This sounds like it could be a permissions issue. What you might try doing is instead of just entering a standard number such as 8461234 is to append the ip or dns name of the CUCM such as 8461...@10.1.10.10<mailto:8461...@10.1.10.10> and see if that gets you anywhere.
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Becker, Jesse Sent: Thursday, November 18, 2010 1:49 PM To: Discussion list for users of sipXecs software Subject: [sipx-users] Transferring offsite disconnects All, Our college is currently using Cisco Unified Communications (V8) for all phones. I was recently granted approval to pilot SipX as an alternative in our IT department by the college administration. I am currently trying to replicate our Helpdesk auto attendant (currently house on Cisco Unity/CUCM) and migrate it to SipX. Before getting onto the issues, let me give you some more background on how things are connected. Our Local and Long distance is provided by PAETEC via 3 PRIs. These PRIs are connected to a Cisco 2811 router and managed as trunk groups via gateways on CUCM (Call Manager). I have a SIP Trunk configured on CUCM to allows calls to be redirected/sent to the sipXecs server. It is configured to send certain dial patters direct to the sipXecs server (via translation patterns). I also have SIP pattern for my sipXecs domain and IP configured to be sent to the sipXecs server. On the sipXecs, I have the CUCM system setup as an unmanaged gateway. I then use dialing rules to send desired calls from sipXecs to CUCM extensions as well as use the CUCM as a PSTN to the outside world. Currently, sipXecs extensions can call CUCM extensions and vise versa with no issues. Several DIDs are translated on CUCM to allow the outside world to dial directly to a sipXecs phone without any issue. sipXecs phones can also dial local and long distances numbers through CUCM to the outside world without any issue. So far everyone has been working great. The next step was to move the Automated Attendant for our office house by Unity to SipX system. I had no issues setup up the AA, however, one of the components is for some of the menu options to ring some of our office phones and if they go unanswered, be transferred of site to our central helpdesk. To accomplish this, I configured hunt group that would ring the desired phones (at the same time) for 10 second. I then configured a fallback destination which was a dialing pattern to call out through the PSTN (via CUCM) to our central Helpdesk.When I test from a sipXecs phone, this works fine. The desired phones ring for 10 seconds, then there is a short pause and the call starts to ring out to the central helpdesk which is then answered. The issue is when the call begins from the outside word. In this scenario, some someone dials a DID that is redirected to the sipXecs AA. When selecting the menu option, it rings the desired phones for 10 seconds, brief pause and then begins to ring out through the PSTN to our central helpdesk. However, once answered, the call gets disconnected. I am trying to track down the issue here. I am wondering whether this is a call AWK issue if it may be an anti-call trombone mechanism build-in to either sipXecs or CUCM (given that the source of the call is from the PST, goes through CUCM, to sipXecs and then circles out from which it came). I understand this is a unique and possibly complex scenario. I am open to any recommendations. I am guessing this would work fine if we had a more direct PSTN sources instead of going through CUCM for incoming and outgoing calls. -- Jesse Becker | Technical Manager | CCNA, Linux+, Network+ SUNY Ulster, 491 Cottekill Road, Stone Ridge, NY 12484 Tel 845-687-5064 | Fax 845-687-5105 beck...@sunyulster.edu<mailto:beck...@sunyulster.edu> | www.sunyulster.edu<http://www.sunyulster.edu> Check out our knowledge base: http://kb.sunyulster.edu
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