similar to: Desperately need help with Asterisk setup

Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a "beat" sounds, and then nothing else. In the console, I can

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid

Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

Hello Using Asterisk 12.8.2. I now have the "via ICE" messages in the RTP debug (see below). If you look in the SIP debug (see below), you also now see the "ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the webRTC client. But still no audio ! None at all ! In both directions. You can see in the SIP debug that the IP-address in de

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

Hello, this is killing me: i can''t find a way to let idle sessions expire after a certain period of time. If they don''t expire, they will pile up until there''s no more memory available. Please, can anybody give me a hint? Basically, all I need is a hash or list with all the sessions in DRbStore. Greetings Michael Kastner

Hi there, I am desperately trying to build a suitable wine.conf, but somehow I just don't manage to. I have been reading through different wine documentation and help pages. I used winecheck to try to figure out what is wrong. However, winecheck tells me constantly that I have to configure the paths of my drives in a special way: /my/mount/point. I tried to apply that (of course not

Friends! I have recently discovered that chan_sip, chan_sip2 and chan_sipx all lack support of SIP multicast. This has a major impact on my network, since I haven't got the bandwidth needed to call all of you and send you this message. With that feature missing, I have to go back to old Internet communication methods and use e-mail instead:

Hello, I am desperately trying to get Apache Solr to work with Dovecot FTS. I would really appreciate if someone could please help me! I have already done the following:- 1. I can ssh into my server and see that Apache Solr is up and running. ssh -t -L 8983:localhost:8983 user at mydomain.com 2. In the collection1 core selector I have the following files:- solrconfig.xml and schema.xml The

I have been trying to setup Samba on my home network but... it is giving me a hard time. I went to the diagnosis.txt that comes with Redhat 7.2 and tried some of the tests. I tried the first three and on the first one I got an error message. "session request to [servername] failed (Not listening for calling name". I got a second one in that same first test "session request to

I just moved a new user onto my Samba server. He needs two things, and neither of them work! 1. When he tries to access one of the printers, he gets Access denied. Only for one of them. I can't find anything in the logs. When I try to access the printer as his user, I get: [2004/12/06 11:16:59, 5] smbd/uid.c:change_to_root_user(296) change_to_root_user: now uid=(0,0) gid=(0,0) I

On Thu, 8 Mar 2018 15:58:43 -0600 (CST) Brent Davidson via samba <samba at lists.samba.org> wrote: > I am desperately in need of help. I have a Centos 7.2 server running > Samba 4.6.13 as an active directory domain controller. I am trying to > join a new Centos 7.4 server running Samba 4.6.13 to the domain. The > domain command will not connect to the other server. > >

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

All; I'm running Asterisk 11.6-cert9 and am trying to play a pre-recorded audio file to extensions using the Page() command. The dial plan looks like this: exten => s,n,Page(${AVAILCHANS},A(${AUDIOMSG})) and the paging by itself works great. However, when I try it with the audio file, it starts to play correctly, then abruptly hangs up after 6 or 7 seconds. When I turn debug on, this

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

2010/10/24 Olivier <oza_4h07 at yahoo.fr> > > > 2010/10/14 Danny Nicholas <danny at debsinc.com> > >> ------------------------------ >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olivier >> >> *Sent:* Thursday, October 14, 2010 3:34 PM >> *To:*