similar to: Internet exposed asterisk server.

Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000

Hi, Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server? I'm getting an error: "403 Authentication user name does not match account name" As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header

I place a call into Asterisk (from SIP phone) and the To header does not have a tag. Asterisk then sends it's Trying response, still no tag in the To header. The phone then replies with OK, this time the To header includes a tag. Is there any way to retrieve this response To header (including the tag field) from the dial plan? I have tried the PJSIP-HEADER read of the To header, but it

Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP

Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should

Could some please confirm that this behavior is incorrect. I am seeing issues where it appears that asterisk is not following the Record-route on it's reply messages. Please let me know if you require any other information. Thanks Example: xxx.yyy.154.243(PSTN-GW) <--sip--> xxx.yyy.77.23(Asterisk) <--sip--> xxx.yyy.91.74(SNOM or SER proxy) <--sip---->

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

After doing some system work, including upgrading the Samba server to 3.0.28a from 3.0.24, upgrading the kernel to 2.6.24, and changing the firewall rulesk, the XP workstations which belong to that domain, the right click "run as ..." option is slow to bring up a dialog. The phenotype is this: right click some program (for instance, a shortcut to the "command prompt")

Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect

We have frequent timeout problems after upgrading our imap servers from dovecot 1.1.16 to dovecot 1.2.11. One server acts as proxy only, and the other one is the "real" imap server". The credentials for the proxy service are stored in a remote MYSQL database. There were no trouble with dovecot 1.1.16. But now, with the most recent version, we get frequent login failures. It

The PJSIP endpoint is configured for ulaw only. Not sure how or why we are seeing the g729 on calls for this endpoint. Would this be a case that asterisk detects the rtp stream is g729 even though it's negotiated as ulaw? Why would asterisk change the format to g729 when disallow = all and allow = ulaw are the endpoint settings? [121] type = endpoint context = IS transport = transport1 aors

I have the raw data with 9 column and 197977 row: > dummy State Months Bedroom 1 xxx Jan 1 2 xxx Jan 2 3 xxx Jan 1 4 yyy Jan 1 5 yyy Jan 2 6 yyy Jan 1 7 zzz Jan 3 8 zzz Jan 1 9 zzz Jan 2 10 xxx Feb 3 11 xxx Feb 4 12 xxx Feb 2 13 yyy Feb 1 14

$ dovecot -n # 1.1.11: /etc/dovecot/dovecot.conf # OS: Linux 2.6.28-11-server x86_64 Ubuntu 9.04 protocols: imap imaps managesieve I need to make an IMAP (actually imaps) server available over the Internet. Unfortunately, VPN is not available (not all clients support VPN), so I will have to expose the imaps port to the Internet. My question is: how reliable is Dovecot in such a setup? I am not

Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the

Hello all, taking examples from various pointers, I am attempting to put together an outbound dialing example using SIP (Cisco 7960) with 2 X100P. Everything seems to be working without generating errors, but the problem is the phone hangs up (102/Bye). Any pointers/advice are much appreciated Here is the section in extensions.conf: extensions.conf ; From CISCO at work ; exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a mail server CentOS 64bit (4 Gb RAM) with Dovecot 1.2a4 and three accounts. I use Thunderbird 2. If I enable SSL connection in Thunderbird 2, after three-five minutes I got a lot of different errors in Thunderbird (Server is not IMAP, Connection lost...). Everything comes back to normal if I restart Dovecot in the server, but after 5 minutes

Thanks for the reply. Le 2014-08-26 12:30, steve a ?crit?: > On Tue, 2014-08-26 at 12:02 +0200, Cyril Feraudet wrote: >> Hi all, >> >> I get an error when I try to join domain from CentOS 6.5. Have you an >> idea ? >> >> >> /etc/samba/smb.conf : >> --------------------- >> [global] >> workgroup = XXX >>

Hello, First of all sorry for my English. I am experiencing with Samba and I have a problem. I have an old server (OLD) with Red Hat 9 and Samba 2.2.7a that is working well. Now I try to start up a new server (NEW) with Red Hat Enterprise 4 and Samba 3.0.22. If I try to connect from NEW to itself by using smbclient I got the shared resources list correctly. If I try to connect to NEW from OLD,

Hello! We recently upgraded one of our customers from 1.4.44 to 1.8.15-cert1. We have several other customers running both versions. The customer in question does not use us as their provider as they?re located in a different country. When they make outgoing calls, there is a 3 second delay between answering the call and the call being established. When debugging this, I found that Asterisk

Hi, I've been trying to build a password forwarding proxy to Gmail without success... The SSL connection to Dovecot is happening no problem (as far as I can tell), but for some reason the conversation between Dovecot and Gmail is getting timed out. I know this is supposed to be simple... :-( But could somebody please give me some help by pointing what I'm not doing right? No matter