Re: [Asterisk-Users] How often do YOU register?
Hi, Being SER user I use 5 minutes (300 seconds). But you have to balance between load on your registrar server (like * in this case) and keeping your database up to date. Too short re-registration in huge system means literally tens of registration per second. To long registration means: a) users will seem available for a prolonged time, even during power off, their network failure, etc. b) if user will change IP address and you have long expiratoin time, he will be visible under two or more addresses. Most SIP proxies will perform parallel forking - they will contact all IPs. Of coure provided that you do not perform any form of registrar database 'purge' or cleaning, when registering UA from new IP address Matt wrote: Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations? -- Regards, Arek Bekiersz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] adress book
Hello Joao, I'm using SER and Asterisks-based system, with centralized LDAP backend. To access LDAP I use SOAP and DSML.This is now used for every provisioning/management/billing/ivr activity in the system. In future I plan to have centralized phonebook based in LDAP. I think that having centralized LDAP directory and accessing it from clients via SOAP/XML is a best option. If security is an issue, SOAP/XML Digital Signature and Encryption could be used here. Maybe we will live until times when hardware vendors will support SOAP clients in their phones, or at least XML browsers or some sort of thin clients. I see SNOM is doing something in XML - maybe worth checking. I think that will be the soft-phone manufacturers that will first adapt idea of central phonebook, based on SOAP message exchanges with centralized LDAP directory servers. -- Regards, Arek Bekiersz Joao Pereira wrote: Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] adress book
Yes its how I started (ldap+radius( But it depends what you want to do. 1) If you want to have nice display in softphone (or hardware phone with LCD) of global system phonebook and/or private phonebook - I'm sorry, no vendor is supporting this. I was trying to convince few videophone manufacturers to support XML or SOAP, but there are other reasons they won't do it now (they all go proprietary). 2) If you want to just implement a SIP-like phonebook functionality, like you hook-off, you press # and number of phonebook entry and system dials for you, then voila. You can do it yourself. Just write clever SER module, or PHP script and use Ldap. -- Regards, Arek Bekiersz Voipers Portugal wrote: Hi, I am using SER with centralized LDAP backend which is accessed by RADIUS. Maybe it could work out for you. Jose Simoes On 1/30/06, *Arek Bekiersz* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello Joao, I'm using SER and Asterisks-based system, with centralized LDAP backend. To access LDAP I use SOAP and DSML.This is now used for every provisioning/management/billing/ivr activity in the system. In future I plan to have centralized phonebook based in LDAP. I think that having centralized LDAP directory and accessing it from clients via SOAP/XML is a best option. If security is an issue, SOAP/XML Digital Signature and Encryption could be used here. Maybe we will live until times when hardware vendors will support SOAP clients in their phones, or at least XML browsers or some sort of thin clients. I see SNOM is doing something in XML - maybe worth checking. I think that will be the soft-phone manufacturers that will first adapt idea of central phonebook, based on SOAP message exchanges with centralized LDAP directory servers. -- Regards, Arek Bekiersz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Router
Hi, Try one of Venus 2804, 2808 or 2832 from Tainet corporation. They support SIP or MGCP and they come with VPN. http://www.tainet.net Proceed to Product/VoIP/Venus -- Regards, Arek Bekiersz Mohamed Farid wrote: Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - no outband DTMF with Mediatrix
Dear List members, I have this problem with Mediatrix 24-FXS-line gateway and out-of-band DTMF. It seems not working - the RTP mode is not working and when I select INFO mode, the Mediatrix is behaving just the same as with RTP mode. Here is a bunch of logs to explain this: 1. The RTP out-of-band mode (dtmfmode=rfc2833): This is OK reply from Asterisk to Mediatrix when RTP mode selected. Seems OK ;-): [...] SIP/2.0 200 OK CSeq: 1091919829 INVITE v=0 o=root 35059 35059 IN IP4 xxx s=session c=IN IP4 xxx t=0 0 m=audio 12210 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 [...] And then, during connection with asterisk, when we use DTMF, this shows on debug: [...] May 3 17:49:42 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received May 3 17:49:42 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received [...] 2. INFO mode (dtmfmode=info): Proper INVITE from Mediatrix: [...] INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 CSeq: 1657017135 INVITE Content-Type: application/sdp Contact: Port 3 sip:[EMAIL PROTECTED] Supported: replaces User-Agent: MxSipApp/4.4.11.68 MxSF/v3.2.7.30 v=0 o=MxSIP 4563726510189014186 6429835688411497953 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 a=sendrecv m=audio 5004 RTP/AVP 8 18 4 0 13 111 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 X-nt-inforeq/8000 [...] And then nothing happens, Asterisk shows no DTMF events. Thanks for any help, Arek Bekiersz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER - choppy sound with G.729
Hi, We are using Asterisk running on FreeBSD,as IVR / Voicemail for SER.We have redirected certain callsfrom SERto *.On * there is some 'testing' extension. It's simply playing some demo now;-) As long as I use plain G.711 the sound is nice. When I switch toG.729 the sound is choppy, not recognizable. What is going on? Debug shows everything is normal.. I understand that all jingles / sounds are recorded in gsm format. Maybe I should try to convert them to g.729, as * isnot converting it correctly on the fly? ButI don't see anyhigh-CPU-usage when* is doing this anyway... I'm usingCisco ATA, or X-pro softphone, so I am aware of silence-suppression (I have switched it off). Thank You for any help, Arek Bekiersz [EMAIL PROTECTED]
