similar to: MailMan slowness...

don't quote me on this but I believe the earlier assumtion is correction. I think you need to have RTP going bothways otherwise the call will disconnect. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Shaw Sent: Friday, August 13, 2004 12:40 PM To: asterisk-users@lists.digium.com Subject: Re:

I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when I or anyone else calls from PSTN -> * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell

They say on their website that they allow you to use your own device provided you give them the MAC address. Has anyone tried using * with it? Looks like they have quite a few rate centers and also phone support... Their website is horrible though... Just wondering, it'd be good to get user experiences from different providers other than IconnectHere and BroadVoice... -Chris

I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

Now that most of you have worked overtime to show why most people are continually pissed at Nix Users (all except two of course). The problem I can see is the downright technosnobbery involved. There is nothing wrong with Linux. I play around with RH9 and FreeBSD and find that most things run fine. But you get into a problem where it keeps asking for the same blamed libraries over and over on

I have a bridge that has gigabit interfaces. The machine in question has the fun job of being a Bridge, Firewall and SMB server. Both of the Gigabit interfaces are connected to workstations directly via Xover cable (well MDI-X to be exact). My question is, if I enable jumbo frames on the gigabit interfaces will that make any difference in overall transfer rate of the bridge? I was thinking it

Did my last post make sense? Is this a known issue with the bridge-nf code? Is there something I can do to help? Should I just shut up now? -Thanks in advance Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

Ok I have googled and googled and combed through the wiki for an answer to this and have come up empty. What I'm finding is that when a user changes their VM password, it is saved somewhere like maybe the CSV database or something because when you log in, the new password works fine, but it's not saving to voicemail.conf. So new passwords are lost when asterisk is restarted and people

Back to the ACTUAL TOPIC of this thread... This phone looks kinda nice, where can one get hold of it? How about it's * compatibility? I realize that it says it does things like 3-way conference and attended transfers, but how about in *? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

Ok.. this is really wierd... I just cleaned up my dialplan a bit by adding some macros with a strange side effect... On my incoming context which has no macros in it, far end ringing used to work... now that I have macros defined, far end ringing has stopped working all together... The macros DO work, but when they transfer, the far end ringing sounds terrible and even skips a few rings... If I

CVS 7/16/04 (the latest one I have b4 today) seems to have this problem too... Anyone else having problems with the current CVS ignoring calls after about 5 minutes of being up? I've also noticed that no matter what I set default_expiry to in sip.conf, it starts at that number and then jumps to 44 seconds... not sure where it's getting the number 44 from, it seems to use that all the

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

I have an idea, tell me if this wouldn't work... I know it's really ugly, but it might help some people until we can get round robin DNS checks for peers... Since * does not do GetHostByName() again until you reload your config, and BroadVoice and I'm sure other sip providers are using round robin DNS, why not create 2 [<your server here>-out] contexts in sip.conf, and then in

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

On Friday, 30 November 2018 20:35:07 GMT David Miller wrote: > From: Jens Axboe <axboe at kernel.dk> > Date: Fri, 30 Nov 2018 13:12:26 -0700 > > > On 11/30/18 12:56 PM, Davidlohr Bueso wrote: > >> On Fri, 30 Nov 2018, Kees Cook wrote: > >> > >>> On Fri, Nov 30, 2018 at 11:27 AM Jarkko Sakkinen > >>> <jarkko.sakkinen at

Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31

CentOS Errata and Security Advisory 2007:0775 https://rhn.redhat.com/errata/RHSA-2007-0775.html The following updated files have been uploaded and are currently syncing to the mirrors: ia64: updates/ia64/RPMS/mailman-2.1.5.1-34.rhel4.6.ia64.rpm -- Pasi Pirhonen - upi@iki.fi - http://pasi.pirhonen.eu/ Top-postings silently ignored -------------- next part -------------- A non-text attachment

CentOS Errata and Security Advisory 2007:0775 https://rhn.redhat.com/errata/RHSA-2007-0775.html The following updated files have been uploaded and are currently syncing to the mirrors: s390: updates/s390/RPMS/mailman-2.1.5.1-34.rhel4.6.s390.rpm s390x: updates/s390x/RPMS/mailman-2.1.5.1-34.rhel4.6.s390x.rpm -- Pasi Pirhonen - upi@iki.fi - http://pasi.pirhonen.eu/ Top-postings silently

CentOS Errata and Security Advisory CESA-2006:0600 mailman security update for CentOS 3 i386: https://rhn.redhat.com/errata/RHSA-2006-0600.html The following updated file has been uploaded and is currently syncing to the mirrors: i386: updates/i386/RPMS/mailman-2.1.5.1-25.rhel3.7.i386.rpm source: updates/SRPMS/mailman-2.1.5.1-25.rhel3.7.src.rpm You may update your CentOS-3 i386 installations