similar to: Early audio (long distance codes) not working after upgrading to 1.8?

Hey! I am installing Sangoma A102D wanpipe driver and i got following error. what is this ? why dir isn't there ? wanpipe-3.5.16 # make dahdi DAHDI_DIR=/usr/src/dahdi wanpipe-3.5.16 # make install ....Send .... .... Installing Wanpipe Firmware update utility in /etc/wanpipe/util/wan_aftup install -D wan_aftup /usr/sbin/wan_aftup install -d /etc/wanpipe/util/wan_aftup/scripts install:

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

Hi all, I am having a very strange problem. I am terminating a PRI (5ESS switch type, national plan, 23B+1D (24)) into a Sangoma A104 and am not able to produce any audio heard on the PSTN end of the call. Not sure what's wrong - the card worked before under a Trixbox setup. I'm running kernel 2.6.19 (tried 2.6.24.3 but had to downgrade as wanpipe stuff would not compile), zaptel

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk

Hi All, I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend. My dialplan: exten => _XXXX,1,Dial(SIP/${EXTEN},60,rt) Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument In 1.6 there was no problem, I have got Channel is

Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of the bug tracker tickets that it is instigated by a small network leg looks to be on point to me,

Thanks for replying. It's there, /usr/src/zaptel and it's valid. The script checks for its existence before compiling. The only thing that's unusual about this zaptel is that it is patched to support KB2 echo can. Gavin: Thanks, already cc'd Sangoma support. Hope they are working today! -----Original Message----- From: Eric "ManxPower" Wieling

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Sangoma support did something and the driver is there. Rebooting now with the new card. Hold me, I'm scared. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Friday, December 22, 2006 10:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma Wanpipe 2.3.4-3 compilation fails un der FC2

LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o Building modules, stage 2. MODPOST *** Warning: "register_wanec_iface" [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! *** Warning: "unregister_wanec_iface" [/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined! CC /root/wanpipe/patches/kdrivers/wanec/wanec.mod.o LD [M]

Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months

Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, why udev's for board in "/dev/zap"(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : )

I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get "Ring requested on unconfigured channel". How can I busyout these these channels so that incoming

Just a little questions, what's the difference between asterisk 1.8 and asterisk 11? Best regards.

We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back

Hi list. some info: zaptel 1.4.0 wanpipe 2.3.4-2 kernel 2.6.19.1 Debian I'm trying to build wanpipe on my server, but I got a error that it can't find config.h.. I found a post on an other unrelated mailing list which stated that includes/linux/config.h has been removed from 2.6.19. It also suggested replacing all references to config.h with autoconf.h make -C

Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux 2.2.0-rc2 which is what you get when you get dahdi-linux-current.tar.gz Anyone have a workaround or patch? Error below ==================== Building modules, stage 2. MODPOST CC /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.mod.o LD [M] /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.ko make[1]: