similar to: Help! Connecting two Astersik via SIP channels

Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>

I am trying to setup dynamic updates with bind_dlz backend, but for some reason if any windows client or linux with nsupdate tries to remove AAAA record, server just 'cancelling transaction', while A and PTR records (both on reverse ipv4 and ipv6) working fine. If i'am remove AAAA record manually via samba-tool or windows mmc then AAAA record can be updated, but after that it again

virtio blk has some legacy feature bits that modern drivers must not negotiate, but are needed for old legacy hosts (e.g. that dn't support virtio scsi). Allow a separate legacy feature table for such cases. Signed-off-by: Michael S. Tsirkin <mst at redhat.com> --- include/linux/virtio.h | 4 ++++ drivers/virtio/virtio.c | 25 ++++++++++++++++++++++++- 2 files changed, 28

virtio blk has some legacy feature bits that modern drivers must not negotiate, but are needed for old legacy hosts (e.g. that dn't support virtio scsi). Allow a separate legacy feature table for such cases. Signed-off-by: Michael S. Tsirkin <mst at redhat.com> --- include/linux/virtio.h | 4 ++++ drivers/virtio/virtio.c | 25 ++++++++++++++++++++++++- 2 files changed, 28

Hi, I,ve a run a model like this mcrm<-glm(catroj~month,binomial) being catroj a binary response variable with two levels (infected and non infected) > anova(mcrm3,test="Chisq") Df Deviance Resid. Df Resid. Dev P(>|Chi|) NULL 520 149.81 mes 3 16.86 517 132.94 0.0007551 *** When I?m trying to do a post

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An

Hi to all!! I'm successful to connect Asterisk to PostgreSQL database... If it's possible, can anyone learn me how to store sip user in PostgreSQL database and how to configure voicemail?? Thanks for all!!!

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Dear Sirs, I had compiled PWlib and OpenH323 correctly in my Fedora Core 2. But when I try to compile asterisk-oh323 I get the following error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' How can I solve it? Thank you for your help. Juanjo

Hi, I am new to the Asterisk world. I don't know much about the architecture, but I am involved in installing and configuring the VoIP system. My requirement is to build a VoIP system using the 4 input lines (ISDN up0 telephone lines), it must be possible to receive calls from outside through the 4 ISDN up0 input lines, and also possible for outgoing calls, conferencing .etc. I

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to place calls on hold using Asterisk Manager Actions? Amaury -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk

On 2017?01?27? 16:16, Christoph Hellwig wrote: > We don't really need struct virtio_pci_vq_info, as most field in there > are redundant: > > - the vq backpointer is not strictly neede to start with > - the entry in the vqs list is not needed - the generic virtqueue already > has list, we only need to check if it has a callback to get the same > semantics >

On 2017?01?27? 16:16, Christoph Hellwig wrote: > We don't really need struct virtio_pci_vq_info, as most field in there > are redundant: > > - the vq backpointer is not strictly neede to start with > - the entry in the vqs list is not needed - the generic virtqueue already > has list, we only need to check if it has a callback to get the same > semantics >

I try to set up Samba 2.2.4 / LDAP as a PDC and it almost works. The only thing I don't understand is why a domain user can't have a primaryGroupID of 513 (which looks like it should be a safe default). But if I set it, login is denied with an error C0000078 on the client, and something like [2002/06/03 10:32:28, 3] smbd/sec_ctx.c:set_sec_ctx(314) setting sec ctx (65534, 65534) -

In my Asterisk server following things have been done to detect answering machines before the answered call connects to the agents in queue. In extension_additional.conf ============================== [ext-queues] include => ext-queues-custom exten => 5000,20,Macro(user-callerid,) ; changed the priority to 20 ............... ============================== In extension_custom.conf

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Hi, I'm stumped. I can access the postfix server to read my mail. BUT I'm the only user that can. Everyone else get rejected. Here's the syslog entry for another user: ...dovecot: pop3-login: Disconnected (auth failed, 1 attempts): user=<jmarino>, method=PLAIN, rip=192.168.0.51, lip=192.168.0.189 And here's an entry for me: ...dovecot: pop3-login: Login: