similar to: Similar option as promiscredir to use in transfer (REFER)

Hi, I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while to fix all the deprecated stuff, but everything seems to be working fine now, except for a little tiny thing. I lost all color in my CLI, which makes it harder to debug. Is there something that needs doing? I didn't explicitely disable colorization from the command line, and I did try using nocolor=no in the config files.

Hi everyone, Sorry if I'm repeating the e-mail, but I'm having problems with the list. I'm currently trying to enable call transfer to different domains in asterisk box (Asterisk 1.2.13 running on Debian etch). I have a configuration that requires me to transfer call to separate domains like ext at 10.10.10.10:5050. My calls come from a R2 channels in a board installed in the machine.

Hi all, I don't know if this is the right list to ask, since I'm using Trixbox version 1.0.0.28, that has asterisk 1.2.17. I'm trying to configure the ring timeout value for my local extensions (when dialing from one to another), and the dial command simply ignores my values... I have one extension 0017 in my box, so I used the command Dial(SIP/0017|100|rTtWw) to dial to it. The call

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

I am playing with tcpenable... on 13.11.2 so in sip.conf I have tcpenable=yes tcpbindaddr=192.168.1.8:5070 but when I "telnet localhost 5070" I get no connect. iptables -L -n -v | grep 5070 0 0 ACCEPT tcp -- * * 0.0.0.0/0 0.0.0.0/0 state NEW tcp dpt:5070 firewall is good. Is my syntax not correct above to run on port 5070 for SIP over TCP?

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue.

Hi All, I have a process running on port 5070... I'm using CentOS 7. iptables is running firewalld should be stopped and disabled. When I telnet localhost 5070 I get connection refused. When I stop iptables I still get connection refused. netstat -tnlv | grep 5070 tcp 0 0 192.168.1.8:5070 0.0.0.0:* LISTEN so the process is running and listening. ps ax |

Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 -> 192.168.2.20:5080

Hi, On a few of the hpc cluster nodes, i am seeing a new lustre error that is pasted below. The volumes are working fine and there is nothing on the oss and mds to report. LustreError: 5080:0:(import.c:607:ptlrpc_connect_interpret()) data3-OST0000_UUID at 192.168.2.98@tcp changed handle from 0xfe51139158c64fae to 0xfe511392a35878b3; copying, but this may foreshadow disaster

Chip Clock HT Cache Bus Speed --------------------------------------------------------- 5080 3.7 GHz YES 2MB 1066 MHz 5160 3.0 GHz NO 4MB 1333 MHz Does the .7 GHz and HT worth more then 4MB cache and higher bus speed? The application is VoIP so there is not a lot of IO so I would not think Bus Speed would matter. I am finding mixed information on HT, some say it is great, others say it

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

How can I mute the message "please enter the number you wish to call and press the # key" in a2billing??? I tried use_dnid = YES but still I keep getting the message prompt... thanks

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP

_ Vbox 4.2.18 VM booted with FreeDOS floppy, kernel 2041, no config.sys, no autoexec.bat, no TSR's, no memory managers. _ The booting floppy includes syslinux.com 6.02. _ Two HDD images attached to the VM, MBR + 1 FAT16 formatted partition each. Executing syslinux.com -i c: (or "d:") results in the following error: "Invalid Opcode at AD04 5080 0206 5080 2021 3666 FFFD 0083

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be

Hi, im trying to build a UAC and I'm coming up with some trouble whenever I receive a SIP 491 Request Pending Response. This happens because I try to place a call on hold using an Invite request rigth before Asterisk sends me a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting