similar to: having trouble to register cisco 7975 with pjsip

Ok after I added tcp transport and disable force_rport phone get registered, but still have issues with calls, when I call from cisco from, it work except hangup. when I call to cisco phone asterisk return congested debug of call <--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 ---> INVITE sip:111 at 192.168.1.61:51179;transport=tcp SIP/2.0 Via: SIP/2.0/TCP

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018

Here's the output: Feb 9 15:51:26 SSI001 kernel: SFW2-INext-ACC-TCP IN=eth0 OUT= MAC=00:0f:ea:73:88:12:00:40:2b:67:5b:a7:08:00 SRC=192.168.1.54 DST=192.168.1.2 LEN=48 TOS=0x00 PREC=0x00 TTL=128 ID=51248 DF PROTO=TCP SPT=1964 DPT=139 WINDOW=65535 RES=0x00 SYN URGP=0 OPT (020405B401010402) Feb 9 15:51:28 SSI001 kernel: SFW2-INext-ACC-TCP IN=eth0 OUT=

Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!

I posted some questions a few months back and received lots of great help in getting my dovecot/postfix/LDAP setup working. Initially, I ran into the pipe failed too many open files error. I am running this setup on OpenBSD/sparc64 4.1-stable and so I put dovecot in its own class and upped the maxfiles setting for that class. This was back with dovecot 1.0 and everything worked fine. I

Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>

https://bugzilla.samba.org/show_bug.cgi?id=7975 Summary: Would be useful to warn/error on files excluded by -- update Product: rsync Version: 3.1.0 Platform: Other OS/Version: Other Status: NEW Severity: enhancement Priority: P3 Component: core AssignedTo: wayned at samba.org

Nick Awesome wrote: > Hay guys, got trouble with registration with cisco 7975 The "force_rport" option is incompatible with Cisco, it needs to be explicitly set to no in the endpoint. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

Nick Awesome wrote: > Ok after I added tcp transport and disable force_rport phone get > registered, but still have issues with calls, > > when I call from cisco from, it work except hangup. > > when I call to cisco phone asterisk return congested If you use UDP with force_rport=no it'll work. If you use TCP then set rewrite_contact=yes so it'll reuse the established

Full_Name: Matthias Wahl Version: 2.0.1 and 2.1.0a OS: Mac OS X 10.3.9 Submission from: (NULL) (84.159.39.159) Version 2.1.0a does get installed, R is available under applications, however when trying to start R I get an error message that R is terminated due to an error and that's it. Re-installing didn't help at all. Then I tried Version 2.0.1. and there I don't even get an icon

Hi All, I have a Drobo, connected to a CentOS 6.4 box. The box sees it as /dev/sdg. I want to format it ext3 (as they dont support ext4) but when I try I get: # fdisk -u /dev/sdg WARNING: GPT (GUID Partition Table) detected on '/dev/sdg'! The util fdisk doesn't support GPT. Use GNU Parted. WARNING: The size of this disk is 17.6 TB (17592186044416 bytes). DOS partition table

Hi, when I run > plot.default(z1, z2, xlab = "x", ylab = "y", main = "xxxx", pch = "+") > abline(lm(z1 ~ z2)) then the plot is plotted perfectly (scatterplot), however, the lm() function doesnt appear on the plot. What could be wrong? (Yesterday it worked perfectly, with the lm() line.) Running R 2 on OS X. Mathias Hunsk??r Furevik Norway

Greetings All, I am a Postfix/Courier-IMAP user and am interested in Dovecot. I have install Dovecot via FBSD ports. I have got to the part where I can telnet in to the server. -bash-2.05b$ telnet 192.168.1.61 143 Trying 192.168.1.61... Connected to 192.168.1.61. Escape character is '^]'. * OK dovecot ready. x login pete testing * BAD [ALERT] Plaintext authentication is disabled, but

Smbclient worked fluently to create tar compatible backups of all files on an SMB/CIFS share. For which I used an Ubuntu 8.04 LTS Server Edition configuration with LAMP, OpenSSH, Samba and BackupPC packages installed. Testing with smbclient \\\\win-pc\\share -U administrator -d 3 (debugging level 3) returns: (...) name_resolve_bcast: Attempting broadcast lookup for name win-pc Got a positive

Fedora: Linux abcde 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/Linux Asterisk: 1.2.4 SIP Problem 1. Asterisk sends SIP messages to Softphone. 2. Softphone receives SIP messages and replys back. 3. Asterisk doesn't receive these replies and keeps on sending. Asterisk: Reliably Transmitting (no NAT) to 192.168.1.4:5060: OPTIONS sip:192.168.1.4 SIP/2.0 Via:

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Begin forwarded message: > From: Matthias Wahl <matthias.wahl at macnews.de> > Date: 28 giugno 2005 18:05:54 GMT+02:00 > To: stefano iacus <jago at mclink.it> > Subject: Re: [Rd] Does not run under Mac OS X 10.3.9 (PR#7975) > > > I don' have a folder 'Utilities' in applications, however here is > what I get when I chose > 'send error

I don''t know whether this is the right place to ask, but kindly point me to an FM that I can R if it isn''t. My wife is creating lots of Kazaa traffic, and I am using rsync to create a full mirror of Red Hat''s FTP site, Aurora Linux FTP site, the LDP site, and some other stuff. Clearly, when one is moving well over 100GB over a 128 Kbps link, this is going to take a