similar to: no route to host

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

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

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

Hi, I'm currently setting up an Apache SSL reverse proxy for Exchange 2003 Outlook Web Access. The setup that I have works fine on my Gentoo laptop or on a Trustix server, however, when I try to set it up on an RHEL based distro, with the exact same virtual host settings, I get some weird error with the authentication mechanism. I have tried with both CentOS 4.2, based off the server CD

Hi All, I have the data like this : >sample <- read.csv(file="sample.csv",sep=",",header=TRUE) > sample stdate Domain sex age Login 1 01/11/09 xxx FeMale 25 2 2 01/11/09 xxx FeMale 35 4 3 01/11/09 xxx Male 18 30 4 01/11/09 xxx Male 31 3 5 02/11/09 xxx Male 32 11 6 02/11/09 xxx Male 31 1 7 02/11/09

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

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

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

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")

Dear All, I have the data some thing like this, I am showing here three days data only: > dummy.data <- read.table(file='dummy.txt',sep='', header=TRUE) > dummy.data StDate Domaindesc Logins 1 05/01/10 xxx 10 2 05/01/10 xxx 45 3 05/01/10 xxx 2 4 05/01/10 yyy 45 5 05/01/10 yyy 20 6 05/01/10 yyy

Hi all, I have data like this: >sample <- read.csv(file="sample.csv",sep=",",header=TRUE) > sample stdate Domain sex age Login 1 01/11/09 xxx FeMale 25 2 2 01/11/09 xxx FeMale 35 4 3 01/11/09 xxx Male 18 30 4 01/11/09 xxx Male 31 3 5 02/11/09 xxx Male 32 11 6 02/11/09 xxx Male 31 1 7 02/11/09

Hi, I''m having an issue with changing the source address of packets origininating from a specific user using a combination of iptables and iproute2. We want to change packets originating from a specific service so that they originate from that service''s IP address, so that we can move it to a different physical server and have the packets appear to come from the same

Anybody else seen this one? We're running Dovecot 1.0.rc7 on a different port on our live server (we're generally live with 1.0.beta5) and after a HUP to reload the user database (in a passwd-file) this morning we got:- > dovecot: Sep 22 07:45:05 Info: IMAP 14482 someuser xxx.xxx.xxx.xxx : Disconnected in IDLE > dovecot: Sep 22 07:45:05 Warning: imap-login: SSL_read() syscall

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

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

Hi, I've done upgrade from Ubuntu 13.04 (samba 4.0 alpha) -> 13.10 -> 14.04 (samba 4.1.6). Now it looks I've lost some accounts. samba-tool dbcheck reports problems but cannot fix them with --fix param: ERROR: incorrect GUID component for member in object CN=MobileEX,CN=Users,DC=yyy-local,DC=localdomain -

Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2

Hi All! I am puzzled. I am using openssh 5.3p1 on RHEL/CentOS but am seeing the same behaviour for osx and debian: ssh -R xxx:yyy:yyy:yyy remotehost Will effectively drop the first (xxx) argument and result in ssh -R 127.0.0.1:yyy:yyy:yyy remotehost ssh -R yyy:yyy:yyy remotehost I had a look at the source code where apparently all 4 values are ready and handed down to the appropriate

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

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 server string = Samba Server Version %v log file = /var/log/samba/log.%m max log size = 50 realm = XXX.YYY security = ads idmap uid = 10000-20000 idmap gid = 10000-20000