similar to: No sound with internal calls depending on which phones

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

Le 08.12.2014 21:06, Marc Muehlfeld a ?crit : > Hello Denis, > > Am 08.12.2014 um 20:25 schrieb (lists) Denis BUCHER: > >> We have perfectly working roaming profiles on Samba 3.3.10 (SuSE) with Windows 7 clients. We configured our new server with same domain name, Samba 4.1.11 (Debian). On the new server, for newly created profiles, it works perfectly, we can login, logout,

Dear all, We have perfectly working roaming profiles on Samba 3.3.10 (SuSE) with Windows 7 clients. We configured our new server with same domain name, Samba 4.1.11 (Debian). On the new server, for newly created profiles, it works perfectly, we can login, logout, profiles are created and saved. But if we want to copy an existing profile from current server to the new one, it's

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

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, 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've configured with the very same script 1 Intel Xeon and 1 Intel Pentium4 machines. On one MOH is working properly On the other, I can read on console, lines such as those bellow but I can't hear anything. In which direction, should I further investigate ? If this help, here is my setup: me ---<PSTN-ISDN> ---- Patton 4638 ---<SIP>--- Asterisk 1.6.1.18 --

I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password:

Hello, I have a large data frame that is organized by date in a peculiar way. I am seeking advice on how to transform the data into a format that is of more use to me. The data is organized as follows: STN_ID YEAR MM ELEM X1 X2 X3 X4 X5 X6 X7 1 2402594 1997 9 1 *-00233* *-00204* *-00119* -00190 -00251 -00243 -00249 2 2402594

Hello, My setup is : phone ----- PSTN/ISDN ----- Patton SN4638 ------- Asterisk (Asterisk is in 1.6.1.18, Patton in 5.3) When I call the Asterisk, I can read from console that : - the call comes in, - the line MusicOnHold(,10) in my diaplan is reached and played, - I see RTP packets coming in and out (hundreds of lines such as: Got RTP packet from 192.168.102.200:4890 (type 00, seq 005360,

My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE

Hi, I have been trying to get my asterisk box to send SMS's to my Panasonic dect phone via a Linksys pap2. I believe I have the message centers setup correctly between * and the phone. The pap2 is configured to only use G711a. The Asterisk version is 1.0.7. In my /etc/asterisk/extensions.conf I have [smsphone] exten = 199,1,Goto(smsmorx,${CALLERIDNUM},1) [smsmorx] exten =

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

I'm testing two options from dial command and can not make them to work. L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) w: Allow the called user to start recording

Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1=> Dial SIP/phone1 Call 2=> Dial SIP/phone2 Call 3=> Dial SIP/phone1 <...> If you have an sample please let me know

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqbala at improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106