[asterisk-users] AstLinux 0.7.3 released
The AstLinux Team is happy to announce the release of AstLinux 0.7.3. This update contains mostly bug fixes and security updates. All current users of AstLinux are encouraged to update to this release. Updating can be performed from the web interface or from the command line using a few simple commands. For the Changelog and other instructions, please visit: http://www.astlinux.org/release/073 Enjoy, Darrick -- _ -- 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] func SHARED, how to use?
27.09.2010 16:25, Philipp von Klitzing пишет:
> Hi!
>
>
>> Well, only problem I see, is to how pass channel name from macro to h
>> extension...
>>
> SHARED() or CDR(userfield)
>
> Philipp
>
>
>
Looks like I still don't understand how SHARED works :-(
Let's say, I dial my softphone:
exten=6052,n,Dial(SIP/6052,,M(test))
I have macro:
[macro-test]
exten => s,1,NoOp(test ${CHANNEL} )
exten => s,n,Set(SHARED(foo,${CHANNEL})=456 )
then I want get this 456 in h:
exten => h,1,NoOp(${SHARED(foo,${CHANNEL})})
if use $CHANNEL then channel is DAHDI and I get nothing.
As I understand I have to provide SIP channel name, but how can I export
it's name from macro to h?
Thank you!
--
_
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Re: [asterisk-users] SCCP (skinny) phone behind NAT: RTP dest addr wrong
> packet tracing RTP at the border router shows (on the LAN segment > of the asterisk host), e.g. > 11:43:54.380567 IP (tos 0x0, ttl 64, id 737, offset 0, flags [DF], proto > UDP (17), length 200) >pbx1.cybertheque.net.20002 > 192.168.0.3.29112: UDP, length 172 > Shouldn't 'nat=yes' in skinny.conf force asterisk to use the public IP > of the phone? Setting 'nat=no' or commenting-out the 'nat=' statement does not change the incorrect behavior. Help still appreciated... Michael -- _ -- 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] NAT issue (i think?)
Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 9:05 PM, Jim Dickenson wrote: > Do you not need to do a ./configure command before make & make install? If > so issue the ./configure command again and see if that fixes the problem. > No, it does not exist for DAHDI. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Problems compiling Asterisk on Debian
Do you not need to do a ./configure command before make & make install? If so issue the ./configure command again and see if that fixes the problem. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 27, 2010, at 4:32 PM, Danny Dias wrote: > Thanks Jim, > > What do you mean with "redo" ? > > I did not run the ./configure, i'm installing dahdi-linux and just need : > make && make install > > The problem is when i issue make > > Thanks for your answer my friend! > > 2010/9/28 Jim Dickenson > Did you install the header files after ./configure was run? If so redo the > ./configure command and see what that does. > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > On Sep 27, 2010, at 3:57 PM, Danny Dias wrote: > >> Hello Paul, >> >> Here is the output of the commands: >> >> r...@sangoma-testing:/home# ls -la /lib/modules/ >> total 12 >> drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . >> drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. >> drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 >> >> r...@sangoma-testing:/home# ls -la /usr/src/linux >> lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> >> linux-headers-2.6.26-2-amd64 >> >> Seems to be OK, isn't? >> >> Thanks! >> >> >> 2010/9/27 Paul Belanger >> On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias wrote: >> > The same problem! >> > >> What is the output from the following? >> >> $ ls -la /lib/modules/ >> >> $ ls -la /usr/src/linux >> >> -- >> Paul Belanger | dCAP >> Polybeacon | Consultant >> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) >> blog.polybeacon.com >> >> -- >> _ >> -- 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 > > > -- > _ > -- 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 -- _ -- 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] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 7:36 PM, Danny Dias wrote: > Is that Ok? > $ uname -r Also what version of Debian? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 6:57 PM, Danny Dias wrote: > r...@sangoma-testing:/home# ls -la /lib/modules/ > total 12 > drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . > drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. > drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 > $ cat /lib/modules/2.6.26-2-amd64/build/.config > r...@sangoma-testing:/home# ls -la /usr/src/linux > lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> > $ cat /usr/src/linux/.config -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Problems compiling Asterisk on Debian
Sorry Paul, My mistake...i repeat the commands again: r...@sangoma-testing:/usr/src# ls -la /lib/modules/ total 28 drwxr-xr-x 7 root root 4096 2010-09-27 19:35 . drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. drwxr-xr-x 3 root root 4096 2010-09-27 19:29 2.6.26-1-amd64 drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 drwxr-xr-x 2 root root 4096 2010-09-27 19:35 2.6.26-2-openvz-amd64 drwxr-xr-x 2 root root 4096 2010-09-27 19:35 2.6.26-2-vserver-amd64 drwxr-xr-x 2 root root 4096 2010-09-27 19:35 2.6.26-2-xen-amd64 r...@sangoma-testing:/usr/src# ls -la /usr/src/linux ls: cannot access /usr/src/linux: No such file or directory Is that Ok? 2010/9/28 Danny Dias > Hello Paul, > > Here is the output of the commands: > > r...@sangoma-testing:/home# ls -la /lib/modules/ > total 12 > drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . > drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. > drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 > > r...@sangoma-testing:/home# ls -la /usr/src/linux > lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> > linux-headers-2.6.26-2-amd64 > > Seems to be OK, isn't? > > Thanks! > > > 2010/9/27 Paul Belanger > >> On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias >> wrote: >> >> > The same problem! >> > >> What is the output from the following? >> >> $ ls -la /lib/modules/ >> >> $ ls -la /usr/src/linux >> >> -- >> Paul Belanger | dCAP >> Polybeacon | Consultant >> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) >> blog.polybeacon.com >> >> -- >> _ >> -- 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
Re: [asterisk-users] Problems compiling Asterisk on Debian
Thanks Jim, What do you mean with "redo" ? I did not run the ./configure, i'm installing dahdi-linux and just need : make && make install The problem is when i issue make Thanks for your answer my friend! 2010/9/28 Jim Dickenson > Did you install the header files after ./configure was run? If so redo the > ./configure command and see what that does. > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > On Sep 27, 2010, at 3:57 PM, Danny Dias wrote: > > Hello Paul, > > Here is the output of the commands: > > r...@sangoma-testing:/home# ls -la /lib/modules/ > total 12 > drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . > drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. > drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 > > r...@sangoma-testing:/home# ls -la /usr/src/linux > lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> > linux-headers-2.6.26-2-amd64 > > Seems to be OK, isn't? > > Thanks! > > > 2010/9/27 Paul Belanger > >> On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias >> wrote: >> > The same problem! >> > >> What is the output from the following? >> >> $ ls -la /lib/modules/ >> >> $ ls -la /usr/src/linux >> >> -- >> Paul Belanger | dCAP >> Polybeacon | Consultant >> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) >> blog.polybeacon.com >> >> -- >> _ >> -- 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 > > > > -- > _ > -- 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
Re: [asterisk-users] Problems compiling Asterisk on Debian
Did you install the header files after ./configure was run? If so redo the ./configure command and see what that does. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 27, 2010, at 3:57 PM, Danny Dias wrote: > Hello Paul, > > Here is the output of the commands: > > r...@sangoma-testing:/home# ls -la /lib/modules/ > total 12 > drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . > drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. > drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 > > r...@sangoma-testing:/home# ls -la /usr/src/linux > lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> > linux-headers-2.6.26-2-amd64 > > Seems to be OK, isn't? > > Thanks! > > > 2010/9/27 Paul Belanger > On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias wrote: > > The same problem! > > > What is the output from the following? > > $ ls -la /lib/modules/ > > $ ls -la /usr/src/linux > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- 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 -- _ -- 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] Problems compiling Asterisk on Debian
Hello Paul, Here is the output of the commands: r...@sangoma-testing:/home# ls -la /lib/modules/ total 12 drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 r...@sangoma-testing:/home# ls -la /usr/src/linux lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -> linux-headers-2.6.26-2-amd64 Seems to be OK, isn't? Thanks! 2010/9/27 Paul Belanger > On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias > wrote: > > The same problem! > > > What is the output from the following? > > $ ls -la /lib/modules/ > > $ ls -la /usr/src/linux > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- 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
Re: [asterisk-users] Asterisk Redundancy
> Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com Bit out of my pricing. It must be possible to do it using downloadable open-source. Dan -- _ -- 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] Asterisk Redundancy
> How do you handle replicating voice mails? I do that by putting the voicemails into MYSQL and replicating that. -- _ -- 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] Asterisk as a distributed paging system
On 09/22/2010 05:49 PM, Matteo Fortini wrote: > I'm building a paging system composed of roughly 10 switches in daisy > chain, with an embedded box with a speaker and a microphone for each > switch. The embedded box runs my software. I'm not sure if this is much use to your setup - but I've implemented a public announcement system (in UK sometimes called tannoy) using a pc and a pair of loud speakers. I used linphonec command line sip client - with auto-answer set. This has the disadvantage, of course, that it can't be used as a phone as well, to make phone calls from the machine driving the speakers (unless you install another sip client) - just as a speaker. You might be able to run linphonec on an embedded platform as well - with a speaker attached. I know there were some compiled for the NSLU2/arm platform already. You will have to combine it with Page() to get multiple nodes working at the same time. I only used one. I did a write-up here - see if you find any useful information: http://forum.voxilla.com/asterisk-support-forum/asterisk-public-announcement-system-loud-ringer-bell-49339.html Sebastian > > I need the system to be resilient to any network partition, so that > anyone can send announces from any mic to all the reachable clients. I'd > need also to page a subset of all the speakers. > > I'm currently using some software I wrote which sends voice over > multicast RTP and coordinates all the sites with multicast messages. > > I don't own the switches so each site will be assigned an address by > DHCP, that's why I'm using multicast. > > I heard of asterisk and SIP as a possible alternative to my software, > and I'd rather use tested and widely adopted software. > > Is there a way asterisk could be of use, or would I need to bend it too > much? > > Thank you in advance, > Matteo > -- _ -- 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] Asterisk 1.8.0 Release Candidate 2 Now Available
On 10-09-26 02:55 PM, Ira wrote: > At 10:37 PM 9/24/2010, you wrote: >> You probably need to install libssl-dev then rerun ./configure. At >> least I did (Debian Lenny). Seems chan_sip needs res_crypto which >> needs libssl. > > Thanks, I tried to figure out what I needed but I failed. That was > it, though on CentOS it seems to be openssl-devel. FYI, this is no longer an issue as of today. I opened an issue per the Asterisk development team, and Tilghman fixed the issue. https://issues.asterisk.org/view.php?id=18062 The next release candidate will allow chan_sip to use, but not require, the OpenSSL development libraries. Thanks! Leif. -- _ -- 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] 1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality
On 10-09-26 01:00 PM, bilal ghayyad wrote: > First of all, I am looking to have the H323 Gatekeeper service available at > Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing > H323 gatekeeper functionality or not? > > Until 1.4.26.2 version, there is no h323 gatekeeper functionality. So, any > implementation for this feature has been done in the other versions? From what I'm aware of, no additional work has been done on the H323 modules available for Asterisk that implements any sort of gatekeeper functionality. Leif. -- _ -- 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] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...
Thanks guys. Amazing feedback. Sounds like 2.5" is a better choice for being less in size (easier access for voicemail for example), as fast as 3.5" HDD in RPM, and allows 6 HDDs per Node which allows more RAID choice. However, it does come to be more expensive. Thanks again, On Mon, Sep 27, 2010 at 10:57 AM, Benny Amorsen > wrote: > bruce bruce writes: > > > Other than the price difference (2.5" is more expensive and can't find > > many of the 1TB or so) is there any preference, advantage, or > > disadvatage of chosing 2.5" HDD or 3.5" when it comes to the server > > operations or Asterisk operation? > > There is no difference. Pick the server which offers the disk bandwidth > and I/O's per second which you need. > > Do you really need 1TB disks? If you do, be careful what you place on > those disks. Reading e.g. a voice mail or a speak off a large slow > platter which is busy writing CDR's does not sound good at all. > > > /Benny > > > -- > _ > -- 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
Re: [asterisk-users] How to pick a codec on the fly
> From: da...@debsinc.com
> To: dan...@tryba.nl; asterisk-users@lists.digium.com
> Date: Mon, 27 Sep 2010 13:30:08 -0500
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
> Sent: Monday, September 27, 2010 1:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote:
> > I'm trying to test an IVR system with recorded prompts and would
> > like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
> > ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and
#3
> > is slin; Need it the other way so I can do DAHDI--> IAX testing.
>
> exten => 1234,1,Set(_SIP_CODEC=alaw)
> exten => 1234,n,Goto(0234,1)
> exten => 2234,1,Set(_SIP_CODEC=slin)
> exten => 2234,n,Goto(0234,1)
>
> Should do the trick.
>
> --
>
> Daniel Tryba
>
> Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X.
> -- Executing [...@from-pstn:7] Goto("DAHDI/1-1", "default|s|1") in new stack
> -- Goto (default,s,1)
> -- Executing [...@default:1] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@default:2] Goto("DAHDI/1-1", "select-func|s|1") in new
> stack
> -- Goto (select-func,s,1)
> -- Executing [...@select-func:1] WaitExten("DAHDI/1-1", "5|m") in new
> stack
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> == CDR updated on DAHDI/1-1
> -- Executing [...@select-func:1] Set("DAHDI/1-1", "_SIP_CODEC=ulaw") in
> new stack
> -- Executing [...@select-func:2] Dial("DAHDI/1-1", "IAX2/xxx/332|30|m") in
> new stack
> -- Called xxx/332
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Call accepted by XXX.XXX.XX.XX (format gsm)
> -- Format for call is gsm
> -- IAX2/ffb-18075 answered DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> -- Hungup 'IAX2/xxx-18075'
> == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
>
> --
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah
Sent: Monday, September 27, 2010 1:40 PM
To: Asterisk Users
Subject: Re: [asterisk-users] How to pick a codec on the fly
i think it's SIP_CODEC now .. and not _SIP_CODEC?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
"Good" and "Bad" news
On a SIP call, SIP_CODEC still works; this same patch not built into iax
(apparently we don't want Asterisk 1 to set the codec for Asterisk 2).
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Re: [asterisk-users] Can't turn debug on in a 1.2 box
Thank you guys for the responses, actually that piece that was missing is "logger reload", only when I issue this command i can see the dedug turning on, i though "reload" is enough to reload everything but it seems like "logger reload" is specifically needed to apply changes, Thanks a lot Steve and Paul! On Fri, Sep 24, 2010 at 6:54 PM, Steve Edwards wrote: > > On Thu, Sep 23, 2010 at 10:06 AM, khalid touati > wrote: > > >> do you guys know how i can turn debug on or just know why it's not > >> getting enabled? > > On Fri, 24 Sep 2010, Paul Belanger wrote: > > > *CLI> set debug 15 > > *CLI> reload > > If you change these lines in the '[logfiles]' section of logger.conf and > enter 'logger reload' at the Asterisk CLI, you will get more than enough > debugging info on the console and in your syslog file (probably > /var/log/messages). > > console = > debug,dtmf,error,event,notice,verbose,warning > syslog.local0 = > debug,dtmf,error,event,notice,verbose,warning > > Please remember to change them back and reload when you have identified > your problem(s). > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- 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 > -- Abdullah -- _ -- 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] How to pick a codec on the fly
> From: da...@debsinc.com
> To: dan...@tryba.nl; asterisk-users@lists.digium.com
> Date: Mon, 27 Sep 2010 13:30:08 -0500
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
> Sent: Monday, September 27, 2010 1:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote:
> > I'm trying to test an IVR system with recorded prompts and would
> > like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
> > ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and
#3
> > is slin; Need it the other way so I can do DAHDI--> IAX testing.
>
> exten => 1234,1,Set(_SIP_CODEC=alaw)
> exten => 1234,n,Goto(0234,1)
> exten => 2234,1,Set(_SIP_CODEC=slin)
> exten => 2234,n,Goto(0234,1)
>
> Should do the trick.
>
> --
>
> Daniel Tryba
>
> Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X.
> -- Executing [...@from-pstn:7] Goto("DAHDI/1-1", "default|s|1") in new stack
> -- Goto (default,s,1)
> -- Executing [...@default:1] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@default:2] Goto("DAHDI/1-1", "select-func|s|1") in new
> stack
> -- Goto (select-func,s,1)
> -- Executing [...@select-func:1] WaitExten("DAHDI/1-1", "5|m") in new
> stack
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> == CDR updated on DAHDI/1-1
> -- Executing [...@select-func:1] Set("DAHDI/1-1", "_SIP_CODEC=ulaw") in
> new stack
> -- Executing [...@select-func:2] Dial("DAHDI/1-1", "IAX2/xxx/332|30|m") in
> new stack
> -- Called xxx/332
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Call accepted by XXX.XXX.XX.XX (format gsm)
> -- Format for call is gsm
> -- IAX2/ffb-18075 answered DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> -- Hungup 'IAX2/xxx-18075'
> == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
>
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah
Sent: Monday, September 27, 2010 1:40 PM
To: Asterisk Users
Subject: Re: [asterisk-users] How to pick a codec on the fly
I think it's SIP_CODEC now .. and not _SIP_CODEC?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
FWIW, SIP_CODEC is value for use in Asterisk 1, _SIP_CODEC passes the value
on to Asterisk 2.
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Re: [asterisk-users] How to pick a codec on the fly
i think it's SIP_CODEC now .. and not _SIP_CODEC?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: da...@debsinc.com
> To: dan...@tryba.nl; asterisk-users@lists.digium.com
> Date: Mon, 27 Sep 2010 13:30:08 -0500
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
> Sent: Monday, September 27, 2010 1:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to pick a codec on the fly
>
> On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote:
> > I'm trying to test an IVR system with recorded prompts and would
> > like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
> > ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3
> > is slin; Need it the other way so I can do DAHDI--> IAX testing.
>
> exten => 1234,1,Set(_SIP_CODEC=alaw)
> exten => 1234,n,Goto(0234,1)
> exten => 2234,1,Set(_SIP_CODEC=slin)
> exten => 2234,n,Goto(0234,1)
>
> Should do the trick.
>
> --
>
> Daniel Tryba
>
> Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X.
> -- Executing [...@from-pstn:7] Goto("DAHDI/1-1", "default|s|1") in new stack
> -- Goto (default,s,1)
> -- Executing [...@default:1] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@default:2] Goto("DAHDI/1-1", "select-func|s|1") in new
> stack
> -- Goto (select-func,s,1)
> -- Executing [...@select-func:1] WaitExten("DAHDI/1-1", "5|m") in new
> stack
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> == CDR updated on DAHDI/1-1
> -- Executing [...@select-func:1] Set("DAHDI/1-1", "_SIP_CODEC=ulaw") in
> new stack
> -- Executing [...@select-func:2] Dial("DAHDI/1-1", "IAX2/xxx/332|30|m") in
> new stack
> -- Called xxx/332
> -- Started music on hold, class 'default', on DAHDI/1-1
> -- Call accepted by XXX.XXX.XX.XX (format gsm)
> -- Format for call is gsm
> -- IAX2/ffb-18075 answered DAHDI/1-1
> -- Stopped music on hold on DAHDI/1-1
> -- Hungup 'IAX2/xxx-18075'
> == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Asterisk Redundancy
Has any of you tested Vyatta Load balancing and fail over solution with Asterisk? It uses heartbeat and works like magic with regular traffic but didn't have the time nor chance to test it with VoIP traffic.. but I think it's the same way. Anyone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen Sent: Monday, September 27, 2010 5:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Redundancy Michelle Dupuis writes: > Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com > (also on the voip wiki) > > You get the cluster/heartbeat & replication without needing to add openSER > or full HAlinux. A simpler approach - easier to config and manage How do you handle replicating voice mails? /Benny -- _ -- 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
Re: [asterisk-users] How to pick a codec on the fly
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Monday, September 27, 2010 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to pick a codec on the fly
On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote:
> I'm trying to test an IVR system with recorded prompts and would
> like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
> ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3
> is slin; Need it the other way so I can do DAHDI--> IAX testing.
exten => 1234,1,Set(_SIP_CODEC=alaw)
exten => 1234,n,Goto(0234,1)
exten => 2234,1,Set(_SIP_CODEC=slin)
exten => 2234,n,Goto(0234,1)
Should do the trick.
--
Daniel Tryba
Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X.
-- Executing [...@from-pstn:7] Goto("DAHDI/1-1", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing [...@default:1] Answer("DAHDI/1-1", "") in new stack
-- Executing [...@default:2] Goto("DAHDI/1-1", "select-func|s|1") in new
stack
-- Goto (select-func,s,1)
-- Executing [...@select-func:1] WaitExten("DAHDI/1-1", "5|m") in new
stack
-- Started music on hold, class 'default', on DAHDI/1-1
-- Stopped music on hold on DAHDI/1-1
== CDR updated on DAHDI/1-1
-- Executing [...@select-func:1] Set("DAHDI/1-1", "_SIP_CODEC=ulaw") in
new stack
-- Executing [...@select-func:2] Dial("DAHDI/1-1", "IAX2/xxx/332|30|m") in
new stack
-- Called xxx/332
-- Started music on hold, class 'default', on DAHDI/1-1
-- Call accepted by XXX.XXX.XX.XX (format gsm)
-- Format for call is gsm
-- IAX2/ffb-18075 answered DAHDI/1-1
-- Stopped music on hold on DAHDI/1-1
-- Hungup 'IAX2/xxx-18075'
== Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
--
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Re: [asterisk-users] How to pick a codec on the fly
On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote: > I'm trying to test an IVR system with recorded prompts and would > like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 > ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 > is slin; Need it the other way so I can do DAHDI--> IAX testing. exten => 1234,1,Set(_SIP_CODEC=alaw) exten => 1234,n,Goto(0234,1) exten => 2234,1,Set(_SIP_CODEC=slin) exten => 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba -- _ -- 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] How to pick a codec on the fly
Hi list, I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI--> IAX testing. Any ideas? Google wasn't really helpful on this one. Danny Nicholas -- _ -- 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] SCCP (skinny) phone behind NAT: RTP dest addr wrong
Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base directories) working: cisco 7920 on wireless LAN behind NAT (172.16.5.0/24) rtp problem: cisco 7920 on public access point behind NAT symptom: asterisk is sending rtp to the NAT address of the phone, e.g. 192.168.0.3 in skinny.conf, nat=yes is set for the context block of the phone. tcpdump of port 2000 reveals one packet after placing a call from the phone that includes the NAT ip address of the phone (c0a8 0003): 12:14:18.812960 IP (tos 0x0, ttl 50, id 16034, offset 0, flags [none], proto TCP (6), length 72) 71.215.193.161.1025 > 216.251.177.106.2000: Flags [P.], cksum 0x461d (correct), seq 148:180, ack 569, win 16384, length 32 0x: 4500 0048 3ea2 3206 b62f 47d7 c1a1 E..H>¢..2.¶/G×Á¡ 0x0010: d8fb b16a 0401 07d0 7c8a ecc6 710d 8d62 Øû±j...Ð|.ìÆq..b 0x0020: 5018 4000 461d 1800 p...@.f... 0x0030: 2200 c0a8 0003 ba71 "...À¨..ºq.. 0x0040: 6d00 m... packet tracing RTP at the border router shows (on the LAN segment of the asterisk host), e.g. 11:43:54.380567 IP (tos 0x0, ttl 64, id 737, offset 0, flags [DF], proto UDP (17), length 200) pbx1.cybertheque.net.20002 > 192.168.0.3.29112: UDP, length 172 Shouldn't 'net=yes' in skinny.conf force asterisk to use the public IP of the phone? Help much appreciated! Michael -- _ -- 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] Asterisk 1.62.13 - CPU spikes every 10 minutes
Thanks Dave. I'll do a complete system restart when I can, and see if it helps. (an Asterisk restart didn't). > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Dave Platt > Sent: Monday, September 27, 2010 13:35 > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 > minutes > > For what it's worth... I also observed an unusually high > level of CPU activity on my small (home-system) Asterisk > installation, shortly after I switched to 1.6.2.13. > > The pattern in my case was (as best as I could tell) a > CPU load of 2% to 10%, pulsing upwards every few seconds. > I couldn't resolve it any more finely than that, at the > time I saw it. > > This is a very small Asterisk install (just two or three > SIP extensions registered at the time, no calls in progress, > no significant activity noted on the net interfaces at > the time), running on an Atom N270 CPU system. I am using > the "internal_timing = yes" option (pthreads timing rather > than dahdi). > > I did a "core restart when convenient", Asterisk restarted, > and the CPU load dropped to negligible levels. It has > remained low ever since (about 7 minutes of CPU used, over > the space of a couple of days). > > I have a hunch that in 1.6.2.13, some resource isn't being > cleaned up properly after calls terminate or extensions > de-register, and that the system is spending a significant > amount of time "chasing its tail" navigating through these > obsolete resources. Just a hunch, though... nothing to back > this up. > > > > -- > _ > -- 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
Re: [asterisk-users] Asterisk Redundancy
On Sep 27, 2010, at 1:29 PM, Michelle Dupuis wrote: > That's exactly what we recommend for DB/realtime installs. HAAST's focus is > the failover, promotion, assignment of IP, etc. but links to standard tools > for file/db sync. In line with the philosophy of try to not be everything to > everybody... > > From: asterisk-users-boun...@lists.digium.com > [asterisk-users-boun...@lists.digium.com] On Behalf Of Vahan Yerkanian > [va...@arminco.com] > Sent: Monday, September 27, 2010 1:02 PM > To: Asterisk Users List > Subject: Re: [asterisk-users] Asterisk Redundancy > > On 9/27/10 8:57 PM, Michelle Dupuis wrote: >> HAAST runs a sync script a regular intervals (time to sync data prior to a >> failover check etc) >> >> HAAST includes a sample script which syncs voicemail (and config, etc) files >> using rsync from master to slave. After a master/slave reversal the process >> automatically reverses. >> >> MD > What about ODBC/IMAP voicemail storage? Works great with MySQL > Master<>Master replication for me. > > Vahan Thousand ways of scaling the redundancy mountain here... Big questions are if this is geographic redundancy, how many nodes, etc. For simple 2 box redundancy on a lan, I choose DBRB with HeartBeat... ---fred http://qxork.com -- _ -- 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] Asterisk 1.62.13 - CPU spikes every 10 minutes
For what it's worth... I also observed an unusually high level of CPU activity on my small (home-system) Asterisk installation, shortly after I switched to 1.6.2.13. The pattern in my case was (as best as I could tell) a CPU load of 2% to 10%, pulsing upwards every few seconds. I couldn't resolve it any more finely than that, at the time I saw it. This is a very small Asterisk install (just two or three SIP extensions registered at the time, no calls in progress, no significant activity noted on the net interfaces at the time), running on an Atom N270 CPU system. I am using the "internal_timing = yes" option (pthreads timing rather than dahdi). I did a "core restart when convenient", Asterisk restarted, and the CPU load dropped to negligible levels. It has remained low ever since (about 7 minutes of CPU used, over the space of a couple of days). I have a hunch that in 1.6.2.13, some resource isn't being cleaned up properly after calls terminate or extensions de-register, and that the system is spending a significant amount of time "chasing its tail" navigating through these obsolete resources. Just a hunch, though... nothing to back this up. -- _ -- 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] Asterisk Redundancy
That's exactly what we recommend for DB/realtime installs. HAAST's focus is the failover, promotion, assignment of IP, etc. but links to standard tools for file/db sync. In line with the philosophy of try to not be everything to everybody... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Vahan Yerkanian [va...@arminco.com] Sent: Monday, September 27, 2010 1:02 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk Redundancy On 9/27/10 8:57 PM, Michelle Dupuis wrote: > HAAST runs a sync script a regular intervals (time to sync data prior to a > failover check etc) > > HAAST includes a sample script which syncs voicemail (and config, etc) files > using rsync from master to slave. After a master/slave reversal the process > automatically reverses. > > MD What about ODBC/IMAP voicemail storage? Works great with MySQL Master<>Master replication for me. Vahan -- _ -- 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
Re: [asterisk-users] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias wrote: > The same problem! > What is the output from the following? $ ls -la /lib/modules/ $ ls -la /usr/src/linux -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Problems compiling Asterisk on Debian
The same problem! r...@sangoma-testing:/usr/src/dahdi-linux-complete-2.4.0+2.4.0# make all && make install && make config make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/firmware' Attempting to download dahdi-fwload-vpmadt032-1.25.0.tar.gz --2010-09-27 13:07:26-- http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fwload-vpmadt032-1.25.0.tar.gz Resolving downloads.digium.com... 76.164.171.232, 2001:470:e0d4::e8 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 149360 (146K) [application/x-gzip] Saving to: `dahdi-fwload-vpmadt032-1.25.0.tar.gz' 100%[==>] 149,360 81.0K/s in 1.8s 2010-09-27 13:07:28 (81.0 KB/s) - `dahdi-fwload-vpmadt032-1.25.0.tar.gz' saved [149360/149360] make[2]: Leaving directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux' make: *** [all] Error 2 2010/9/27 Paul Belanger > On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias > wrote: > > What should i do? > > > Try with the lastest DAHDI version, 2.4.0. > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- 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
Re: [asterisk-users] Asterisk Redundancy
On 9/27/10 8:57 PM, Michelle Dupuis wrote: > HAAST runs a sync script a regular intervals (time to sync data prior to a > failover check etc) > > HAAST includes a sample script which syncs voicemail (and config, etc) files > using rsync from master to slave. After a master/slave reversal the process > automatically reverses. > > MD What about ODBC/IMAP voicemail storage? Works great with MySQL Master<>Master replication for me. Vahan -- _ -- 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] Problems compiling Asterisk on Debian
I've these versions of DAHDI running into another Server with no problem...it seems to be a problem with dependencies, but i can't find the trick :( 2010/9/27 Paul Belanger > On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias > wrote: > > What should i do? > > > Try with the lastest DAHDI version, 2.4.0. > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- 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
Re: [asterisk-users] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 06:28:04PM +0200, Danny Dias wrote: > You do not appear to have the sources for the 2.6.26-2-amd64 kernel > installed. > exit 1 > make: *** [modules] Error 1 > > > The same result :( Where did you get the dahdi source? There is no "official" 2.1.0.4 in any debian version. If you are running stable you should either use the backports version: http://packages.debian.org/lenny-backports/dahdi Or make your own package from testing/unstable sources. -- Daniel Tryba -- _ -- 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] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias wrote: > What should i do? > Try with the lastest DAHDI version, 2.4.0. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Problems compiling Asterisk on Debian
¿ahhh? 2010/9/27 Roger Burton West > On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: > > >What should i do? > > aptitude install module-assistant > m-a a-i dahdi > > > -- > _ > -- 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
Re: [asterisk-users] propagate sip reinvites with directrtpsetup=yes
ok, thanks. i was beginning to suspect as much but was hoping to limit the number of components in our configuration. thanks, gene On Mon, Sep 27, 2010 at 11:13 AM, Kevin P. Fleming wrote: > On 09/27/2010 11:02 AM, Eugene Oden wrote: >> is there a trick to get asterisk (1.6.2.13) to propagate >> codec-changing sip reinvites when directrtpsetup=yes? >> >> i'm trying to route calls to a gateway without keeping asterisk in the >> rtp stream. > > You are looking for a SIP proxy; Asterisk is not a SIP proxy, and no > amount of configuration will convince it to act like one. > >> the gateway is first routing the call to a media server. when >> connecting the call to the downstream carrier a different codec is >> selected. >> >> the reinvite makes it to asterisk but asterisk isn't sending it along >> to the originator so the transmit/receive codecs are mismatched >> causing one-way audio. > > Asterisk never "sends along" re-INVITEs, because the two channels > involved in an Asterisk "call" are distinct. If the codecs are > mismatched between the two call legs, Asterisk will try to transcode them. > > The 'directrtpsetup' feature is still marked *experimental*, and that is > primarily because it defeats much of Asterisk's normal behavior; in > addition, there a quite a few normal, working call scenarios for which > it will fail... so it's there, but if you use it, you can expect > difficulties. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kflem...@digium.com > Check us out at www.digium.com & www.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
Re: [asterisk-users] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: >What should i do? aptitude install module-assistant m-a a-i dahdi -- _ -- 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] Problems compiling Asterisk on Debian
Done my friend: r...@sangoma-testing:/lib/modules/2.6.26-2-amd64/kernel/drivers# module-assistant prepare Getting source for kernel version: 2.6.26-2-amd64 apt-get install linux-headers-2.6.26-2-amd64 Reading package lists... Done Building dependency tree Reading state information... Done linux-headers-2.6.26-2-amd64 is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. Creating symlink... apt-get install build-essential Reading package lists... Done Building dependency tree Reading state information... Done build-essential is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. Done! r...@sangoma-testing:/lib/modules/2.6.26-2-amd64/kernel/drivers# cd /usr/src/dahdi-linux-2.1.0.4/ r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 The same result :( 2010/9/27 Daniel Tryba > On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: > > r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make > > echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel > > installed." > > You do not appear to have the sources for the 2.6.26-2-amd64 kernel > > installed. > > exit 1 > > make: *** [modules] Error 1 > > > > What should i do? > > The easiest way would be to use module-assistant > # aptitude install module-assistant > # module-assistant prepare > > This checks which kernel you are running and install the right packages, > eg: > module-assistant prepare > Getting source for kernel version: 2.6.26-2-vserver-amd64 > apt-get install linux-headers-2.6.26-2-vserver-amd64 > Reading package lists... Done > Building dependency tree > Reading state information... Done > The following extra packages will be installed: > linux-headers-2.6.26-2-common-vserver linux-kbuild-2.6.26 > The following NEW packages will be installed: > linux-headers-2.6.26-2-common-vserver > linux-headers-2.6.26-2-vserver-amd64 > linux-kbuild-2.6.26 > 0 upgraded, 3 newly installed, 0 to remove and 0 not upgraded. > Need to get 4366kB of archives. > After this operation, 35.8MB of additional disk space will be > used. > Do you want to continue [Y/n]? > > -- > > Daniel Tryba > > -- > _ > -- 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
Re: [asterisk-users] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: > r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make > echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel > installed." > You do not appear to have the sources for the 2.6.26-2-amd64 kernel > installed. > exit 1 > make: *** [modules] Error 1 > > What should i do? The easiest way would be to use module-assistant # aptitude install module-assistant # module-assistant prepare This checks which kernel you are running and install the right packages, eg: module-assistant prepare Getting source for kernel version: 2.6.26-2-vserver-amd64 apt-get install linux-headers-2.6.26-2-vserver-amd64 Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: linux-headers-2.6.26-2-common-vserver linux-kbuild-2.6.26 The following NEW packages will be installed: linux-headers-2.6.26-2-common-vserver linux-headers-2.6.26-2-vserver-amd64 linux-kbuild-2.6.26 0 upgraded, 3 newly installed, 0 to remove and 0 not upgraded. Need to get 4366kB of archives. After this operation, 35.8MB of additional disk space will be used. Do you want to continue [Y/n]? -- Daniel Tryba -- _ -- 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] Problems compiling Asterisk on Debian
Thanks Dean, I've done it before, that's why i'm here asking :( take a look: r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# apt-cache search linux-headers-$(uname -r) linux-headers-2.6.26-2-amd64 - Header files for Linux 2.6.26-2-amd64 r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# sudo apt-get install linux-headers-$(uname -r) Reading package lists... Done Building dependency tree Reading state information... Done linux-headers-2.6.26-2-amd64 is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 2010/9/27 Dean Hoover > > Source files aren't automatically installed on every install. > > This link should help: > http://www.cyberciti.biz/faq/howto-install-kernel-headers-package/ > > Dean Hoover > Milwaukee, Wisconsin > > On 9/27/2010 11:09 AM, Danny Dias wrote: > > Hello, > > > > I'm trying to compile DAHDI on DEBIAN but i have the following error: > > > > r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make > > echo "You do not appear to have the sources for the 2.6.26-2-amd64 > > kernel installed." > > You do not appear to have the sources for the 2.6.26-2-amd64 kernel > > installed. > > exit 1 > > make: *** [modules] Error 1 > > > > What should i do? > > > > Thanks! > > > > > -- > _ > -- 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
Re: [asterisk-users] Problems compiling Asterisk on Debian
Source files aren't automatically installed on every install. This link should help: http://www.cyberciti.biz/faq/howto-install-kernel-headers-package/ Dean Hoover Milwaukee, Wisconsin On 9/27/2010 11:09 AM, Danny Dias wrote: > Hello, > > I'm trying to compile DAHDI on DEBIAN but i have the following error: > > r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make > echo "You do not appear to have the sources for the 2.6.26-2-amd64 > kernel installed." > You do not appear to have the sources for the 2.6.26-2-amd64 kernel > installed. > exit 1 > make: *** [modules] Error 1 > > What should i do? > > Thanks! > -- _ -- 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] propagate sip reinvites with directrtpsetup=yes
On 09/27/2010 11:02 AM, Eugene Oden wrote: > is there a trick to get asterisk (1.6.2.13) to propagate > codec-changing sip reinvites when directrtpsetup=yes? > > i'm trying to route calls to a gateway without keeping asterisk in the > rtp stream. You are looking for a SIP proxy; Asterisk is not a SIP proxy, and no amount of configuration will convince it to act like one. > the gateway is first routing the call to a media server. when > connecting the call to the downstream carrier a different codec is > selected. > > the reinvite makes it to asterisk but asterisk isn't sending it along > to the originator so the transmit/receive codecs are mismatched > causing one-way audio. Asterisk never "sends along" re-INVITEs, because the two channels involved in an Asterisk "call" are distinct. If the codecs are mismatched between the two call legs, Asterisk will try to transcode them. The 'directrtpsetup' feature is still marked *experimental*, and that is primarily because it defeats much of Asterisk's normal behavior; in addition, there a quite a few normal, working call scenarios for which it will fail... so it's there, but if you use it, you can expect difficulties. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.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
[asterisk-users] Problems compiling Asterisk on Debian
Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? Thanks! -- _ -- 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] propagate sip reinvites with directrtpsetup=yes
is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't sending it along to the originator so the transmit/receive codecs are mismatched causing one-way audio. thanks, gene -- _ -- 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] Asterisk Redundancy
HAAST runs a sync script a regular intervals (time to sync data prior to a failover check etc) HAAST includes a sample script which syncs voicemail (and config, etc) files using rsync from master to slave. After a master/slave reversal the process automatically reverses. MD From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen [benny+use...@amorsen.dk] Sent: Monday, September 27, 2010 10:35 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk Redundancy Michelle Dupuis writes: > Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com > (also on the voip wiki) > > You get the cluster/heartbeat & replication without needing to add openSER > or full HAlinux. A simpler approach - easier to config and manage How do you handle replicating voice mails? /Benny -- _ -- 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
Re: [asterisk-users] Asterisk as a distributed paging system
Hi! thank you for your good answers. Another related question: I tried using Page() and it works perfectly, but I need to implement a slightly different behavior, and I'm looking into ways of implementing it. When a user picks up the phone and chooses to page the speakers, the call should start (so that it's ready for talking), but in muted status. When the user pushes a push-to-talk button, then a bell sound needs to be played through all the speakers, then she can start talking freely. Everytime the PTT button is released, the mic needs to mute, but that's something I can work out in the softphone. How can I implement it? I am thinking of using some call parking method and some DTMF code to pass to the next state, but I am open to advice, since I'm quite new to Asterisk. Could I also create a macro to do the same thing Page is doing, but with ConfBridge? Last question: is there a way of reinviting periodically remotes to the conference, so that they can recover after e.g. a reboot? Thank you in advance, Matteo Il 22/09/2010 21:51, Philipp von Klitzing ha scritto: > Hi! > > >> I need the system to be resilient to any network partition, so that >> anyone can send announces from any mic to all the reachable clients. >> I'd need also to page a subset of all the speakers. >> > Most of the major phone vendors (that are employed by the users of this > list) have support for multi-cast of some sort. In recent firmware > release notes I have read that SNOM has now also added a feature to feed > multicast directly from a phone (and not just play multicast audio on the > speaker as long as the phone is not in use). > > >> I'm currently using some software I wrote which sends voice over >> multicast RTP and coordinates all the sites with multicast messages. >> > app_page has been around for quite some in Asterisk, and the new Asterisk > 1.8 now also adds the channel driver "MulticastRTP". > > >> Is there a way asterisk could be of use, or would I need to bend it >> too much? >> > Look here: > http://www.voip-info.org/wiki/view/Asterisk+cmd+Page > http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom > > I have made good experience with MAST for multicasting SNOM phones: > http://www.aelius.com/njh/mast/ > > Philipp > > > -- _ -- 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] groupcount - show usage
hi, i'm using groupcount to limit max calls in pbx i want show/graph made calls usage it's possible make this from cli/ami? something like asterisk>group show usage name channels group1 3 group2 5 google doesnt help thanks --- Marek Cervenka === -- _ -- 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] Record() Cmd and My SQL
On Sun, Sep 26, 2010 at 10:49 PM, Govind, Mahesh (NSN - IN/Bangalore) wrote: > Another reason for storing in the database is to , enable some other > apps to access the recording at some point of time . Yeah, still not a good reason. File systems allow multiple read streams to the same file. Perhaps you should consider a webservice with a squid proxy in front of it if you want to have hundreds of simultaneous accesses. Just store a recording as a file, and if you have metadata, store the metadata in a database. -- _ -- 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] RFC3329 support in Asterisk
On Mon, Sep 27, 2010 at 6:37 AM, sijan ahamed wrote: > Is RFC3329(Security Mechanism Agreenment for SIP) supported in Asterisk ? > No, not at this point in time. http://svn.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h However, patches welcome. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...
bruce bruce writes: > Other than the price difference (2.5" is more expensive and can't find > many of the 1TB or so) is there any preference, advantage, or > disadvatage of chosing 2.5" HDD or 3.5" when it comes to the server > operations or Asterisk operation? There is no difference. Pick the server which offers the disk bandwidth and I/O's per second which you need. Do you really need 1TB disks? If you do, be careful what you place on those disks. Reading e.g. a voice mail or a speak off a large slow platter which is busy writing CDR's does not sound good at all. /Benny -- _ -- 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] Asterisk Redundancy
Michelle Dupuis writes: > Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com > (also on the voip wiki) > > You get the cluster/heartbeat & replication without needing to add openSER > or full HAlinux. A simpler approach - easier to config and manage How do you handle replicating voice mails? /Benny -- _ -- 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 and dahdi on Arch linux
Hi all, Has anyone tried Asterisk on Arch? I am currently running the latest Arch and I am thinking about installing Asterisk or is some other distro better? I have been using Debian in the passed. Many thanks for any reply! Christian-- _ -- 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] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...
On Mon, 27 Sep 2010 15:00:36 +0200, Roberto Piola wrote: >> >> Hard drive speed may differ between 5400rpm and 7200rpm. > >in high performance server environment, you can get 1 and 15000rpm >drives as well... the fastest, the better (and the more expensive). > >moreover, if you have 6 disks in raid 1+0, you have better write >performance than 3 disks in raid5 We use WD Velociraptor which are 2.5" 10 krpm SATA disks. They're mounted in a heatsink that makes the fit into the same mount as a 3.5" disk. Good preformance. Decent prices and capacity. The failure rate has been low. One of the issued with the fastest drives is that in order to achieve the highest spin rates they often have smaller platters and so lower capacity. Cost/GB goes up. Michael > >-- >_ >-- 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 > -- Michael Graves mgravesmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...
> > Hard drive speed may differ between 5400rpm and 7200rpm. in high performance server environment, you can get 1 and 15000rpm drives as well... the fastest, the better (and the more expensive). moreover, if you have 6 disks in raid 1+0, you have better write performance than 3 disks in raid5 -- _ -- 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] func SHARED, how to use?
Hi! > Well, only problem I see, is to how pass channel name from macro to h > extension... SHARED() or CDR(userfield) Philipp -- _ -- 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] A2billing
Hi, Thank you !! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda > To: asterisk-users@lists.digium.com > From: hvarda...@gmail.com > Date: Mon, 27 Sep 2010 15:47:31 +0500 > Subject: Re: [asterisk-users] A2billing > > Hello > > Change your a2b 1.8 version to a2b 1.7.1 version > > 1.7.1 version is a stable. > And forum for a2b is http://forum.asterisk2billing.org/index.php > > > > -- > Vardan Harutyunyan, > Senior System Administrator > > Enterprise Incubator Foundation > 123 Hovsep Emin Street, > Yerevan 0051, Republic of Armenia > Tel: + 374 10 219735 > Fax: + 374 10 219777 > E-mail: i...@eif.am > www.eif-it.com > > Flavio Miranda wrote: > > > > Hi, > > > > I am trying to configure a2billing 1.8 in my asterisk 1.6 but no value > > to DIALPREFIX and DESTINATION PREFIX is accepted when I try to create a > > RATE. > > > > thanks! > > > > > > Att, > > > > Flavio Roberto Miranda > > MSN:flaviormira...@hotmail.com > > Skype: flaviormiranda > > > > > -- > _ > -- 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
Re: [asterisk-users] Debug compile fails
On Fri, Sep 24, 2010 at 02:30:12PM -0400, Paul Belanger wrote: > > Am I missing something? > > > DEBUG_THREADS Thanks, I guess I should have RTFM :) -- Daniel Tryba -- _ -- 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] A2billing
Hello Change your a2b 1.8 version to a2b 1.7.1 version 1.7.1 version is a stable. And forum for a2b is http://forum.asterisk2billing.org/index.php -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Flavio Miranda wrote: > > Hi, > > I am trying to configure a2billing 1.8 in my asterisk 1.6 but no value > to DIALPREFIX and DESTINATION PREFIX is accepted when I try to create a > RATE. > > thanks! > > > Att, > > Flavio Roberto Miranda > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > -- _ -- 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] RFC3329 support in Asterisk
Hello, Is RFC3329(Security Mechanism Agreenment for SIP) supported in Asterisk ? Regards, Sijan -- _ -- 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] PSTN to SMS and SMS to PSTN
Dear All, As per this article http://www.voip-info.org/wiki/view/Asterisk+cmd+Smsasterisk support PSTN to SMS and from SMS support endpoints to PSTN. In my scenario we have SS7 based E'1 on which our SMS provider sending SMS on our DID numbers and my all DID's are registered on OPENSER. What I want to do I want to receive SMS from PSTN on E’1 and forward on Register user’s if Register Endpoint is SMS supported or Instant message supported then message should deliver on it or reject and vice versa if Endpoint generate SMS or instant message if should be forward on PSTN and related to own DID number’s message should not forward on PSTN if should forward internal. First I want to confirm it is possible or not, and second I want to do as project interested people please contact me on miana...@msn.com thanks. -- Regards, M. Asif Raza -- _ -- 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
