similar to: Asterisk, DNS SRV, 1.8

Has anyone successfully had a SIP phone fail over from Asterisk Server A to Server B using DNS SRV? If so, which phone worked for you? I'm assuming you set up your DNS SRV records so that the IP addresses of A and B are associated with the same name, and both servers have equal priority and equal weight. In order to make calls through B after A goes down, do you have to wait as long as the

Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device

I tried that (only needed to add rewrite_contact=yes) but it didn't help. BTW, the CONTACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message----- From: Eric Wieling [mailto:ewieling at

I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtmf The question is, how do I change it without changing the source code? On Sat, Jun 7, 2014 at 1:00 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To

Heya, Just wondering if anyone has deployed a DNS SRV example that I can call to test my new asterisk install? Just want to listen to an IVR or recorded message to test I can call test@test.com or whatever. Can't find one on google :( Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.18/89 - Release Date:

I've tried to find documentation on if Asterisk supports DNS SRV records for sip servers. Reading the source of channel_sip.c it seems not: hp = gethostbyname(hostname); if (!hp) { ast_log(LOG_WARNING, "Host '%s' not found at line %d\n", hostname, lineno); return -1; } This is in the register part. Am I correct in this or have I missed something?

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me

Hello, When I called companies with auto animate menus my system does not seem to detect menus on ther other side. For instance I called this number (407) 886-3338 when I input the ext. number of any option on the list I don't get a response however if I called the same number from my google account or my cell phone number it works fine meaning I can select any option or input ext number.

Thanks, Eric. I just tried a hangup handler, but it's showing a similar problem: When the peer hangs-up, the hangup handler is not invoked and the caller channel remains open. same => n(callPeer),Set(GLOBAL(Peer${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount} + 1]) same => n,Set(CHANNEL(hangup_handler_push)=handleHangupByCallerOrPeer,doesntMatter,1(args)) same =>

type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888 at

I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19

Something perhaps noteworth, since this is a multihomed system I bound the transport to 172.31.253.4:5060 I don't *think* that would cause Asterisk to use that IP in the FROM...at least it shouldn't. -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 2:58 PM To: 'Asterisk Users Mailing

This has been super-helpful, Eric. However, the handleHangupByPeer priorities below are still not run when the peer hangs-up. The last line in the cli when the peer hangs-up is still: Strict RTP learning complete - Locking on source address (Although sometimes there is also: Retransmission timeout reached on transmission) same =>

Hi Yves.. This may be silly... but what is the useragent of your sip configuration? In the case that useragent has some special characters like "(.", please remove it and tell us if there is any change!!. Regards. rv 2014-11-22 14:50 GMT-03:00 Eric Wieling <EWieling at nyigc.com>: > Try setting directmedia=no in sip.conf. > > -----Original Message----- > From:

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

Please can i ask you i want to know which code can help me to provide the taxation of voip/toip services in asterisk Le mar. 18 sept. 2018 à 01:36, Patrick Wakano <pwakano at gmail.com> a écrit : > Thanks everyone for the answers! > I did explored some options at the PHP level and probably will do > something in this direction, but in fact what I was really looking was >

Hi, i'm looking to update some SRV DNS Record , but i didn't find the correct syntax to handle priority, weight and port. The goal is to higher the priority of one of my different DCs. Thanks for your help Alain

My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 AAAA CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? ----- Andrew Thompson http://aktzero.com/

Hi, I am looking to use DNS SRV records for load balancing and failover across multiple Asterisk servers. The Asterisk servers share the exact same configuration via mySQL replication. I would like to know which particular SIP phones support DNS SRV and would like to hear of any success stories. Many SIP phones claim to support DNS SRV, yet there is usually very little documentation on how to

This patch teaches ovirt-test to use the usual DNS SRV records to connect to qpidd so we don't have to enter the server etc. on the command line. Signed-off-by: Ian Main <imain at redhat.com> --- src/ovirt-agent/ovirt-test.rb | 8 ++------ 1 files changed, 2 insertions(+), 6 deletions(-) diff --git a/src/ovirt-agent/ovirt-test.rb b/src/ovirt-agent/ovirt-test.rb index