similar to: obd call drops after few seconds : only for mobile numbers

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hello, lfs prints different obd(idx) compared to lctl dl. We use single striping. cluster1 tmp # lfs find --obd scia-OST0017_UUID /data/scia/L0/V0.00/20100327/SCI_NL__0PNPDE20100327_193441_000040582088_00071_42209_1158.N1 /data/scia/L0/V0.00/20100327/SCI_NL__0PNPDE20100327_193441_000040582088_00071_42209_1158.N1 cluster1 tmp # lfs getstripe

Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. Now i`m unable to load chan_dahdi and libss7: myserver*CLI> module load chan_dahdi.so ?ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method 'ss7' at line 37. myserver*CLI> module load libss7.so Unable to load module libss7.so

Hi. Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to Asterisk List. If somebody knows where to search (dahdi lists or libSS7 lists) will be appreciated. Im getting this error after a certain time, My config is: Hardware: 3 Digium Quad E1 TE4XXP libss7 version: SVN-branch-1.0-r286 DAHDI Version: 2.4.0 Echo Canceller: Asterisk 1.6.2.14 CentOS release 5.5 (Final) Kernel

Hei! I'm trying to send special characters out to ss7 link, but libss7 seems to convert them to zeroes. The challenge is that our service provider demands some of the regional numbers to be sent in format D0+number. When I use D in front of the number in dialplan, libss7 replaces it with 00, So I have a dial string: exten => _[A-Z].,1,Dial(DAHDI/g1/DD0501,,g) But in SS7 trace I

Hello List, To whom it might concern: I have been working in some SlackBuilds (script for making Slackware Packages) for my personal use, but thought they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as

For those who are using SuSE: At last they've managed to create ready-to-run packages for openSUSE_11.1. They are there since a couple of hours... (For other versions it was allready available for some time on the OBS) /srv/distro/repo/network:/telephony:/asterisk/openSUSE_11.1/x86_64/asterisk16-devel-1.6.0.6-82.2.x86_64.rpm

Hello List, I have been trying to setup T38 gatewaying with the following setup SIP ->Asterisk -> DAHDI TE410P with Libss7 -> TELCO I'm using asterisk Asterisk 1.8.3.2 and DAHDI Version: SVN-trunk-r9697M Echo Canceller: HWEC I'm aware there's no support for T38 gateway but I have been trying to get the patches https://issues.asterisk.org/view.php?id=13405 to work. It seems

The Asterisk Development Team has announced the release of libss7 2.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libss7 The release of libss7 2.0.0 resolves several issues reported by the community and would not have been possible without your participation. Please note that this version of libss7 has been released in anticipation of what

The Asterisk Development Team has announced the release of libss7 2.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libss7 The release of libss7 2.0.0 resolves several issues reported by the community and would not have been possible without your participation. Please note that this version of libss7 has been released in anticipation of what

Dear all, i wan to configure digium te412p card as ss7 interface. i installed libss7,dahdi and asterisk. Do i need to install lip pri also to work on ss7? best regards. Kashif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090418/7fa92295/attachment.htm

Hi, I am trying to set up asterisk with ss7. Whenever I try to load module chan_dahdi.so, I get the error [Mar 26 17:33:27] ERROR[10437]: chan_dahdi.c:10458 mkintf: Unable to find linkset -1 I have compiled dahdi, libss7, asterisks (am using asterisk 1.6) in that order. Have already set signalling to ss7 in dahdi_channels.conf How do I sort this out? Thanks for your help in advance. Peter.

Hello We are experiencing some difficulties with T38 faxing. I have a Asterisk 11.5.0 with libss7 and Sangoma A104DE digital interface card . The operating system is Centos 6 We are using this server to terminate calls to Telco. So calls are coming to asterisk from sip and we are sending calls to Telco with Dahdi. (It is a one way interconnection only from asterisk to telco ,not from telco to

The Asterisk development team has released new versions of three libraries used with Asterisk. They are: libpri-1.2.8: This release contains a number of bugfixes that had been unreleased for months, along with clarification of the licensing of the source code. The change log is here: http://downloads.digium.com/pub/telephony/libpri/ChangeLog-1.2.8 libpri-1.4.7: This release contains primarily

I have setup my asterisk box using freepbx. I can call extension and make outbound calls. the outbound calls drop between 10-30sec. we are using bandwidth.com and they have logged our call. below is your bad followed by what they say is a good call. I can't figure out where the problem is on your end. I know we are missing some stuff at the bottom but I don't know where to start.

Hi, Does anybody have an idea as to why dahdi_test results drop to unacceptable levels after doing a wanrouter stop/start using a Sangoma card? See below that it drops from 99.99% to 98.55%: [root at bin]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999512% 99.992874% --- Results after 2 passes --- Best: 100.000 -- Worst: 99.993 -- Average: 99.996193, Difference:

Hi, I?m trying to use SS/ in Asterisk. I'm thinking in chan_ss7 and libss7, and I want to know some other experience with this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100121/f8c4937e/attachment.htm

I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of

Try setting directmedia=no in sip.conf. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when.... Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: