FYI - I don't think I should see packet loss.There is packet loss during the DTMF. I am going out on a lomb here, but would suggest it is a call the patton is discarding because it does not know how to re-route it. I have seen this with call park return.
If this continues to be the case it would be an issue where the patton is having a problem with the destination. "Why" we don't know. I would suspect a dial plan entry in the SN to be able to convert or transform it in a way that would alleviate that. (i.e. why is the from "anonymous" where it should know where it is from). The firmware on the patton is correct. Do you have a way to put a 3.2 firmware on the destination (v v x)? On Thu, Jul 19, 2012 at 4:18 PM, Philippe Laurent <[email protected]> wrote: > Just heard from Patton, who indicated that there was a trans-coding issue, > with one side speaking ulaw and the other alaw during the transfer. Dunno > why this is, since all devices (Patton, PBX, phones) are set to prioritize > ulaw over alaw. The Patton 4112 apparently does not support trans-coding. > > So, I set the Patton to talk only ulaw, saved and reloaded (you know, for > good luck), and boom, it works. Not great, but it works. During the > transfer, I get music for about 2 seconds, and then silence. No ringing, no > anything, until a phone in the hunt group picks up. Ringing of continued > tunes would ease the mind of the person on the other end. > > Patton debug is on the way, but won't have access to the client's site > until this evening to do any testing. > > Philippe > > On Thu, Jul 19, 2012 at 4:06 PM, Tony Graziano < > [email protected]> wrote: > >> Can you shoot me a sip debug of the transaction? >> >> Since it is a sipx ivr (version 4.4/latest, right?), both transactions >> are "attended transfers" from the IVR (I am assuming). >> >> If it were me, I would try to create a phantom user with no VM >> permissions and have the forwarding on "all the time" to the intended >> recipient, then add that phantom user as an option on the IVR and see if >> that works. I have found that hunt groups have been geeting frustratingly >> difficult lately. Post back the results. >> >> On Thu, Jul 19, 2012 at 3:37 PM, Philippe Laurent <[email protected]> wrote: >> >>> Yeah, I know, this is a SipX list, not a Patton list. I'm working with >>> their tech support, but we're not getting very far, and since the Patton >>> 4112 is connected to a SipX box, and there a quite a few Patton fans out >>> there working with SipX, I wanted to make sure I wasn't missing something >>> obvious in the link. >>> >>> Patton 4112: SIP registered to SipX extension, FXS port connected to >>> pharmacy IVR >>> >>> Here goes: >>> 1. Caller dials 8 from the auto attendant, transfers to the pharmacy IVR >>> on the Patton FXS port. Success. >>> 2. If the caller decides they want to speak to someone while sitting on >>> the IVR on the Patton, they dial 2. >>> 3. The FXS IVR then dials the 600 extension to ring the hunt group on >>> SipX. The sequence used by the FXS IVR to transfer the call is Hook-flash, >>> then dial 600, then terminate. This fails to transfer the call (no >>> ringing), and the call terminates. >>> >>> Clearly something isn't done right, or I've missed a concept. Should the >>> FXS IVR should be dialing a different string set to transfer the call? The >>> vendor says they can define the transfer on their side to match >>> requirements. >>> >>> Many thanks in advance for your time. >>> >>> _______________________________________________ >>> 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: 434.465.6833 >> ~~~~~~~~~~~~~~~~~~ >> Linked-In Profile: >> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >> Ask about our Internet Fax services! >> ~~~~~~~~~~~~~~~~~~ >> >> Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab >> 2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013> >> >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: [email protected].**net<[email protected]> >> >> Helpdesk Customers: >> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> >> Blog: http://blog.myitdepartment.net >> >> _______________________________________________ >> 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: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! <http://sipxcolab2013.eventbrite.com/?discount=tony2013> -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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