Displaying 20 results from an estimated 200 matches similar to: "SIP DTMF problem with SNOM"
2009 Jan 20
5
the FXS ports of Digium and damaging if connected to Tel Line
Hi All;
I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo.
What is the solution for this disaster?
Regards
Bilal
2012 Feb 25
0
No IVR audio. Jump in RTP sequence number
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
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
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
2015 Nov 12
3
No sound with internal calls depending on which phones
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
2008 Mar 20
1
Newbie: Two problems with Asterisk Config, Please Help
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
2015 Nov 12
3
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
2009 Jan 16
2
UpdateConfig : Appending line fails
Hello list,
Can someone please point me out why would a stream like the following
only write ONE line (the first) on the given file?
Action: login
Username: test
Secret: 123456
Action: UpdateConfig
SrcFilename: voicemail2.conf
DstFilename: voicemail2.conf
Action-000000: Append
Cat-000000: default
Var-000000: 127
Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do
ActionID:
2020 Sep 24
2
Negotiates g729 but RTP contains g711
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 Oct 14
1
Default MOH not working on 1.6.1
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
--
2011 May 24
2
Data Frame housekeeping
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
2010 Oct 24
1
Can't hear MOH from PSTN
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,
2020 May 17
1
PJSIP sending RTP to private address
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
2008 Feb 14
1
Winbind problem with more details.
Everyone,
One of our developers was kind enough to insert some bug checking into the mod_auth_pam and mod_auth_sys_group so that we could see a little more of what was going on with our authentication failures. Here is what we just saw. Two of our users NA\connelmp and NA\guminssa both started getting messages that they were not part of the required group. Here is the log for
2003 Jul 29
6
kernel deadlock
We have a reproducible problem with FreeBSD-4.7 which is apparently a
deadlock.
The system is undergoing a filesystem stress test.
The machine is pingable, but console and most other features are
unresponsive.
The console debugger can be accessed.
The following information is available with db's "ps".
I suspect the wchan of "inode" to be what everything is waiting on.
2014 Feb 03
1
Incoming Fax Issue with Asterisk 11.7 and Digium Fax
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
2013 Nov 28
1
RTP packets send, but no audio
Hello,
What does it mean when "rtp set debug ip" shows RTP packets that have
been send, but there is no audio ?
There was no audio on my call in both directions, but "rtp set debug"
shows that there were RTP packets send.
There is no firewall active on my Asterisk server :
[root at sip asterisk]# /sbin/service iptables status
iptables: Firewall not running.
Kind
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
hi,
i have following topology
PSTN - Asterisk ---- internet ----- router - jssip client (wss)
Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP
connection to PSTN
router - public IP/private IP (NAT)
jssip client - private IP - sip over websocket to Asterisk PJSIP
~30% of calls has problem with no audio. reason is that Asterisk is
sending RTP to private IP of jssip
2008 Mar 17
1
Desperately need help with Asterisk setup
Hi,
I am new to Asterisk and I am having a setup problem that I am trying to
resolved for the last couple days without any success. I am pretty much
desperated on this issue and I don't know why. Can someone please kindly
help me to troubleshoot this? I can't hear any audio from Asterisk when
running Playback or VoiceMail tests.
I have my Asterisk server ( running on Debian,
2016 Aug 09
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
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
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
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