HI Joe Thanks for your help.
I am running CUCM 7.1.5 I have configured a non secure sip trunk to the IP address of the SipX server I have used the standard SIP profile, I Have left all settings as default, please let me know if there are specific settings of interest, or I can send a screenshot if that would help. I have configured the CUCM as a SIP trunk in SipX I can see that when making a call from SipX to CUCM that there is no attempt from SipX to call CUCM before the call fails. Many Thanks Steve -----Original Message----- From: Joe Micciche [mailto:[email protected]] Sent: 09 September 2011 21:29 To: Stephen Barry Subject: Re: [sipx-users] Cisco Integration for Voicemail -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Steve, I do not have the voicemail setup here that you are trying, but can try to help as we have both CCM and sipX running and calling between them is solid. - - What version of Call Manager are you running? - - How have you configured the SIP trunk in CCM to sipX? - - Can you provide info on the SIP profile you used for the SIP trunk in CCM? - - Did you set up the SIP trunk as a SIP Trunk or unmanaged gateway in sipX? joe On 09/09/2011 01:20 PM, [email protected] wrote: > Message: 3 Date: Fri, 9 Sep 2011 16:46:51 +0000 From: Stephen Barry > <[email protected]> Subject: Re: [sipx-users] Cisco > Integration for Voicemail To: Discussion list for users of sipXecs > software <[email protected]> Message-ID: > <3FA4F8C69ECF85478AA5FD2A88F8CF5006DCFB46@ln5-hex1mb-01.hex1.sensical. > net> > > Content-Type: text/plain; charset="us-ascii" > > Bump. > > Anyone able to help? Is there anyone I can turn to for paid support > even? > > Thanks Steve > > -----Original Message----- From: > [email protected] > [mailto:[email protected]] On Behalf Of Stephen > Barry Sent: 05 September 2011 22:46 To: Discussion list for users of > sipXecs software Subject: Re: [sipx-users] Cisco Integration for > Voicemail > > Hi Tony. > > My x-lite client is configured as follows: > > User ID: 2311 Domain: voip.system.ads (also tried hostname and IP > here) Password: xxxxxxxx Authorization Name: 2311 > > I have the following DNS records > > _sip._udp.voip.system.ads SRV sip1.voip.system.ads > sip1.voip.system.ads A 192.168.41.91 > > My SIP trunk to CCM is not configured for registration as CCM does not > support this. > > I have the following Dial Plan entry (if this is relevant/helps at > all) > > Name: Test > > Dialed Number: Prefix 53 and Two Digits Prefix 54 and Two Digits > Prefix 230 and One Digit > > Require Permissions All Unticked, but I have also tried creating a new > permission category and ensured my Xlite user has access to it, this > made no difference. > > Resulting Call: Dial 'Blank' and Append entire dialled number. > > Schedule: Always > > Gateways CCM Gateway > > Many Thanks Steve > > > From: [email protected] > [mailto:[email protected]] On Behalf Of Tony > Graziano Sent: 05 September 2011 20:52 To: Discussion list for users > of sipXecs software Subject: Re: [sipx-users] Cisco Integration for > Voicemail > > are you registering by hostname, ip or sipdomain? On Sep 5, 2011 > 3:24 PM, "Stephen Barry" <[email protected]> wrote: >>> Hello >>> >>> I am trying to use Sipx for voicemail on Cisco Call Manager, I am >>> keen to exploit IMAP integration. >>> >>> I am able to leave myself a voicemail. However MWI is not working. >>> >>> My test extension is 5397 which is an extension on Cisco Call >>> Manager. I have defined this as a user on SipX to enable VM. >>> >>> I have also defined another extension 2311 and I using x-lite I can >>> register and dial the voicemail pilot 2310 fine. However I cannot >>> dial anything on the call manager despite having a dial plan entries >>> for 53XX and 54XX that point to the SIP trunk to CCM, capturing >>> traffic I always get a proxy auth required error returned to x-lite >>> when dialling these extensions. >>> >>> I am thinking this may be related to my MWI issue, either way I need >>> to be able to dial both ways. >>> >>> I have seen similar reported problems in the list but I cannot work >>> it out. I appreciate that my understanding of dial plans and infact >>> sipx in general may be lacking. >>> >>> I am happy to pay for someone to explain/assist is there anyone able >>> to help me? >>> >>> Many thanks Steve - -- ================================================================== Joe Micciche [email protected] Red Hat, Inc. http://www.redhat.com Senior Communications Engineer X (81) 44554 +1.919.754.4554 Key: 65F90FE1 ================================================================== -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk5qdxsACgkQJHjEUGX5D+HgSgCdEUgTfUUoBj08bUiQIrkCQ4t/ SYoAnRQBdxopkLzQb5vkv6h3K+beZ+Ww =hDYN -----END PGP SIGNATURE----- _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
