similar to: problems in REFER request to a different machine

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.

I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like "promiscredir=yes" in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo!

Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but

CentOS Errata and Bugfix Advisory 2014:0778 Upstream details at : https://rhn.redhat.com/errata/RHBA-2014-0778.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: f91473eda2f7e38a532cd3d445e3ed10109ad72b5163f8bf53427083d4e3fd16 ql2400-firmware-7.03.00-1.el6_5.noarch.rpm x86_64:

CentOS Errata and Security Advisory 2019:0778 Moderate Upstream details at : https://access.redhat.com/errata/RHSA-2019:0778 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: b6a9b7941571bb174ce804b64d2ad94cc24f6af4e201196494c097de4a4ffaa8 java-11-openjdk-11.0.3.7-0.el7_6.i686.rpm

Hi, spandsp-0.0.2pre18 works fine for txfax with HT488(version-1.0.1.2), but rxfax doesn't work. After some FAX sounds, it hangup! Could someone tell me how to debug? The following is the * CLI> log to 192.168.0.161:43222 -- Executing NoOp("SIP/4881-bde9", "") in new stack -- Executing RxFAX("SIP/4881-bde9",

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When

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, 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

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When

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Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

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 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, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

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

Dear Customer Service, My name is Tamamura and I am in charge of Audio-Technica. I want to be able to play FLAC using the API that BT IC has. I use BT IC that other companies have released. In this case, Do I need to get permission from you? Best regards Minoru tamamura ==================================================== 〒915-0003 福井県越前市戸谷町87-1 (株)オーディオテクニカフクイ　技術部 　　　第3技術課　玉村　実 TEL：0778-25-6700

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

Hello, I was interconnecting Asterisk (v1.0) with a strict router (ie, no ;lr in routes) and I think I found a bug in the way Asterisk prepare new requests inside a dialog. I'm sending some captures (ngrep) along with my comments. This is a 200 OK (INVITE) received by Asterisk ========================= U 2005/02/10 16:41:55.065538 143.173.202.82:5060 -> 143.173.202.83:5070 SIP/2.0 200

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