similar to: dialing via SIP URI

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Hi All, In the midst of trying to pilot a deployment of Microsoft Lync (mainly for non-voice collaboration, specifically IM) and integrate it with our Asterisk (11.6.0 if it matters) deployment and a "everything in one place" tool when people are out of the office. I have everything on the voice side playing nice from the Lync side (Lync->Lync, Lync->Asterisk,

Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael

Does anyone know if Asterisk can accept cleartext auth (SIP), as in it recv's a call destined to: 1234:blah@har.har.com The problem I'm having is simply for faxing, normal calls come in as g729 and of course we need ULAW for faxes. sip.conf snippet [sipfarm] insecure=very host=blahblah.netlogic.net type=peer context=sip-out username=+18165551212 secret=blah canreinvite=no disallow=all

Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via

Hi, in a HTML page I can write href="mailto:joe@doe.com" and clicking on the link will open the default mail application. Is anything like that possible with any of the soft phones (SIP or IAX [Windows])? Any and all information is greatly appreciated. rgds pos

On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote: > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. > > I have a lot of endpoints and registrations on same SIP server. And it's > problem in pjsip now. Is not it? > > I

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]

I'm using the SJphone softphone, and I've got a nice little SIP-only setup, using (amongst others) stanaphone, VOIPtalk and FWD. But I'd like to be able to use my SJphones to dial directly to folks who provide a SIP URI, eg: 100@calluk.com, without either having to switch profiles in SJphone (to direct SIP) or having to define calluk.com (in this example) as a destination in

On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote: > 07.03.2015 0:24, Kevin Harwell ?????: > > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> > wrote: > >> Hello. >> >> Asterisk 13.2. >> I transfer configs from chan_sip to res_pjsip. >> In chan_sip i have

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

Try a Nokia E61/E62... Version 3 supports SIP and WiFi and they have a big battery that allows long talking and standby times. CS -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Justin Moore Gesendet: Donnerstag, 24. Mai 2007 10:16 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff:

In a recent interview someone from Digum indicated that the G722 wideband codec was being worked into Asterisk. This will make Asterisk compatible with Polycom's new HDVoice products like the IP650 phone. This is very interesting, potentially exciting, but it brings up certain questions. Who will benefit as long as calls must typically pass into existing PSTN infrstructure, and so be

Hi all, saw a few messages here, and read the part on the wiki on using asterisk to dial out via another SIP service provider, who incidently is also using Asterisk. First the details; PHONE1 Extension: 2002002001 IP Address: 192.168.128.25 ASTERISK1 Extension: 1111111111 IP Address: ASTERISK1 ASTERISK2 IP Address: ASTERISK2 Destination PSTN Extension: 2222222222 (Information changed

I know this one must be easy but I'm an newbye so please help. In my extensions.conf I want to have a line like: Exten => 9XXXXXX,1,Dial(Zap/4/${SOMETHING},40,r) Ie: I want to dial all the XXX-es, but not the 9; How do I do that? What do I write in place of ${SOMETHING}? Navigating the wiki didn't provide any usefull advice... Thanks.

On 4/29/19 5:42 PM, lameventanas@gmail.com wrote: > > > On 29/04/2019 22.01, Michal Privoznik wrote: >> On 4/29/19 1:06 PM, lameventanas@gmail.com wrote: >>> I want to run libvirtd as a special user, and allowing users that belong >>> to a special group to connect via qemu+unix:///system (eg: unix socket). >>> >>> I did everything necessary to

On 4/29/19 1:06 PM, lameventanas@gmail.com wrote: > I want to run libvirtd as a special user, and allowing users that belong > to a special group to connect via qemu+unix:///system (eg: unix socket). > > I did everything necessary to do so: created a libvirt user and group, > added the libvirt user to the kvm group, added my normal user to the > libvirt group, and made sure the

Does the SugarCRM included with AAH 1.3 not meet this criteria for you? ------------------------------ Message: 3 Date: Tue, 26 Jul 2005 17:48:08 +0100 From: "JunkMail" <junkmail@segurajuda.dyndns.org> Subject: [Asterisk-Users] Dial using URI(web) or using FORM(web) To: <asterisk-users@lists.digium.com> Message-ID: <009101c59201$cd5cc9c0$0a00a8c0@segurajuda.local>

On Tue, Apr 30, 2019 at 10:45:03AM +0100, Peter Crowther wrote: > On Tue, 30 Apr 2019 at 10:40, Michal Privoznik <mprivozn@redhat.com> wrote: > > > Is there any problem running libvirtd as root? > > > > Yes, in the regulated environment in which I work! I have to do far more > thorough threat analysis than I would do if I knew which capabilities it > had. So

On 4/30/19 3:15 PM, Peter Crowther wrote: > On Tue, 30 Apr 2019 at 10:48, Daniel P. Berrangé <berrange@redhat.com> > wrote: > >> On Tue, Apr 30, 2019 at 10:45:03AM +0100, Peter Crowther wrote: >>> On Tue, 30 Apr 2019 at 10:40, Michal Privoznik <mprivozn@redhat.com> >> wrote: >>> >>>> Is there any problem running libvirtd as root?