similar to: Call being answered, but no audio on either end (Intermittent)

Thanks Gene. Here is my localnet: localnet=172.16.64.0/255.255.240.0 Which matches our subnets network address and subnet mask. Are you recommending that I make it more restrictive? Thanks, Geoff > -----Original Message----- > From: Gene Willingham [mailto:gwillingham@comcast.net] > Sent: Tuesday, June 14, 2005 9:13 PM > To: asterisk-users@lists.digium.com > Cc:

I think I found the source of this. Been tracing it for a week. Look in sip.conf. It appears the definition of localnet has a bearing on how some sip devices handle invites and NAT. I had changed the localnet to 192.168.3.0, but did not change the netmask. localnet=192.168.3.0/255.255.0.0; All RFC 1918 addresses are local networks When I changed the netmask to 255.255.255.0 the problem

Hi All, I made the decision recently to update some old call servers from 1.2.x to the latest 1.4.28. I spent 2 weeks in the lab testing every production requirement for these voice servers. Nothing too special, SIP in, database lookup, SIP out, a couple of special applications, write CDR to billing server, nothing intensive at all. I also performed stress testing for call capacity, database

I think you are absolutely right. The audio I heard earlier sounds exactly like a timing issue. So: wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs I'm going to make this change and reload at lunchtime, I'll document it and post it to the list if it works.

Followup on this issue, it appears that using a single PRI's clock as the master clock avoids clock drift between the PRI's and we get no more artifacts. So, : wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs -----Original Message----- From: Michael L. Young

Aha, it just happened to me, so now I can characterize the audio: It basically sounds like it's missing every other sample - fuzzy and distorted. Timing?

Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call

I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: "very bad phasing reverb & feedback (from my rock & roll days)". This is quite intermittent, as in most cases, the user

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm having some problems using the asterisk modem driver with an ISDN4Linux card (AVM Fritz). (I realise that it's better to use CAPI, but unfortunately this card doesn't have any CAPI drivers). Every so often the ISDN just stops working (it neither dials out nor accepts incoming calls). Trying to dial out just logs: --

Hello all, I've got an intermittent issue with my asterisk set-up, and I'm pulling my hair out! Mostly everything works fine, but I get an error once every few days that sometimes (but not always) causes asterisk to stop accepting new calls and also stop responding to commands on the console (asterisk -r). The usual errors look something like this: [Feb 19 13:09:52] ERROR[18728]

Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and

Dear OpenSSH gurus, I have recently (yesterday) upgraded my machine running Red Hat Linux 6.1 with all of the binary rpms generated from the source rpms openssh-2.2.0p1-2.src.rpm and openssl-0.9.5a-3.src.rpm. Since the upgrade, I have been experiencing intermittent failures. The failures are always accompanied by this error message: response: Authentication response too long: 1433299822

6727404: intermittent fdisk errors while installing xVM Server guests http://bugs.opensolaris.org/bugdatabase/view_bug.do?bug_id=6727404 Is that (at least case #1) the same as OpenSolaris 2008.05 bug 2133 (comment #22)? http://defect.opensolaris.org/bz/show_bug.cgi?id=2133 In some cases the Solaris ata driver is returning a slightly truncated capacity with the DIOCTL_GETGEOM /

> > --- Original message --- > Subject: [Samba] Intermittent file rename problem with Vista, works > with XP. > From: "Massner, Keith" <kmassner at mgmresorts.com> > To: <samba at lists.samba.org> > Date: Friday, 23/07/2010 2:44 PM > > So... It's been driving me nuts. I get intermittent permission > denied > errors when I try to

Hello, I'm on Samba 4.6.3. I'm seeing intermittent error 54f response to LDAP binds on 1 of 4 DCs. I wrote a script to test the DCs: https://pastebin.com/egsY3W9H, which produced these results: https://pastebin.com/6UsvwVgX. I'm not seeing any errors in the samba log. Thanks, Arthur This e-mail and any attachments may contain CONFIDENTIAL information, including PROTECTED

A problem has come up in our computer labs. We use one account for a given lab, except in the main lab where students have individual accounts. In either case, printing is only intermittently successful. We use a "net use lpt1 \\server\printer" command in a batch file run at login to connect the printer. The command completes successfully. The CUPS status page for the printer

Rare and intermittent message groups suggest a race condition ... ? Here's the pattern of messages: rsync: make_bak_dir mkdir <path> failed: File exists (17) rsync: keep_backup failed: <path> -> <path>: No such file or directory (2) rsync: stat <path> failed: No such file or directory (2) How rarely? We run a scripted rsync daily on ~50 personal computers and

We are having intermittent problems where the person we call reports static when we place an outgoing PSTN call. Only the person called hears static, to us the conversation sounds fine. Never happens on inbound calls. It doesn't matter what channel you call from (IAX, SIP, or Zap). We have a Sangoma A108D with hardware echo cancellation with 2 PRIs to Level3 and 2 PRIs to a Nortel Option

Hi guys, How are you ? I've installed samba-3.0.7 from source on a Slackware Linux 10.0 server configured with "--with-ldap" as a backend (OpenLDAP-2.2.13) and with nss_ldap installed. I have my profiles share configured as follows: [profiles] comment = Profile Share path = /data/profiles writeable = yes guest ok = yes browseable = no profile acls = yes csc policy = disable

I''ve got an issue and I think it''s webrick related. I''m loading images dynamically, from a database. I''ve got an Image controller with a "show" action that takes an id as a parameter. The problem is that images sometimes come up broken when loading more than one on the same html page. I think this is because my browser tries to load them simultaneously