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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 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, 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

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,

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

Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally

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

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

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}" =

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

Samba Version: 3.4.7 OS: Ubuntu Lucid 10.04 Setup: This samba box is a member of a win2k active directory domain and functions as a file server. Files/directories shared out utilize file system acls. smb.conf portion for share in question: [Accounting] comment = Accounting Share path = /netdrives/accounting browsable = yes read only = no map archive = no map system = yes

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 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

Dear all, I?m testing a server and try to simulate a server in production. We have a SSL certificate and I have configured the test server with the same servername as it is in production. To access it, I change the hosts file in my laptop to reach the test server. However, the Java application running in the server tries to access some local web content. I have changed the hosts file

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 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

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

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