similar to: Multiple IPs and SIP

By reading the Wiki's I found out that an Asterisk server with many (>10000) extensions and/or SIP users can become slow when reloading. But what happens when you also have many contexts in extensions.conf? More precisely, one context for each SIP user? I need this because I will have users with random usernames that they can choose, but I obviously cannot set that username as the outgoing

We are planning to deploy a pretty large asterisk server with many SIP extensions (might be up to 10000 in the future), and I have a few questions:   1) is this possible, or are we running into some kind of limitation in the software that I wasn't aware of and that I didn't find by browsing through the archives and through Wiki? No, we don't need any G729-G711 transformations, it would

That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had choppy audio. After disabling HT, everything was fine again. Nothing measurable, indeed, but you could definitely hear it. So there *must* be something. -Manuel -----Messaggio originale----- Da: Peter Corlett

> thanks, using your example, and this url: > http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0 9186a00800dea82.shtml <http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note 09186a00800dea82.shtml> > I got it to work... then I realized that there's no way the SIP phone > on asterisk is going to get the MWI ( message waiting

> I want to have call forwarding (from the POTS) > turned on at the close of work and turned off > automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job Regards -Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax

I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to switch back to 3.23? # make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names

I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get that error: *CLI> load cdr_odbc.so Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Unable to load module cdr_odbc.so But the file is there... # ls -lag /usr/local/lib/libiodbc.so* lrwxrwxrwx 1 root

>> allow=ulaw >Why don't you remove this? Because I need some other users to be able to call out using ULAW over the same PSTN gateway... -Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com

>> I'm trying to compile cdr_addon_mysql but keep getting this error. >> Again, searching the Wiki and the mailing list archive didn't bring up >> anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to >> switch back to 3.23? >> >> >> # make >> cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o

I have been playing with res_odbc in these last days, but it doesn't want to work. This is the output when starting Asterisk, so everything seems OK: [res_odbc.so] => (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:132 load_odbc_config: registered database handle 'mysql' dsn->[MySQL-asterisk] Jul 7

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm

I've just setup a new asterisk server, and I need to have G.729 working on this system. The problem is I don't have any IDE drives (and therefore no /dev/hda etc), but only /dev/sda.   Is there really *no* way to license G.729 on a SCSI-only system? IMHO it's really stupid to replace an entire server because of a licensing issue. There *must* be a solution.   Anyone, please? Or at

> Is it possible to tell asterisk not to strip the leading 0 > of *incoming* MSNs? I use asterisk with i4l and whenever > I get a call from an long-distance party, the leading 0, which > should be there according the german numbering, is not. Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest

I am trying to make Asterisk communicate with a voice switch which doesn't need (and like) authentication on outgoing SIP calls. I have configured it as follows in my sip.conf:   [myswitch] type=friend host=192.168.1.100 port=5060 context=default canreinvite=no To dial out using this switch (it acts as a PSTN gateway) I use this in extensions.conf:   exten =>

If I have inbound SIP calls arriving from a provider's gateway to an asterisk server on my LAN, which then routes the call back out via the provider's gateway to a PSTN number, once the call is answered do all the voice packets pass through my asterisk PBX, or is SIP intelligent enough to patch the two PSTN ends of the call direct to each other going only via two ports on the

I have recently posted a message regarding hung SIP channels when using the Monitor() command. Well, I was wrong.   The channel hanging wasn't caused by the Monitor command. They also hang when they aren't monitored. The cause seems to be our PSTN gateway provider. When for some reason their PSTN gateway crashes or reboots (OK, this should happen, but anyway...) and RTP/SIP data stops

Hi Everyone, I have a provider who is asking me to send SIP signals through 111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2: 244.244.244.244. This provider authenticates by IP and I think is using Sonus gear and hence they have some load balancer or something... I have always simply done this to work it out: host=111.111.111.111 peer=type and everything worked. But

Hi Vincent, > In my environment, I have a total of 4 DCs (Samba 4.7.6) running in VMs. > Their uptime schedule goes like this: > dc00 : usually 100% unless there's a failure. > dc01 : same as above > dc02 : a few days per week. > dc03 : a few days per month. may I inquire why you are have setup such a scenario? If all DC are on same site, it is not necessary to have such a

I have thew same problem. Ui tried with dyn dns in the externip field in sip.conf but I think the Asterisk does not allow this. Unfortunally I have every day a new ip. Maybe I can write a script witch takes my actual ip from externat and put it into the externip field. Maybe this solves the problem. -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com