[asterisk-users] define extension to send calls to gatekeeper
hello every one,
i have an asterisk system and want to act as gateway and send calls to
cisco gatekeeper.
this is my h323.conf file:
[general]
port=1720
binaddr=192.168.0.YY
context=from-trunk
faststart=yes
h245tunneling=yes
gatekeeper=192.168.0.XX //cisco address
progress_setup=8
progress_alert=8
dtmfmode=rfc2833
jbenable=yes
jbforce=no
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=fixed
jblog=no
and this is extension that i defined for it in extensions.conf:
exten=_2.,1,Dial(H323/2${EXTEN:1})
(if i define my extension like
(exten=_2.,1,Dial(H323/to-cisco/2${EXTEN:1}), asterisk returns core dump
error.
but, i can't send my calls to cisco gatekeeper. do you have any suggestion
what is wrong with my configuration?
thanks in advance
SAM
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[asterisk-users] MOH don't work after update
Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 thanks for your help Olivier -- _ -- 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] Issue dialing out
On Sat, Jun 15, 2013 at 04:24:21PM -0400, Andre Goree wrote: Thanks so much for your suggestions. I'm running 1.0.x (yes, archaic, and in fact my actual task is migrating this system to asterisk11+Freepbx -- very fun in and of itself without regards to this issue...but I digress), and so I needed to run pri debug span span, which I've now done. I attempted the call again have pasted the debug output here: http://pastebin.com/cHHnMfh6 You mentioned the telco receiving a DISCONNECT almost immediatly. Your debug is only up to a PROGRESS. I only have experience with euroisdn but callflow would be: -SETUP -CALLPROCEDING -PROGRESS -CONNECT -CONNECT ACK -DISCONNECT (eg from caller) -RELEASE -RELEASE COMPLETE But PROGRESS means the recipient is generating some audio (your unreachable message?). If this is an error message you would expect a RELASE from the telco after the recording if the caller doesn't hangup first. You should study the difference of zap-zap and sip-zap callsetup. -- _ -- 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] I need a second opinion on a new phone system deployment
Thanks again to everyone that's responded thus far. I have once again bundled the questions and answers into a single email, and am responding below. On 6/14/2013 9:43 AM, Nunya Biznatch wrote: Howdy All, They say opinions are like belly buttons, everybody has one. (that's the clean version of the saying). So I'm asking for yours. I hope you see it as a fun exercise. I'm designing a phone system from the ground up. Will be about 1000-1300 seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23 buildings. Userbase is emergency services organization, 24/7/365 operation. Down time is not an option, but blips are acceptable. Repair time is immediate. We need failover for the failover essentially. However, money is a major factor, so I have to do it all for nothing. So here's what I'm thinking. Please throw in your 2 cents. Network will be separate for phones. Fiber infrastructure available between buildings as well as copper. Internet access will be limited to a single administrative console on a temporary basis, and then only when remote 3rd party support is required. Access for 3rd party support will be supervised through remote access tools such as VNC, GoToMeeting, etc... etc... System will have zero access to local data network. This means all ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be specific to the phone system. Yes, I know some responders at this time will become fixated on me gaining this connectivity. It ain't gonna happen. It's not an option. Period, end of story. These are the parameters I must work within. Trying to fix that will be a non-starter. The phone system will upgrade an existing TDM-based system. Mitel SX2000 with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect at least a one to two-year transition, meaning we will have time to find problems, work bugs, and learn over time, with minimized impacts. It also means we'll be supporting two systems for some time. PBX is 97% serving your basic phone on the desk. Nothing special. Customers expect the usual list of features. There will be a goodly number of hints required for BLF on maybe 150 phones. There is one office of about 30 phones in a call-center environment that will need that service. They would be considered low volume (but don't tell them that). My Skills... I am not a Linux kung fu master, but I have built and managed my share of Linux servers on mutiple Linux flavors. I am a DCAA, having been through formal training, and have been playing with Asterisk for years, but always in fits and spurts and never in a live environment so I am by no means a kung fu master there either. I have started dabbling with virtualizations via XEN, but I am not comfortable enough with it to go live this first round. I can see myself implementing it in about three years once we're totally comfortable with what we have, so I can then have time to get that skill sorted. I was a network engineer for the US no3. telecom for a number of years, 10-years in comm-electronics in the military before that. Telecom my entire career. I've got the kung-fu to handle the network side of the house, and having administrated multiple PBXs for decade-plus, I've got the concepts down. No plans to build databases for things like directories, etc... I'm not greatly confident in those skills, and to date, haven't found anything that really stands out that would make me require that. You may think otherwise, so please chime in. I say that, but at the same time I recognize I may require a GUI interface once fully deployed to allow lower-skilled people to follow the motions to complete simple moves, adds, and changes. I'm fighting the uphill battle that is the GUI is new, CLI is old mentality. System will use G.722 for VoIP Phones. So there's the groundwork. Here's the hardware plan. Plan is to build my own servers following industry standards (ATX) and using industry standard equipment. Why? Spares? Whether redundant or not, I will still have spares for the most common elements on the shelf so equipment can be returned to service as quickly as possible. This will also allow me to be comfortable with more basic server configurations and help keep cost down. For example, Servers with single power supplies vs. dual. Also, components will be standardized for all equipment to aid in supply requirements. First the layout. 2-servers acting as gateways. Each handling 2 PRIs for outside trunks. They'll also handle the analog ports. Failover will be in the form of degraded trunk access if one should fail, but the second will be able to support services in degraded fashion. 2-servers acting as VoIP PBX. A primary and a spare. Meaning one will be capable of handling the load of the entire system, and the other will pickup when the other dies, an active/passive cluster. Will also take care of voicemail. Use of heartbeat, pacemaker,
[asterisk-users] PCI Passthrough of T1 cards
Anyone try this? I saw a post here: http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html But not sure if it's possible. What I am asking is if there are any T1 cards with virtual functions implemented in their drivers to allow pci-passthrough? Kind Regards, Nick. -- _ -- 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] MOH don't work after update
On Sun, Jun 16, 2013 at 2:43 AM, Olivier CALVANO o.calv...@gmail.comwrote: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 Do you have the format_mp3 module loaded? Add-on modules are in the addons subdirectory. Typically, these modules are not built and installed by default, and have to be enabled in menuselect. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] MOH don't work after update
i use the package centos, i can't use menuselect no ? but i think's that is loaded: ipbx*CLI module load format_mp3.so Unable to load module format_mp3.so Command 'module load format_mp3.so' failed. [Jun 17 04:56:42] WARNING[8910]: loader.c:892 load_resource: Module 'format_mp3.so' already exists. ipbx*CLI 2013/6/16 Matthew Jordan mjor...@digium.com On Sun, Jun 16, 2013 at 2:43 AM, Olivier CALVANO o.calv...@gmail.comwrote: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 Do you have the format_mp3 module loaded? Add-on modules are in the addons subdirectory. Typically, these modules are not built and installed by default, and have to be enabled in menuselect. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
