similar to: SIP header values in the dialplan

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

I was messing about integration of a Cirpack softswitch with Asterisk and banged my head against a problem previously noted on the list. http://lists.digium.com/pipermail/asterisk-users/2003-November/026436.ht ml What is the status of this problem? Has it been fixed? I scrambled through chan_sip.c, but couldn't find ay reference to "multipart". Regards, Jesper Dalberg

Hi list, I have been looking around for a while now, and cant seem to get to the bottom of my problem. My setup is that I have a separate SIP server that servers my SIP subscribers, and I want to use Asterisk purely for voicemail for now. So I set up a common SIP extension at my SIP server, and made Asterisk register it, so that normal users can forward calls to that common extension, and

There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Op 30-5-2018 om 14:01 schreef Barry Pearce: > Hi, > > Im on Manjaro linux fully up to date running > > dovecot 2.3.1-2 > pigeonhole 0.5.1-2 > > > All is running well except I am having problems with fileinto - it > refuses to use variables, and mailboxexists is also failing for me. > > Im just dealing with plus addressing - should be simple but the

On Wed, 6 May 2020 14:16:32 +0800 Jason Wang <jasowang at redhat.com> wrote: > We tried to reserve space for vnet header before > xdp.data_hard_start. But this is useless since the packet could be > modified by XDP which may invalidate the information stored in the > header and IMHO above statements are wrong. XDP cannot access memory before xdp.data_hard_start. Thus, it is

On Wed, May 06, 2020 at 04:34:36PM +0800, Jason Wang wrote: > > On 2020/5/6 ??4:21, Jesper Dangaard Brouer wrote: > > On Wed, 6 May 2020 14:16:32 +0800 > > Jason Wang <jasowang at redhat.com> wrote: > > > > > We tried to reserve space for vnet header before > > > xdp.data_hard_start. But this is useless since the packet could be > > >

Hi, Im on Manjaro linux fully up to date running dovecot 2.3.1-2 pigeonhole 0.5.1-2 All is running well except I am having problems with fileinto - it refuses to use variables, and mailboxexists is also failing for me. Im just dealing with plus addressing - should be simple but the behaviour Im experiencing isnt right. require ["variables", "envelope",

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas -------------- next part --------------

Thanks for that. Turns out it was an issue with exim stripping the local part suffix from the envelope before passing over lmtp to dovecot. Fixed in the exim router config! Although if you turn on trace with spamtest it does result in a crash during mail receipt: lmtp(test at test.net)<7988><cDK3ADztDls0HwAANIsFaQ>: Panic: Unsupported 0x30 specifier starting at #38 in 'extracted

On 2020/5/6 ??4:21, Jesper Dangaard Brouer wrote: > On Wed, 6 May 2020 14:16:32 +0800 > Jason Wang <jasowang at redhat.com> wrote: > >> We tried to reserve space for vnet header before >> xdp.data_hard_start. But this is useless since the packet could be >> modified by XDP which may invalidate the information stored in the >> header and > IMHO above

On 2020/5/6 ??4:21, Jesper Dangaard Brouer wrote: > On Wed, 6 May 2020 14:16:32 +0800 > Jason Wang <jasowang at redhat.com> wrote: > >> We tried to reserve space for vnet header before >> xdp.data_hard_start. But this is useless since the packet could be >> modified by XDP which may invalidate the information stored in the >> header and > IMHO above

Hi List! I use Asterisk as a pure voicemailbox at a customers place. Right now, a telephone uses up two msns, one for the telephone itself, and one for the telephones mailbox. If the user is absent, a telephonecall is redirected to the voicemail msn of that users telephone. The Problem is: The PBX supports a too small number of msns, so I can't give every user a voicemailbox. Mailboxes are

Hi: when add a new voicemailbox, asterisk will create a new directory to it. since linux has limitation for the number of subdirectory. i wonder how many voicemailbox can asterisk support? thanks. yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030430/bd36cdaa/attachment.htm

Hi! I'm sorry I could not reply on the original message ( https://stat.ethz.ch/pipermail/r-devel/2015-April/070943.html), while I wasn't a subscriber of the r-devel mail-list. But I got the ./configure to work in the end. See below for the solution. ./configure 'LDFLAGS=-R/glob/jesper/software/bzip2-1.0.6/ -R/glob/jesper/software/zlib-1.2.8 -L/glob/jesper/software/bzip2-1.0.6

Hello, Got some at the time surprising errors some days ago when building a local r-devel installation on a cluster, with apparent outdated or missing dev versions of some files. After reading the r-devel news ( https://developer.r-project.org/blosxom.cgi/R-devel/NEWS), it turned out that " Use of the included versions of ?zlib?, ?bzlib?, ?xz? and PCRE is deprecated: these are frozen and

Hello, I'm having an issue with caller ID in voicemail that I'd appreciate any input on. I have two sip peers defined as extension 100 and 101 each with separate voicemail accounts. Each sip peer has its own DID number, which is established via cid_number = 6021231234. When a call is placed from SIP peer #100 to SIP peer #101, and SIP peer #101 wants to reply to #100's

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

On Wed, May 06, 2020 at 02:16:32PM +0800, Jason Wang wrote: > We tried to reserve space for vnet header before > xdp.data_hard_start. But this is useless since the packet could be > modified by XDP which may invalidate the information stored in the > header and there's no way for XDP to know the existence of the vnet > header currently. What do you mean? Doesn't XDP_PASS