if asterisk fails your incoming calls fail also. so you are suggesting throwing asterisk in between SS7 and sipxecs as an e1 gateway instead of an independent e1 gateway?
if so, i think this is adding more potential points of failure. also, you may find certain incompatibilities or configuration needs on asterisk and a more standard diskless e1 gateway much more robust and simple instead. On Thu, Aug 4, 2011 at 4:57 AM, cyril constantin <[email protected]> wrote: > Yes I have set up it like unmanaged gateway. The problem is that we earn > Million Euros per day with our activities, I have no right to make mistakes > if I replace Avaya per Sipxecs. > > > 2011/8/4 Michael Picher <[email protected]> >> >> If they are set as SIP trunks... if you can set them up as unmanaged >> gateways there's less chance of call disruption. >> Mike >> >> On Thu, Aug 4, 2011 at 4:40 AM, cyril constantin >> <[email protected]> wrote: >>> >>> Currently my production config is: >>> In front I have SS7 switches (to receive call from carrier) I have four >>> switches of this type, then these SS7 switches are connected by ISDN links >>> to Avaya and IP Phone are registered remotely (by international links) with >>> H323 to Avaya. >>> >>> So what I expect to have in the near future: >>> Keep SS7 switches and connect it by using ISDN links to Asterisk servers >>> (will act like PSTN gateway), then connect it with SIP trunk to sipxecs >>> servers (already done and works like a charm with SRV lookup) and register >>> all my sip softphone to Sipxecs >>> >>> So all incoming calls coming from Asterisk to Sipxecs or vice versa will >>> fail during proxy or registrar restart ? >>> Regards >>> >>> >>> >>> 2011/8/4 Tony Graziano <[email protected]> >>>> >>>> the registar and proxy resatarts do not interfere with ongoing calls. >>>> it means new calls cannot come in or go out for a 1-2 second duration >>>> while services are stopped/tested/restarted. media is not affected >>>> either. >>>> >>>> if it were me, and it's not, if i were using siptrunks and had that >>>> kind of call volume, then i would consider an external sbc so that all >>>> media for actual calls is not anchored by sipx with the exception of >>>> the media server (voicemail, aa and acd queue). in the case of >>>> openacd, the media is by FS, which reloads, not restarts, so is the AA >>>> and Vm in 4.4. >>>> >>>> This means if the agent was talking with someone you could yank the >>>> ethernet cable from sipx and watch the calls continue with >>>> uninterrupted rtp... if you were using a pri or other telco gateways >>>> the sbc is unneccessary too, just a good pstn gateway. >>>> >>>> >>>> >>>> On Thu, Aug 4, 2011 at 4:13 AM, cyril constantin >>>> <[email protected]> wrote: >>>> > Hi Guys, >>>> > Thanks for your feedback, so no one is using sipxecs for this amount >>>> > of >>>> > calls per day ? >>>> > My call center are running 24/24 7/7 so when I see that I need to >>>> > restart >>>> > services for small changes on sipxecs I can't imagine rollout that in >>>> > production yet (I'm running stable version), for example if I go to >>>> > dialplan >>>> > and apply a change on a rule it's require to restart SIP registrar and >>>> > SIP >>>> > Proxy on all servers, a reload will be preferable. >>>> > I'll try the nightly build to see the improvements. >>>> > >>>> > Best Regards. >>>> > Cyril >>>> > >>>> > >>>> > >>>> > 2011/8/3 Michael Picher <[email protected]> >>>> >> >>>> >> I agree with Josh... might be something you want to start testing >>>> >> with >>>> >> the 4.5.2 dev versions. I wouldn't expect you'd want to run in >>>> >> production >>>> >> until 4.8 ish... >>>> >> That being said, it will be a very robust solution with agents and >>>> >> queues >>>> >> being able to span multiple servers and being able to add servers to >>>> >> increase capacity at will... >>>> >> Mike >>>> >> >>>> >> On Wed, Aug 3, 2011 at 12:24 PM, Josh Patten <[email protected]> >>>> >> wrote: >>>> >>> >>>> >>> It should be noted that OpenACD is far from ready in sipXecs 4.4. >>>> >>> The >>>> >>> integration will be much tighter in 4.6, though I'm not sure if >>>> >>> there will >>>> >>> be much in the way of reporting and supervisor capabilities. Perhaps >>>> >>> one of >>>> >>> the devs could chime in? >>>> >>> >>>> >>> >>>> >>> On Wed, Aug 3, 2011 at 11:16 AM, Tony Graziano >>>> >>> <[email protected]> wrote: >>>> >>>> >>>> >>>> you really should read the wiki on the newer openacd package being >>>> >>>> integrated into sipx now for that kind of call volume. >>>> >>>> >>>> >>>> On Aug 3, 2011 12:13 PM, <[email protected]> wrote: >>>> >>>> > Hi Guys, >>>> >>>> > >>>> >>>> > I just want to know if one of you have experience with sipxecs on >>>> >>>> > call >>>> >>>> > center environment, for handling in my case 200 000 calls per day >>>> >>>> > and 350 >>>> >>>> > agents connected simultaneously during peak hours? >>>> >>>> > >>>> >>>> > I'm really interested by replacing my Avaya per sipxecs regarding >>>> >>>> > what >>>> >>>> > I have observed on sipxecs capacity. >>>> >>>> > >>>> >>>> > So let me know about you experiences. >>>> >>>> > >>>> >>>> > Best Regards. >>>> >>>> > >>>> >>>> > Cyril >>>> >>>> > >>>> >>>> > >>>> >>>> > >>>> >>>> > _______________________________________________ >>>> >>>> > sipx-users mailing list >>>> >>>> > [email protected] >>>> >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> sipx-users mailing list >>>> >>>> [email protected] >>>> >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>> >>>> >>> >>>> >>> >>>> >>> -- >>>> >>> Josh Patten >>>> >>> eZuce >>>> >>> Solutions Architect >>>> >>> O.978-296-1005 X2050 >>>> >>> M.979-574-5699 >>>> >>> >>>> >>> _______________________________________________ >>>> >>> sipx-users mailing list >>>> >>> [email protected] >>>> >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >> >>>> >> >>>> >> >>>> >> -- >>>> >> Michael Picher >>>> >> eZuce >>>> >> Director of Technical Services >>>> >> O.978-296-1005 X2015 >>>> >> M.207-956-0262 >>>> >> @mpicher <http://twitter.com/mpicher> >>>> >> www.ezuce.com >>>> >> >>>> >> >>>> >> _______________________________________________ >>>> >> sipx-users mailing list >>>> >> [email protected] >>>> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> >>>> >>>> -- >>>> ====================== >>>> Tony Graziano, Manager >>>> Telephone: 434.984.8430 >>>> sip: [email protected] >>>> Fax: 434.326.5325 >>>> >>>> Email: [email protected] >>>> >>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>> Telephone: 434.984.8426 >>>> sip: [email protected] >>>> >>>> Helpdesk Contract Customers: >>>> http://support.myitdepartment.net >>>> Blog: >>>> http://blog.myitdepartment.net >>>> >>>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> Ask about our voip fax services! >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [email protected] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> -- >> Michael Picher >> eZuce >> Director of Technical Services >> O.978-296-1005 X2015 >> M.207-956-0262 >> @mpicher <http://twitter.com/mpicher> >> www.ezuce.com >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our voip fax services! _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
