similar to: Simple CDRs w/Asterisk/OpenSER.

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

Sorry if this hit the list twice, sent out yesterday, but didn't see it show up. Hi All, Can Asterisk be used as a SIP proxy, blah, blah, blah??? I've glanced over questions like this through the years, with a good idea on what a SIP proxy is and what Asterisk is and IS NOT. I never really took the time to lab-up SER and test drive it to see what advantages might be gained from using

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration

We have been working a OpenSer/Asterisk solution to replace our Avaya PBXs.The OpenSer is to provide scalability and the Asterisk to provide rich features.I know this has been many times for calling card platforms but I'm not sure if anyone has a good scalable solution they are using on their virtual PBX or in a CPE PBX environment?If so I would like to talk to them about buy their deploying,

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

Hello, I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Practically is an easier way to scale starting from existing asterisk installations. The other

Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because "OpenSER does only signalling while Asterisk does all". My question is: If Asterisk also does only signalling

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

Is anybody using XEN in conjunction with Asterisk and/or Openser? I would like to get some info about such an environment and experience reports. bye Ronald Wiplinger

Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

> On 17/03/2020 14:52 mabi <mabi at protonmail.ch> wrote: > > > ??????? Original Message ??????? > On Tuesday, March 17, 2020 1:38 PM, Aki Tuomi <aki.tuomi at open-xchange.com> wrote: > > > If you have message-id you can use > > > > `doveadm search -u user HEADER MESSAGE-ID "messageid"` to find out the UID. > > I tried the

I'm taking the liberty to announce this event on the Asterisk mailing list, as Asterisk and OpenSER form a valuable combination in SIP architectures. The second edition of OpenSER Summit will take place in San Jose, USA ,on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

Hi All, I am stuck with an issue in the Openser+Asterisk Forking. In this solution we are using Openser as the Registrar. Hence it will store all the contact bindings along with the q values for a given user, say ua1. The current setup is such that the INVITEs are sent to Asterisk by Openser and Asterisk sends out the INVITE. Now if ua1 is registered with two different contacts having

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - ->

Hi! Meanwhile also the location is fixed: it is happening at metalab (http://metalab.at/) - a place for geeks. Thus, we meet there at Thursday, 20.9.2007, 19:00 CEST (=local Vienna time). Metalab is located next to the city hall: http://metalab.at/wiki/Lage Metalab is no pub/restaurant. Thus, don't come hungry! Nevertheless liquid food (drinks) is available. We meet in the library (in

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Hello Users , I Posted to mailing list, No one is replying My issues, My Issue is No Audio when Openser and Asterik integrated in Same LAN , When UAC are Behind the NAT, With out the Asterisk integration Behind the NAT is working Fine. SIP port and RTP ports are forwarded into router to OpenSER System only. openser.cfg listen=192.168.2.11 alias=sip.hyperion.com # Invite Section if ( method==