search for: sbcs

I'm a bit further along but haven't figured out why Authentication is still failing. I've tried a telnet to port 143 and openssl connection to 993. The command I issued, per the debugging page on the wiki, is: a login info at aesoft-sbcs.com crap Here is a snapshot from my logs (yup second try and blank lines to make it easier for me to read). Oct 7 08:17:20 mx0 dovecot: auth(default): client in: AUTH 2 PLAIN service=imap secured lip=216.64.180.226 rip=66.193.34.88 lport=993 rport=65026 resp=AGluZ...

...the data and config files over, correcting ownership and permissions (hopefully) as I went. But now I can't get logged in. Messages in /var/log/dovecot/dovecot-info.log, without saslauthd running, are like this. dovecot: Feb 03 08:55:46 Info: auth(default): passwd-file(raanders at aesoft-sbcs.com,66.193.34.88): no passwd file: username_format=raanders /var/mail/vhosts/aesoft-sbcs.com/passwd dovecot: Feb 03 08:55:48 Info: auth(default): client out: FAIL 1 user=raanders at aesoft-sbcs.com dovecot: Feb 03 08:55:48 Info: auth(default): client in: AUTH 2 PLAIN service=IMAP secured...

...ne of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we were using it anyway when we setup a (Asterisk+SER+SIP Proxy) Box to handle the "on-net dialout" calls. I'm now overwhelmed with the amount of SBCs that are suggested by the vendors to implement a solution. (http://www.juniper.net/solutions/literature/solutionbriefs/351085.pdf) Can anyone drop me some lines about this? I urgently need some feedback on this. Thanks! Joao Pereira www.fccn.pt

...pected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC matches an endpoint based on ip and dials the uplink: -- Executing [+31xxxxxxxxx at outgoingrr:9] Dial("PJSIP/sbcs-00000092", "PJSIP/+31xxxxxxxxx at uplink") in new stack -- Called PJSIP/+31xxxxxxxxx at uplink -- PJSIP/uplink-00000093 is making progress passing it to PJSIP/sbcs-00000092 -- PJSIP/uplink-00000093 answered PJSIP/sbcs-00000092 -- Channel PJSIP/uplink-00000093 joined 'simple_brid...

...e mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261

Anybody know of a UK supplier of "Voip Telephony with Asterisk" " by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261

...rectory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to be me, never do anything right !) Thanks -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261

Is anyone succesfully running Asterisk behind verizon residential DSL? I seem to be having some problems with my Asterisk server switching to Verizon. I'm attempting to do some troubleshooting, but I'm really interested in knowing of anyone's setup that already has Asterisk working with Verizon residential DSL. Thanks AJ ------------------------------------------------------ This

...second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is on the same subnet as the c1700; there are no nat/firewalls/sbcs in the middle. at ~15:15 the asterisk console reads: WARNING[2492]: chan_sip.c:3778 retrans_pkt: Maximum retries exceeded on transmission CB674A02-B25C11DF-B6D5A08D-652FE73E at 10.9.1.9 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. WARNING[2492]: chan_sip.c:3805 retrans_pk...

I am running samba version 1.9.18p4 on a sinix machine (unix svr4). My NT workstations use the german code page 850. In earlier versions of samba (1.9.17p2) the character mapping worked just fine, using character set = iso8859-1 client code page = 850 in my smb.conf. Since upgrading to samba 1.9.18p4 this don't work any more. Is this a known problem? I tripple-checked my smb.conf entries,

Hi! updated to p4 the other day, and now the "character set = iso8859-1" option doesn't work anymore. I didn't change anything in the smb.conf file but this: unix password sync = True time server = yes and it worked in 1.9.18p3 :/ Has anyone else seen this? Samba runs on FreeBSD 2.2.6 from the ports collection. (I don't think that password sync works for this

...screen are- editline/libedit.a db1.ast/libdb1.a stdtime/libtime.a -ldl -lncurses -lm -lresolv -lssl There is obviously something I have not installed, but what ? Have searched archives and thro package descriptions and come up with nothing Any help appreciated -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261

Is there a way to ensure that the source IP address from witch the SIP user register is not tampred with , is there a feild in the SIP register message header can be used to achive this security ? i have an asterisk server in witch SIP users register through an SBC(session border controller) , i wanna make sure that those users are really registering from the IP they are claimming they are

Hi everybody, I'm testing asterisk@home 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from syslogd@asterisk1 at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user xxxx@yahoo.com could

How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication

I am curious if it is advisable to use implement Asterisk as a Session Border Controller for a VoIP reseller environment. Users will terminate calls SIP to my server, which will authenticate them via RADIUS, perform a LCR lookup, select an appropriate trunk (based on LCR), and terminate the call (update RADIUS accounting at end of call). All while acting as a B2BUA to prevent the users from seeing

...sed to get Asterisk certified with Verizon's network? Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same thing with a Linux box (routing box and sip proxy box) without having to spend money on SBCs or expensive Cisco gear, that is the route we would like to go. We are looking at the possibility of handling 140 concurrent calls... that is what they are designing on their end as well. So, I am asking the community for any input. I have read on here and seen on IRC that some in the community...

Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:

...one through the string with no match - * at the string end. - */ - - size_t mb_back_len = str_charnum(back); - size_t mb_s_len = str_charnum(s); - - while(mb_s_len >= mb_back_len) - { - size_t charcount = 0; - char *mbp = s; - /* - * sbcs optimization. - */ - if(!global_is_multibyte_codepage) { - while(charcount < (mb_s_len - mb_back_len)) { - mbp += 1; - charcount++; - } - } else { - while(charcount < (mb_s_len - mb_back_len)) { - size_t skip =...

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike