similar to: Can't hear MOH from PSTN

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 --

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:*

Hi, On a Lenny system, with dahdi 2.4.0, libpri 1.4.11.5 and asterisk 1.6.1.18, I inserted a new Digium HA8 + B400M card. My usual installation fails. I can see it listed : # lspci -n | grep d161 01:0b.0 0200: d161:8007 (rev 11) # lspci -vn 01:0b.0 0200: d161:8007 (rev 11) Subsystem: d161:8007 Flags: medium devsel, IRQ 22 I/O ports at d400 [size=256] Memory at

Hi, I'm facing an issue with which loading wctdm24xxp module fails. Here is relevant dmesg's output : [ 13.455729] dahdi: Telephony Interface Registered on major 196 [ 13.455729] dahdi: Version: 2.4.0 [ 13.510847] ACPI: PCI Interrupt 0000:01:0b.0[A] -> GSI 22 (level, low) -> IRQ 22 [ 15.527788] wctdm24xxp 0000:01:0b.0: Timeout waiting for receive frame. [ 17.527787]

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

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

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, In my lab, I'm testing BRI spans in NT/ptmp mode. My setup is: asterisk 11.2.1 libpri 1.4.14 dahdi 2.6.1 wctdm24xxp (HA8 hybrid with B400M) SIP phone <----> Asterisk with HA8 <----> Patton SN4638 <----> Asterisk <----> SIP phone The single BRI line I'm testing remains down: CLI> pri show spans PRI span 1/0: In Alarm, Up, Active I'm quite certain this

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

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 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

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

This may be a bonehead question, but when I have some error in a view I''m testing, the server spits out a generic page: "Application error Change this error message for exceptions thrown outside of an action (like in Dispatcher setups or broken Ruby code) in public/500.html" This is pretty unhelpful when I''m debugging. Is there an error log for Rails overall

Adding an Answer() before MusicOnHold made it works. Thanks for everyone that helped ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101024/4c5546b0/attachment.htm

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

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

Owen Anderson wrote: > Is this actually the case? I can't find it documented anywhere on > MSDN or the rest of the internet. C:\Program Files\Microsoft SDKs\Windows\v6.0A\Include>grep -n -F MemoryFence WinNT.h 2231:#define MemoryFence _mm_mfence C:\Program Files\Microsoft SDKs\Windows\v6.0A\Include>grep -n -F MemoryBarrier WinNT.h 2288:#define MemoryBarrier __faststorefence

I''m using a db to log two types of events, link events and tag events. At first I had two tables, link_events & tag_events, but this seemed not very DRY because the only column that was different between them was link_id and tag_id respectively. So I made these tables: events: id int user_id int objtype_id int (holds ''1'' or ''2'' depending

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