until I realised the module wasnt being loaded. When I stop asterisk and try to start it, it errors out and does not load and I get the following message: Parsing '/etc/asterisk/cbmysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_init Has anybody seen this error message or have any idea what the problem could possibly be? If more information is required please let me know and I will post up any config files you need. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Getting two Asterisk machines to talk to each other
I want to do something that seems like it should be very simple. Then again so am I ;) Anyway, I have two asterisk boxes on separate subnets. What I want is for a certain extension 3053 when dialed on the main machine to ring through to a SIP phone that is registered on the other Asterisk machine. I'm sure the information is out there I just can't seem to concoct the proper search string to find it. This is what I've tried: I have created a peer entry for each machine in the other's sip.conf I have changed the extension so that when it is dialed on the main machine it does the following Dial(SIP/test-machine/3053)I feel this is most likely my problem. The other machine has the same generic setup as the first (copied config files) with the exception of 3053 ringing the phone and the peer definition in sip.conf of course. When I tried calling the extension via the main machine it simply rang to voicemail. Any help would be appreciated. Thanks. -Glen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to a Huawei softX3000
Greetings, I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be something as simple as perhaps the register string which is currently @ although I've tried a number of variations without success. Here's a snipit from sip.conf allow=ulawauth=md5disallow=alldtmf=inbandhost=xxx.xxx.xxx.xxxinsecure=verysecret=xxxtype=peerusername=xxx Has anyone been able to register Asterisk against this Huawei switch? Normally I'd just muddle though it but I've spent the day working on this with NO success. I should also mention I've done ethereal dumps of devices that successfully register and I can't spot any differences. Thanks! Glen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Sourcing Equipment at the HK Electronics Fair
I've just spent the last 3 days at the Hong Kong Electronics Fair and will be returning for its last day on Sunday. I must say most of the telecom vendors are displaying at least one VoIP phone, with many having a full range of products such as Phones (Corded, Cordless and WiFi), ATA's, & Gateways. The really good news is that most of the phones are looking better and have much better feel to them in terms of quality, weight, etc. Next week I'll be traveling to Guangzhou to catch the last three days of the first phase of the China Sourcing Fair. The Canton Fair is one of, if not the largest, sourcing fair in China so I expect to see more VoIP equipment. If anyone has any requests for equipment you maybe looking to source (especially you dealers, distributors, and service providers) let me know your requirements and I'll keep my eyes open. Look for a full report next week. You can email me off list at gbrowley (at) gmail (dot) com. -- Kind regards, Glen Browley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] perl poe::component::client::asterisk::manager usage
Folks, Is anyone using the poe::component::client::asterisk::manager perl module in a production environment? Any examples/urls would be appreciated. -g ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] differentiating busy & not connected
For those who may be searching, you can accomplish this using the
${DIALSTATUS} variable.
The variable will report CONGESTED if the device (channel) is not
available, & will report BUSY if it is in use.
-g
On Sat, 2005-02-12 at 02:52 +0100, Philipp von Klitzing wrote:
> Hi!
>
> > How can I determine whether a sip device is busy (channel unavailable),
> > or it is not registered/connected?
>
> For the latter try "database show sip" or DBGet(...) respectively. Won't
> work with RealTime in Head though, I assume. ;->
>
> Busy: You can always employ CheckGroup() and the like, although that
> gives you only the Asterisk view of things, not the SIP UAs point of
> view. Maybe sipsak and a custom SIP message can help, but that's of
> course outside of Asterisk.
>
> Philipp
>
>
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[Asterisk-Users] differentiating busy & not connected
folks, How can I determine whether a sip device is busy (channel unavailable), or it is not registered/connected? ChanIsAvail only checks the channel availability, so it's not specific enough. If I could get the output of "sip show peers" in an AGI script, I could create the functionality, but I'm hoping for some dialplan command sequence. Thanks, Glen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Disconnect supervision problem
On September 01, 2004 12:06 PM, Scott Laird wrote: This brings up an interesting point--disconnect supervision *mostly* works for me with a X100P in the US. The exception is when calls go to voicemail; I frequently end up with ~90 seconds of dialtone instead of a message or a clean disconnect. This has remained constant for 6 months, up through RC1. Thanks for mentioning this Scott, it made me try some different tests. We are using Asterisk as an H323-PSTN gateway. So the FXS interfaces are never used, only FXO. And it doesn't seem to matter which direction, PSTN > H323 or vice versa, Asterisk never catches the PSTN disconnect. I just tried dialing from an internal line (FXS) out to a pstn number and then hung up the far-end. Asterisk caught it. So it appears DS is working when bridging Zaptel to Zaptel but not Zaptel to (some) applications and channel drivers. With SIP, DS appears to work when the SIP-phone calls out and the (pstn) far-end disconnects, but not the other way around. According to the asterisk-console, when a pstn callers connects: after they hang up, asterisk will always timeout and then hang up. It never catches the hang up when it actually happens. And also, "zap show channel x" reports the channel is "offhook" even though it isn't (and will still answers calls). At Digium-support's request, I updated to CVS-HEAD-08/31/04-07:58:19. But the problem persists. Anyone else having (or had or fixed) this problem? Cheers Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO Disconnect supervision
Does anyone have disconnect supervision working on their TDM400P or X101P cards (to/from the telco)? Googling "telco disconnect supervision," I found the following description: http://mirror.lcs.mit.edu/telecom-archives/telecom-archives/TELECOM_Digest_Online/1559.html [cut] Re: How Do I Get "Kewlstart" From my Phone Company? Paul A Lee Tue, 24 Aug 2004 16:58:16 -0400 ... It appears that "kewlstart" is just a coined name for loop signaling with disconnect supervision. Disconnect supervision is also called "calling party control", "forward disconnect", "open loop disconnect", "open switch interval", "adjunct control", and perhaps other names. What is supposed to happen is that the CO switch (or other switch serving as the office end) will remove battery voltage from the loop for about 250 ms within 6 seconds after the far-end party disconnects. As far as I can tell, most CO switches now seem to provide disconnect supervision by default on loop-start lines. Consequently, it can be difficult to find someone at telco who knows anything about it. ... As for availability on residential service, just check your current loop-start line(s) with a voltmeter and see if it drops toward zero for about 250 ms when the far end disconnects from the call. If there's no disconnect supervision, you'll see the voltage stay at about 7-8 VDC when off-hook, and about 48 VDC on-hook. [/cut] I can hear what sounds like a battery drop on my lines when the far-end disconnects but asterisk doesn't seem to detect it. Anyone have ideas. Or experience with this? Regards Glen Johnson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Click to Call
Just write a CGI script that places a file in in the outgoing calls directory. /var/spool/asterisk/outgoing, I believe. This will accomplish what you're wanting. -g On Mon, 2004-08-09 at 11:45, Andrew Thompson wrote: > Andrei Goncalves wrote: > > Hello !! > > > > I saw in FWD site a phone on the web.. (click 612 link) > > http://www.freeworlddialup.com/advanced/beta_programs > > > > I´d like to have this application in my intranet.. click on my name, > > than > > calls my number.. > > I´d also like to see that phone on the web... as an option > > > > How can I do that ? > > Is it possible to download ? > > Any related link ? > > A couple of things: > > 1) It's activeX, which means it won't run outside of IE. > > 2) They openly state it requires features that mean it may only run in XP > and ME. > > 3) Even after I went to IE and ran the link(I’m using XP), it wouldn’t call > out for me. > > - > Andrew Thompson > http://aktzero.com/ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Iaxy issue
For anyone interested, the banshee screen I was experiencing was due to my cordless phone. I used a normal corded phone without separate power & it was fine. I suppose there was some type of power overload that the iaxy couldn't handle. -g On Tue, 2004-06-22 at 17:20, Andre Gironda wrote: > I've had an IAXy for about 3 weeks and have not heard a banshee > scream. I use it constantly, and have had cofigurations using a local > Asterisk machine in my house as well as Gafachi and VoicePulse. > VoicePulse was pretty awful. I am going to setup NuFone just so I can > compare the 3. > > -Amdre ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip over h323
This will explain the basics of * configuration: http://www.digium.com/handbook-draft.pdf After reading this document, you will understand contexts. -g On Tue, 2004-07-27 at 10:53, Thomas Kuepper wrote: > Am 27.07.2004 um 15:37 schrieb Philipp von Klitzing: > > > Hi! > > > >> Now my question. Is Asterisk an full h323 gatekeeper like openh323? > > > > No, Asterisk has no gatekeeper functionality, at least not yet. > > > >> And how can i tell asterisk to sent all none SIP-ip calls to the > >> gatekeeper over h323? > > > > One (standard) way to solve this is to place incoming H.323 into their > > own context (e.g. "from-h323"), and incoming SIP calls into another > > context (e.g. "from-sip"). Now define your Dial() statements as > > needed. > > > > ok, thx. what do you meen with context? wherre must i place this? > > > Philipp > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Thomas Küpper > > 01063 Telecom GmbH & Co. KG > Mottmannstr. 2 > 53842 Troisdorf > > Telefon: 02241-9434-506 > Telefax: 02241-9434-846 > > E-Mail: [EMAIL PROTECTED] > E-Mail: [EMAIL PROTECTED] > Homepage: http://www.01063telecom.de > > --- > Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im > Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der > vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede > unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da > wir > nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht > enthaltenen > Informationen garantieren koennen, schliessen wir die rechtliche > Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir > verweisen in diesem Zusammenhang auch auf die fuer uns geltenden > Regelungen > ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem > Inhalt, die in den bank- bzw. unternehmensueblichen > Unterschriftenverzeichnissen bekannt gemacht werden. > --- > This message is confidential and may be privileged. It is intended > solely > for the named addressee. If you are not the intended recipient please > inform > us. Any unauthorised dissemination, distribution or copying hereof is > prohibited. As we cannot guarantee the genuineness or completeness of > the > information contained in this message, the statements set forth above > are > not legally binding. In connection therewith, we also refer to our > governing > regulations of concerning signatory authority published in the standard > bank > or company signature lists with regard to the legally binding effect of > statements made with the intent to obligate us. > --- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard T100P in 1U
Yes, presuming the 1U can accept any standard pci card. -g On Fri, 2004-07-23 at 18:06, Sathya wrote: > Hi, > > Can a wildcard T100P be installed in a 1U server ?? > > Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dropping g729 frames
I'm getting this error continuously when sending to a cisco 5300: frame.c:120 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The connection is highly intermittent, sometimes there's a ring, other times there is not. Is there a way to completely disable vad support in *? -g ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel red alarms with e&m wink
I'm getting red alarms with my T100P card that last from 4-15 seconds. They seem to happen randomly, every couple of days. Has anyone seen this behavior, or does anyone have any ideas regarding what would be causing it? Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] random red alarms with t100p
I'm getting random Red Alarms with my T100P. I am using cvs zaptel from July 13, & have tried using remove clock source & local clock source. There doesn't seem to be any warning: every couple of days the line goes down, & then around 5 seconds later, it comes back up. If anyone as seen this behavior, or has any ideas about where to look, please let me know. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Questing regardning dialplans on a Cisco 5350
The call is inbound on the pots dial-peer, so you should use incoming called-number, as opposed to destination-pattern. dial-peer voice 1 pots incoming-called number [0-9]T no digit-strip direct-inward-dial port 3/0:D I'm not familiar with the [0-9] syntax, but if it works, ok. I usually use "." Also, you can specify the sip destination directly in the dial-peer, which makes using sip with the cisco's more flexible unless you're using a separate sip proxy. session protocol sipv2 session target ipv4:5.5.5.5 -g On Wed, 2004-07-14 at 07:27, [EMAIL PROTECTED] wrote: > > Hi. > > > If I use a Cisco as a PSTN termination GW and need to route all incoming > isdn calls to my asterisk and all outgoing calls from asterisk via the > cisco out to pstn, how do I do that ? > > > in the cisco I have this: > > dial-peer voice 1 pots > destination-pattern [0-9]T > no digit-strip > direct-inward-dial > port 3/0:D > ! > dial-peer voice 50 voip > destination-pattern [0-9] > voice-class codec 1 > session protocol sipv2 > session target sip-server > no vad > dtmf-relay rtp-nte > ! > > > --- > > But theese to dialplans seem to interrupt each other. > > When an incoming call from PSTN goes through this the pattern can be > matched by the first, and then be routed ot on the PSTN again, creating > a loop. > > How do I do this in the smartest and easiest way ? > > /Mike > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 0h323/ h323-registration
> As I understand, oh323 channel cannot act as h323-gatekeeper and it just role as a simple h323-GW ( am i right micheal). Correct. > If this is the case, how can I register my h323 EP ? shall I user a 3rd party GK like (gnu or mvts)? Correct. Mvts is expensive, OpenH323gk is fantabulous, easy to configure, & easy to modify. (http://www.gnugk.org) > plz explain more about h323 registration. This will give you an understanding of what goes on: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a00800c5e0d.shtml -g On Tue, 2004-07-13 at 22:50, mohammad mirzaee wrote: > HI ALL > HI MICHAEL; > > > Thanx Michael for your help, oh323 compiled successfully. > > As I understand, oh323 channel cannot act as h323-gatekeeper and it > just role as a simple h323-GW ( am i right micheal). > > If this is the case, how can I register my h323 EP ? shall I user a > 3rd party GK like (gnu or mvts)? > > plz explain more about h323 registration. > > > > > warmest regards > mohammad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel debugging tools
> But what exactly are you trying to debug ? Specifically, I want to determine when the DTMF tones are being sent over the channel. I'm connected to a NACT switch using cas, & there is a delay of 5-8 seconds from when asterisk begins the Dial command to the time when the NACT switch connects the call. This behavior does not occur with other equipment connected to the NACT switch, so I'm trying to narrow down the problem. > * you could issue "set verbose 10" on the asterisk CLI I've tried "set verbose 10" as you suggested, followed by a "debug Zap/1-1" as soon as the call is attempted. I get the following message repeatedly until the call is connected: "<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1]" Any ideas or directions are appreciated. -g On Mon, 2004-07-12 at 19:14, C. Maj wrote: > On Mon, 12 Jul 2004, Glen Hinkle waxed: > > > Are there any debugging tools for the digium zaptel cards that would > > report the activity on the line, such as DTMF and/or connection > > protocol? > > * zttool is in the zaptel source directory > * you could issue "set verbose 10" on the asterisk CLI > * you could issue "pri debug span x" on the asterisk CLI > > Also, try getting a PRI trace from your telco. > > But what exactly are you trying to debug ? > > --Chris > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel debugging tools
Are there any debugging tools for the digium zaptel cards that would report the activity on the line, such as DTMF and/or connection protocol? I'm looking to debug the connection with a T100P, & I don't have $2000 for a T1 test set. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using Cisco AS5350 as pstn GW .. one-way audio problem
What's your relevant dial peer & sip.conf config? -g On Fri, 2004-07-09 at 03:49, Mikael Andersson wrote: > Glen Hinkle wrote: > > I assume the pstn is your * system. > > Can you get audio both ways if you send the traffic back to *? > > > > pstn -> as5350 -> pstn ? > > > > -g > > > > > > > > Iuse the as5350 for termination at my telco, so it's physicly located there. > When I call pstn -> as5350 -> (sip) asterisk, I can hear the audio from the > asterisk, but audio from pstn will not get through. > > > I tried: psth --> as5350 --> sipphone. and the same result. I can hear > the sipphone but the sipphone cannot hear me. > > > the as5350 is connected to my telco with dual trunked E1's > > > /Micke > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Cisco AS5350 as pstn GW .. one-way audio problem
I assume the pstn is your * system. Can you get audio both ways if you send the traffic back to *? pstn -> as5350 -> pstn ? -g On Thu, 2004-07-08 at 14:09, [EMAIL PROTECTED] wrote: > Hi all. > > I have a strange problem, I've got a AS5350 hooked up to a telco using > two trunked E1's > > The 5350 should only act as a GW to a sipproxyserver. > > THe thing is it seems to be only oneway audio? > > There are no firewall at all, and the audio still only get one-way > > When I call from pstn --> as5350 --> sip-sip-phone I can here the > sip-phone ,, but the sipphone cannot her the pstn-phone. > > What could be the reason for this ? > > /Mike > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel DTMF delay
By the way, I'm not using the "r" option - or any options for that
matter. Here is my what I'm testing with in extensions.conf:
exten => _.,1,Dial(ZAP/g1/${EXTEN})
g1 is a group to a T1 using e&m wink.
-g
On Tue, 2004-07-06 at 15:44, Glen Hinkle wrote:
> Has anyone noticed a delay in sending DTMF to zaptel devices?
>
> I have a T100P connecting to an NACT telecom switch. All calls are sent
> just fine, but there are 4 seconds of delay between when the channel
> goes off-hook & the digits are collected by the NACT switch.
>
> I also have some cisco 5300 boxes connected, but there is no delay.
> Any ideas?
>
> -g
>
>
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[Asterisk-Users] zaptel DTMF delay
Has anyone noticed a delay in sending DTMF to zaptel devices? I have a T100P connecting to an NACT telecom switch. All calls are sent just fine, but there are 4 seconds of delay between when the channel goes off-hook & the digits are collected by the NACT switch. I also have some cisco 5300 boxes connected, but there is no delay. Any ideas? -g ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and H323
I would like know if the asterisk handle each protocol > (SIP and H323) separatedly or if the asterisk > translate the protocol?!?! Yes & yes. Though getting h323 to work seems to vary from system to system. -g On Tue, 2004-07-06 at 14:34, Giscard Fernandes Faria wrote: > Hi guys, I am a newbie in asterisk system. And I wanna > to make some questions. > > I already had a system to solve my VoIP solution, but > this system only accept the SIP protocol. Therefore I > thinking to using the asterisk like a "middle" to > redirect the H323 calls to my existing system!!! > > I would like know if the asterisk handle each protocol > (SIP and H323) separatedly or if the asterisk > translate the protocol?!?! > > If the first statement is true I can use the asterisk, > if not I would like ask if anyone confront a similar > problem and what the solutions used. > > Thanks. > > Giscard > > > > > > > ___ > Yahoo! Mail agora com 100MB, anti-spam e antivírus grátis! > http://br.info.mail.yahoo.com/ > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rh9, asterisk HEAD, & asterisk-oh323-0.6.3a working
I have no new information, just a note of encouragement to those traversing the bowels of h323: I've been trying to get h323 working with asterisk for several months now, trying with chan_h323 & chan_oh323 with all kinds of different combinations. As with several folk on the list, I've had no luck. Either I had no audio, or I could only receive calls, or I could dial but no had no audio. I finally got it working today with the oh323. It seems to work flawlessly in lab settings with both ulaw & g729. The command line options for oh323 are odd, but whatever, it works. I'm using RH9 w/ a cvs HEAD checkout from "Jul 2 17:54 EST". I would post how I did it, but I only followed the README. -g ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323 no audio both ways
Sorry, Tom, I missed this message when it came through. It seems this problem is a continuing issue among the asterisk folk. Tell me, what versions of IOS have you tested with, & do you have any of the h323 options enable/disabled in the 5300? -g On Fri, 2004-06-18 at 21:09, T. Chan wrote: > Hi Glen, I have had the same problem for quite awhile, since around > February, all cvs codes that I have tried, and with h323, I have been > getting no audio. I am forced to stay with mid-Jan version of the cvs > because of this. I tried using ulaw, g729, but same results, I have in a few > occasions dropped a few lines here to ask for advice, but no response, may > be we could try to exchange some ideas. Thanks > > TC > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > [EMAIL PROTECTED] > Sent: Monday, June 14, 2004 6:46 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] chan_h323 no audio both ways > > > I've compiled chan_h323 with the latest cvs code, but my calls don't > pass audio. > > The call connects just fine, as there are no errors reported on either > side, nor in a traffic examination with ethereal. > > I've tried the following: > > voip phone -> asterisk -> asterisk -> voip phone > voip phone -> asterisk -> asterisk > zap -> asterisk -> asterisk > zap -> asterisk -> cisco > cisco -> asterisk > > I'm using ulaw on all connections. > > Any clues, ideas, or directions would be appreciated. > > > Thanks, > Glen > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS
- I'm here with you on this one. I've not been able to figure this out - I triple & quadruple checked that I have the right versions of pwlib & openh323, & I've followed all recommendations in the README, yet I still do not have audio in both directions. I'm also using a cisco 5300, & there is no firewall. Tcpdump has revealed the following: when calls are made from the 5300 to asterisk, the 5300 sends continual udp packets, but asterisk doesn't seem to be responding. when calls are made from asterisk to the 5300, no udp packets are sent. It should be noted that when the calls are made using sip, everything works just fine. -g On Fri, 2004-06-25 at 10:54, Sebastian Nocetti wrote: > hello all, I am having a trouble with Audio using h.323 channel... > > I am doing this > > Call comes into cisco 5300 and is sent to Asterisk, asterisk catch > call with h.323 driver and send call to a SoftSwitch that routes the > call, I can see log debug telling me, CALLED XXX, and then RINGING, > and I can hear ring tones... but when call is answered, I DONT HEAR > ANYTHING... I am using lastest ASTERISK download somebody can help > me to solve this problem > > thanks..!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream HT-286 and NAT
I'm using this scenario & have had no problems. When you say that HT-286-2 begets silence, do you mean it does not ring, or that when you pick up after the ring there is no audio? -g On Fri, 2004-06-18 at 14:31, Nathan Martinez wrote: > I have 2 Grandstream HT-286 devices and an Asterisk server. The * > Server is not using NAT and has port 5060 opened up. One HT-286 is > using traditional NAT and the other HT-286 is behind a residential DSL > router/firewall. I have the HT-286 setup as the "DMZ Host" in the > router/firewall so that all incoming connections are forwarded to the > HT-286. > > HT-286-1 == NAT FW == * Server === Router/FW == HT-286-2 > > In the setup for HT-286-2 , I have filled in the "Use NAT IP" field with > the public IP for that location. I did the same thing for HT-286-1 and > then I mapped a public IP to its private IP in the NAT FW. At this > point, the two devices can call each other without any problems. > > I want to use the HT-286 for our traveling users who will never know > what their IP is. When I remove the "Use NAT IP" entry on HT-286-1 as > well as remove its direct IP mapping from the NAT FW, HT-286-1 can > register with the * Server, but when I try to call HT-286-2, all I get > is silence. If I do a 'sip show channels' it shows that the call is > connected. Here is what I have in my sip.conf for these two units: > > [305] > type=friend > host=dynamic > nat=yes > qualify=100 > > [307] > type=friend > host=dynamic > nat=yes > qualify=100 > > Has anyone used these units in this scenario? Does anyone have any > hints as to what I can try to get this working? Your help is much > appreciated. > > Nathan Martinez > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iaxy issue
Folks, Randomly, when the phone is taken off-hook, the the Iaxy produces a irritating banshee scream as opposed to a dial-tone. Cycling the power fixes the issue, & sometimes it magically goes away by itself. Has anyone experienced this issue & potentially fixed it? I'm using asterisk CVS head as of jun 17 2004. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can i get the last codec_g729.so
g729 must be licensed from voiceage, through digium. You can purchase licenses from digium's online store @ digium.com. -g On Thu, 2004-06-17 at 16:37, Carlos Medina wrote: > Hi there, im having some problems with my asterisk box, it seems codec > is the principal cause of it. Reading in some forums i found that i > can get the new codec_g729 from > ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so i > checked it but the directory new_codec_binary doesnt exist. > > Anybody knows where can i found it?? > > Thanks for your help. > > Carlos Andres Medina > > > __ > Do you Yahoo!? > New and Improved Yahoo! Mail - Send 10MB messages! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip register and nat
As I understand it, & if I understand you correctly, the register parameter is for the client side. The nat=yes parameter is for the server side, so it has nothing to do with your register statement. The sip debug displays "no nat" because sip.broadvoice.com is not "behind" the nat, it's "in front" of it. -g On Tue, 2004-06-15 at 17:29, Kubat, Philip wrote: > This may be a newbie SIP/NAT question. If so I am sorry. But any help > would be appreciated. My Asterisk server is behind an ipchains box and I am > trying to connect to Broadvoice. All works fine without the NAT. I have a > global nat=yes prior to my register, but the sip debug allows shows "no > nat)". Is this "label" issue, and am I barking up the wrong tree? > > Sip.conf > nat=yes > register => 1235551234: password @sip.broadvoice.com:5060/1235551234 > > sip debug > Retransmitting #5 (no NAT): > REGISTER sip:sip.broadvoice.com SIP/2.0 > Via: SIP/2.0/UDP 1.2.3.4:5060 > ... > Event: registration > Content-Length: > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_sms - rocks!
Forgive my ignorance, but I believe this would be of benefit to many listening in. I'm taking it that SMS can be done with PRI, however I don't understand the specifics of it. I have a T100P card connected to a PRI - can I send/receive SMS message with it? I'm not having much luck, & I figure I'm missing something. -g On Mon, 2004-05-10 at 11:19, brian wrote: > Its all at the URL he posted.. script and all and an example > > http://www.automated.it/asterisk/sms.html > > bkw > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of reseaux > > Sent: Monday, May 10, 2004 10:11 AM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] app_sms - rocks! > > > > Dear Andy > > Very nice apps but i cant have succesfull with it.. :-( Can you give > > me some > > info about how your script work and is possible to send sms from my phone > > and > > send with asterisk trought a PRI or FXS to another phone with sms > > capability? > > Thanks in advance > > Dimitri > > > > PS: i have try to send from my Phone to Asterisk and seems work the > > app_SMS!! > > but after that what is need? > > > > On Monday 10 May 2004 12:44 pm, Andy Powell wrote: > > > Ok, > > > > > > I just thought I'd publicly pat Adrian Kennard (revk) on the back for > > this > > > application. This is an excellent contribution and gets my vote for app > > of > > > the year. > > > > > > For those that aren't aware app_sms allows you to send/receive fixed > > line > > > sms messages from asterisk. ( you can take a look at a quick page > > showing > > > this http://www.automated.it/asterisk/sms.html ).. I should point out > > that > > > that this works over IAX2 as well, this means IAXy's with sms capable > > > phones work, and also messaging between asterisk boxes also works. In > > fact, > > > although I'm guessing, if there was a sip phone that did fixed line sms > > I > > > bet that would work too. > > > > > > So big pat on the back for Adrian Kennard (revk), Thanks for all the > > effort > > > and for sharing! > > > > > > Andy > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gsm playback garbled over sip
I'm new to all this so I don't know where to look, some tips would be most appreciated. I've enabled sip debugging and everything looks fine on the client and server side. Using Linphone on the client side. GSM playback from the server console is fine. I've used Linphone to connect to a vegastream VoIP system so I know if that installed and working. I'm basically just trying to get the sample configs working, dialing in to [EMAIL PROTECTED] It connects and playback of the demo ensues, but at the client end it's unrecognisable garbage. Any hints ? -- Glen Gray <[EMAIL PROTECTED]> 17 Dame Court Senior Software EngineerDublin 2, Ireland Lincor Solutions Ltd. Ph: +353 (0) 1 6746413 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] avaya and linux
Does anyone know if avaya voip product is running linux under the hood? Thanks, /glen -- Glen Ford [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gnophone installation problems
On Fri, 2004-04-02 at 16:01, Martin Mielke wrote: > Hi all, > > I installed all needed RPMs by GnoPhone to be installed without problems > but when attempting to install GnoPhone itself I get this message: > > # rpm -Uvh gnophone-0.2.4-1.i386.rpm > error: Failed dependencies: > mozilla >= 0.9.2 is needed by gnophone-0.2.4-1 > libgtkembedmoz.so is needed by gnophone-0.2.4-1 > libgtksuperwin.so is needed by gnophone-0.2.4-1 > > I'm using Mozilla 1.7a installed from a tarball. The needed libraries > are just there: > > # locate libgtkembedmoz.so > /usr/local/mozilla/libgtkembedmoz.so > I presume you used prebuilt binary rpms then. They will most likely have been linked against /usr/lib/mozilla-1.x/ Try getting the gnophone source rpm rebuilding that with rpmbuild --rebuild gnophone.src.rpm -- Glen Gray <[EMAIL PROTECTED]> 17 Dame Court Senior Software EngineerDublin 2, Ireland Lincor Solutions Ltd. Ph: +353 (0) 1 6746413 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] checkout ztdummy
On Fri, 2004-04-02 at 10:43, ePyron Felix Deierlein wrote: > I have tried to follow, but I did not know, wich modul I had to check out.. > Checkout the Zaptel CVS module. Edit the Makefile in the Zaptel dir to uncomment the ztdummy source. -- Glen Gray <[EMAIL PROTECTED]> 17 Dame Court Senior Software EngineerDublin 2, Ireland Lincor Solutions Ltd. Ph: +353 (0) 1 6746413 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linphone connecting to default Asterisk Samples
Hey Guys, I've setup Asterisk on a PC here in the office. I installed the sample configs. With a the sound card connected to some speakers and launching asterisk with the "-vvvc" command line, I can issue the "dial" command and hear the playback of the recorded message. When connecting to the asterisk server from my PC using Linphone I connect to [EMAIL PROTECTED] The problem here is that I am hearing playback of something but is completely garbled. Has anyone got any pointers to what could be the problem here. Could it be Linphone's sip implementation ? I've not tried with any other softphone for linux, yet. -- Glen Gray <[EMAIL PROTECTED]> 17 Dame Court Senior Software EngineerDublin 2, Ireland Lincor Solutions Ltd. Ph: +353 (0) 1 6746413 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re Hardware requirement -Asterisk
On my Linux box mii-tool yeilds the following which shows 100mbs full duplex. [EMAIL PROTECTED] gford]# mii-tool eth0: negotiated 100baseTx-FD, link ok /glen [EMAIL PROTECTED] wrote: My ADSL speed is Uplink 128kbit and Downstream 512kbit. The mii-tool does not tell whether eth0 is in full-duplexed mode. It just say that it is 100baseTx. David Kwok -- Glen Ford [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware requirements - asterisk
dkwok wrote: In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to do voice processing? I am running xlite on 2 pc and making calls through iax, FWD and back to my incoming call menu. Voice degradation happens. David Kwok David, I too am new to Asterisk. Howerver I know howto check nic card. Use mii-tool command. /glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN messenger and *
Speaking of MSN/Windows Messenger, how does one call someone? Using the configuration specified, I've registered it with Asterisk, but it requires that I add a Passport contact. Does anyone have experience calling a sip endpoint without it being a Passport account? -g On Mon, 2003-12-22 at 20:42, Balaji NJL wrote: > use this > > > [3001] > > type=friend > > ;username=3001 > > ;fromuser=Craig1 > > ;secret=secret > > host=dynamic > > mailbox=3001 > > context=sip > > dtmfmode=info > > auth=plaintext > > > > make sure ur MSN version is 4.7.0105. > > > > -B > > > - Original Message - > From: Craig Waddington > To: [EMAIL PROTECTED] > Sent: Monday, December 22, 2003 10:10 AM > Subject: [Asterisk-Users] MSN messenger and * > > > Sorry for the late reply. > > > > I try port 5060 and it just knocks me back straight away, I > cant see it even try to authenticate in the CLI. > > > > X-lite works both inside the LAN and outside using SIP. > > > > Messenger version = 4.7 > > > > John I will try your suggestion with sip.conf thanks for the > help. I notice a few differences, I seem to be missing some > bits.. > > > > Its like it is trying to authenticate with the Linux box and > not asterisk. > > > > Sip.conf > > > > [general] > > port=5060 ; Port to bind to > > bindaddr=0.0.0.0; Address to bind to > > context=sip ; Default for incoming calls > > allow=ulaw > > allow=alaw > > allow=gsm > > allow=ilbc > > > > > > [3001] > > type=friend > > username=3001 > > fromuser=Craig1 > > secret=secret > > host=dynamic > > mailbox=3001 > > context=sip > > dtmfmode=info > > > > I found 3 guides and each one seems to be a bit different and > use different ports. > > > > I am using the X100P, it is a home system, to reduce call > charges for my family overseas. > > > > If I can get Messengger working it will be easier to talk > them through the setup. > > > > > > > > > > __ > Do you Yahoo!? > Yahoo! Photos - Get your photo on the big screen in Times Square ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
