similar to: asterisk-users Digest, Vol 54, Issue 94

Hi, I had a Trixbox 1.4.18 that I "yum update"d to 1.4.22.1. Now, I seem to have a huge problem with phones not staying registered (registrations worked perfectly at 1.4.18). I phone will register the first time I plug it in, and then once you make a call and hangup (or sometimes even during the call) all the lights will go orange meaning a lost registration. Every so often the lights

Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a "sip show peer" on those extensions shows them as "OK". Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",

Hi all, Disregard my previous posts, I've consolidated everything here. I'm having terrible performance issues with samba over a WAN (point-to-point T1 link). Doing a copy of a 2MB file from a samba server to a linux client running smbclient takes over 5 minutes. SCPing the same file takes seconds. The server is running samba version 3.0.25c with kernel 2.6.16.18. I've put up a set

Hi All, Fedora Core 26 # rpm -qa \*samba\* samba-common-4.6.7-0.fc26.noarch samba-common-libs-4.6.7-0.fc26.x86_64 samba-4.6.7-0.fc26.x86_64 samba-client-libs-4.6.7-0.fc26.x86_64 samba-winbind-modules-4.6.7-0.fc26.x86_64 samba-libs-4.6.7-0.fc26.x86_64 samba-client-4.6.7-0.fc26.x86_64 samba-common-tools-4.6.7-0.fc26.x86_64 samba-winbind-4.6.7-0.fc26.x86_64 I am replacing a CentOS 5 Samba Server

Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

Ok, I must be really stupid here - I'm playing with ael and svn trunk. given the following in ael: context isdn10 { 444601 => { Answer(); NoOp(${CALLERIDNUM}); Hangup(); }; }; isdn10 is the incoming isdn context. why do I get this on the console: -- Accepting call from '01702xxxxxx' to 'yyyyyy' on

On 08/19/2017 12:31 AM, Rowland Penny via samba wrote: > On Fri, 18 Aug 2017 19:53:26 -0700 > toddandmargo via samba <samba at lists.samba.org> wrote: > >> Hi All, >> >> Fedora Core 26 >> >> # rpm -qa \*samba\* >> samba-common-4.6.7-0.fc26.noarch >> samba-common-libs-4.6.7-0.fc26.x86_64 >> samba-4.6.7-0.fc26.x86_64 >>

Hello again, managesieve is working. Now I tried a vacation script with the result that the vacation response is sent but I got he following error: deliver(test1 at xxxxxx.de): 2008-12-15 14:34:28 Error: file_dotlock_create(/home/vmail/%d/%n/Maildir/.dovecot.lda-dupes) failed: No such file or directory /home/vmail/xxxxxx.de/test1/Maildir/ is the correct dir, there are the other files like

Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

Hello, Anyone else experiencing problems with the latest spandsp (pre3) and last libtiff beta? I'm getting 8 bytes long file, with the TIFF header only during such connection: -- Accepting call from 'XXXXXXX' to 'YYYYYY' on channel 0/2, span 1 -- Executing SetVar("Zap/2-1", "FAXFILE=/tmp/foch.tif") in new stack -- Executing

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even