Hello, I have noticed when ser forwards to asterisk, the last "registered" host from ser is always the subsequent callee whichever client dials. i.e. 4561 registers 4562 registers 4563 registers 4562 calls 4561. Asterisk shows 4563 dialing 4561. I am forwarding registrations and invites to asterisk. Is this correct? Many thanks, Spencer
I read about SER and I have installed Asterisk on one machine. Unfortunately this machine is broken and the replacement will take longer as I wished. I have a P-4, 1GHz with 1 GB RAM, which was planned to be used as SER anyway. I would like to use this downtime of the Asterisk box to install SER. I understand that SER and Asterisk can work together, while SER makes connection, Asterisk will add features. How do I do that efficiently? 1. SER needs a public IP, does then Asterisk need a pulic IP, or can it just be on the internal LAN? 2. Now the password for each phone is made in the SIP and IAX config files. How can I do that than in SER? If I use reatime, can I use the same database? 3. A user can make a phone call to the users on our system (Asterisk domain) and via gateways. We use ASTCC for billing. How can I use this together with SER. I am not sure if I understand it correct, but would for internal calls (calls without features) be handeled directly from SER, while if the user need a feature, than he would be transfered to Asterisk? How can I setup multiple Asterisk boxes with one SER, but all Asterisk machines share the same dial plan (realtime) I wanted to read more first, but the down time of my * forces me to do a SER setup faster, so that my users can quickly (by office hours Monday) at least make internal calls. Any hint is welcome ;-) bye Ronald
I have on one machine Openser and Asterisk. Since Asterisk was first, I let it have the port 5060 ;-) I have choosen for Openser the port 5062. I tried several hard and soft phones to connect to ser to the port 5062, however each of the phones tries to connect to asterisk. I am totally confused about that, what could redirect all requests to port 5060. (I could not get any answer from ser nor openser mailing list, maybe I am lucky with a hint here) bye Ronald Wiplinger
Oleh Mukha
2005-Oct-19 02:28 UTC
[Asterisk-Users] Problem with select correct network interface (oh323)
i build asterisk on pc with 3 network inerface eth0 (yyy.yyy.yyy.yyy) main public ip eth1 (xxx.xxx.xxx.xxx) seconf public ip used only for voip connection eth2 (zzz.zzz.zzz.zzz) local ip i config oh323 to bind eth1 interface i try make call from my local network -> Asterisk -> provider h323 when i try to call from ata 186 throught my astersik oh323 module asetrisk resive calls from ata but send it to my oh323 providet not from eth1 (with ip xxx.xxx.xxx.xxx) or from eth0 (ip yyy.yyy.yyy.yyy) how can i tel asterisk send data from me to my provider from eth1 (ip xxx.xxx.xxx.xxx) Oleh Mukha IClub 380322722738 www.ic.lviv.ua
On 10/19/05, Yair Hakak <yhakak@gmail.com> wrote:> > i do it this way because i want all the dialplan logic and CDR having to > do with PSTN in asterisk, not SER. > so, calls from the outside are adressed to sipaddress@myserver:5070 and > hit asterisk. asterisk either sends them along to 5060, or handles them > internally (IVR, voicemail, etc) based on the dialplan. > clients on the inside are registered to the SER at 5060 and the SER > automatically forwards them to asterisk. if they are PSTN asterisk serves as > PSTN gateway, if they are internal, asterisk native bridges and drops out, > but still keeps the CDR (i have full SIP addresses in my dial statements > instead of asterisk SIP peers) > the reason i do this is i found that if the endpoints are scattered on the > internet, SER+rtpproxy is much more stable than asterisk as a SIP server > (asterisk kept dropping endpoints). This way SER serves as a completely > "dumb" SIP server, and just sends everything along. there is a minimal > increase in overhead (i could handle internal calls just with SER) but it's > worth it to have all the dialplan logic and CDR's in one place. > also, obviously, if i use an IAX provider for outgoing, asterisk has to > be in the middle. > i agree though, it makes more sense to have SER on 5060 and asterisk > somewhere else. > hope i'm making some sense, please point out if i'm doing something > really stupid. > > -yair > On 10/19/05, trixter aka Bret McDanel <trixter@0xdecafbad.com> wrote: > > > On Wed, 2005-10-19 at 10:55 +0200, Yair Hakak wrote: > > > hello, > > > trace the SIP packets and see if they are actually addressed to 5062. > > > if you post the ngrep or ethereal dump we'll see whats actually going > > > on. I do this with SER on 5060 and asterisk on 5070 and there are no > > > problems - my extensions point to 5060 and my DID's point to 5070 so > > > asterisk serves as the gateway to the PSTN. > > > > > > -yair > > > > > > > > also look for dns packets and see if htey are pulling the server info. > > Some sip clients look for specific server type dns records to see where > > they should go. > > > > 5060 is the default, wouldnt it make more sense to have the default port > > be what you want the devices to goto and have that proxy to the device > > you dont want direct connectivity to? Or am I missing something in that > > > > > > > -- > > Trixter http://www.0xdecafbad.com Bret McDanel > > UK +44 870 340 4605 Germany +49 801 777 555 3402 > > US +1 360 207 0479 or +1 516 687 5200 > > FreeWorldDialup: 635378 > > > > > > -----BEGIN PGP SIGNATURE----- > > Version: GnuPG v1.4.1 (GNU/Linux) > > > > iD8DBQBDVg5f+1olxlzQw5cRAhl5AJ91lwjqMb2EPcDSXH69dOELBOq0IQCgvr8m > > 4NqQAGLmWLokUXjl7Bi7SbI> > =thAz > > -----END PGP SIGNATURE----- > > > > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051019/2602326b/attachment.htm
Moises Silva
2005-Oct-19 07:00 UTC
[Asterisk-Users] Problem with select correct network interface (oh323)
it seems to me that your problem is not Asterisk configuration, but iproute configuration. Look in google about iproute and kernel routing tables. In order to help you, it would be desireable to know how are you dialing. Best Regards. On 10/19/05, Oleh Mukha <fly@ic.lviv.ua> wrote:> > i build asterisk on pc with 3 network inerface > eth0 (yyy.yyy.yyy.yyy) main public ip > eth1 (xxx.xxx.xxx.xxx) seconf public ip used only for voip connection > eth2 (zzz.zzz.zzz.zzz) local ip > i config oh323 to bind eth1 interface > i try make call > from my local network -> Asterisk -> provider h323 > > when i try to call from ata 186 throught my astersik oh323 module > asetrisk resive calls from ata but send it to my oh323 providet not from > eth1 > (with ip xxx.xxx.xxx.xxx) or from eth0 (ip yyy.yyy.yyy.yyy) > > how can i tel asterisk send data from me to my provider from eth1 (ip > xxx.xxx.xxx.xxx) > > > Oleh Mukha > IClub > 380322722738 > www.ic.lviv.ua <http://www.ic.lviv.ua> > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com <http://Easynews.com>-- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051019/bacebe92/attachment.htm