>From what you said, it sounds like I might have a firmware rev that is too new >(3.2.1).
Here are the answers to all your questions anyway... 1. What version of patch sipxbridge is patched to. The latest is "patch20.zip". It might be a good idea to install that and restart siptrunking and sipxconfig. More info than you asked for: [r...@lsvm-sip1 ~]# rpm -qa | grep -i sip sipxcommserverlib-doc-4.0.2-016420 sipx-jasperreports-deps-1.0.0-2 sipxecs-doc-4.0.2-016420 sipxportlib-4.0.2-016420 sipxmediaadapterlib-4.0.2-016420 sipxcalllib-4.0.2-016420 sipx-freeswitch-codec-passthru-amr-1.0.3-1 sipx-freeswitch-codec-passthru-g723_1-1.0.3-1 sipxconfig-tftp-4.0.2-016420 sipxconfig-ftp-4.0.2-016420 sipxtools-4.0.2-016420 sipxproxy-4.0.2-016420 sipxconfig-mrtg-4.0.2-016420 sipxvxml-4.0.2-016420 sipxpublisher-4.0.2-016420 sipxregistry-4.0.2-016420 sipxfreeswitch-4.0.2-016420 sipxivr-4.0.2-016420 sipxconfig-report-4.0.2-016420 sipxtacklib-4.0.2-016420 sipxmedialib-4.0.2-016420 sipxcommons-4.0.2-016420 sipx-freeswitch-1.0.3-1 sipx-freeswitch-codec-passthru-g729-1.0.3-1 sipxecs-release-4.0.2-009184016420 sipxcommserverlib-4.0.2-016420 sipxpbx-4.0.2-016420 sipxconfig-4.0.2-016420 sipxconfig-snmp-4.0.2-016420 sipxproxy-cdr-4.0.2-016420 sipxacd-4.0.2-016420 sipxsupervisor-4.0.2-016420 sipxbridge-4.0.2-016420 sipxconfig-agent-4.0.2-016420 sipxpage-4.0.2-016420 sipxecs-4.0.2-016420 2. Your phone should be at Bootrom 4.2 and Firmware 3.1.3RevC (not 3.2.anything else right now). Bootrom: 4.2.0 Firmware: 3.2.1 3. If you have problems with HOLD/RESUME you need to confirm that the outside caller is getting MOH or not. I always start with an ITSp by removing MOH from the phone (removing the [email protected] from the SIP parameters and sending the profile to the phone) and leaving it on in sipXbridge (MOH is checked). If I have problems after doing this, I look at the ITSP account, because if the domain part is wrong the re-invite to resume the call will fail. If it is working, then I enable it on the phone (send the profile again) and try to see if it works. Not sure how to go about "removing the [email protected] from the SIP parameters). Are you talking about editing the configuration files from the command line? I don't see anywhere to do what you're talking about from the Web Admin page. I did try deleting default.wav under Features à Music on Hold and re-sending profiles to the phones, but that didn't change anything. Who is your ITSp and is there a template for them? If not, can you share the basic settings? Name ConnectMichSIPTrunk Address iax1.lodden.com Location --all-- Shared (checked) Route sipXbridge-1 ITSP Server Domain Name iax1.lodden.com User Name (redacted) Password (redacted) ITSP Server Address iax1.lodden.com Use public address for call setup (checked) Strip private headers (unchecked) Use default asserted identity (unchecked) Register on initialization (checked) ITSP Registrar address (blank) Registration Inteval 600 Method to use for SIP keepalive Empty SIP message Method to use for RTP keepalive None The problem seems to be that when the phone issues a re-INVITE: Time: 2009-11-10T14:43:17.874000Z Frame: 13 /tmp/trace.mtX28792/_.sipxbridge.trace.xml:45 Source: 192.168.9.250:5060 Dest: lsvm-sip1.lynk.com-sipXbridge INVITE sip:[email protected]:5090;x-sipX-nonat;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.9.250;branch=z9hG4bK37802d9C25D3838 From: "Mike Burden" <sip:[email protected]>;tag=59A918DB-A3BC9DBA To: "(redacted)" <sip:(redacted)@voipmich.com>;tag=6708330379926174655 CSeq: 1 INVITE Call-ID: [email protected] Contact: <sip:[email protected];transport=udp>;+sip.rendering="no" Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.1.0054 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Length: 0 The sipXbridge is rejecting it: Time: 2009-11-10T14:43:17.881000Z Frame: 14 /tmp/trace.mtX28792/_.sipxbridge.trace.xml:46 Source: lsvm-sip1.lynk.com-sipXbridge Dest: 192.168.9.250:5060 SIP/2.0 403 Request not issued from SIPX proxy server Via: SIP/2.0/UDP 192.168.9.250;branch=z9hG4bK37802d9C25D3838 From: "Mike Burden" <sip:[email protected]>;tag=59A918DB-A3BC9DBA To: "(redacted)" <sip:(redacted)@voipmich.com>;tag=6708330379926174655 CSeq: 1 INVITE Call-ID: [email protected] Server: sipXecs/4.0.2 sipXecs/sipxbridge (Linux) Contact: <sip:[email protected]:5090> Supported: replaces,100rel Content-Length: 0 From: Tony Graziano [mailto:[email protected]] Sent: Tuesday, November 10, 2009 9:53 AM To: Burden, Mike Subject: Re: [sipx-users] Hold/Resume with Polycom IP550 (was: RE: sipXbridge returning 404) "Hold on there pardner..." Ok, when a call is placed on hold it is important to understand there are several different call hold formats out there. I think it's reasonable to assume if your ITSP supports one (should), it would be to place the call on hold as is, using an ip of 0.0.0.0. sipXbridge negotiates calls hold from the ITSP as 0.0.0.0 and "intercedes" on your phones behalf. The problem you are having might be: 1. What version of patch sipxbridge is patched to. The latest is "patch20.zip". It might be a good idea to install that and restart siptrunking and sipxconfig. 2. Your phone should be at Bootrom 4.2 and Firmware 3.1.3RevC (not 3.2.anything else right now). 3. If you have problems with HOLD/RESUME you need to confirm that the outside caller is getting MOH or not. I always start with an ITSp by removing MOH from the phone (removing the [email protected] from the SIP parameters and sending the profile to the phone) and leaving it on in sipXbridge (MOH is checked). If I have problems after doing this, I look at the ITSP account, because if the domain part is wrong the re-invite to resume the call will fail. If it is working, then I enable it on the phone (send the profile again) and try to see if it works. Who is your ITSp and is there a template for them? If not, can you share the basic settings? On Tue, Nov 10, 2009 at 9:10 AM, Burden, Mike <[email protected]> wrote: I assume that you are talking about 6.5.5 and 6.5.6 of "SIP Trunking with sipXecs: Overview and Configuration"? Yes, that's how I set up the Gateway in the first place. I have a SIP Trunk Gateway set up with sipXbridge-1 as the route and the ITSP account. Inbound and outbound calls are working perfectly. I'm not sure what that has to do with Hold/Resume, though, and I haven't seen anything in the Wiki that would make me think that the Hold/Resume problem has anything to do with the ITSP account? Mike Burden Lynk Systems, Inc e-mail: [email protected] Phone: 616-532-4985 -----Original Message----- From: M. Ranganathan [mailto:[email protected]] Sent: Friday, November 06, 2009 10:35 PM To: Burden, Mike Cc: [email protected] Subject: Re: [sipx-users] Hold/Resume with Polycom IP550 (was: RE: sipXbridge returning 404) On Fri, Nov 6, 2009 at 5:53 PM, Burden, Mike <[email protected]> wrote: > OK, I've got the SIP trunk to my ITSP working (it was a configuration > problem at the ITSP's end) > > > > Now I have a new problem. We are using Polycom IP550 phones. If we put a > call on hold from the IP550, the first softbutton label changes to "Resume", > but hitting the button does not retrieve the call. > > > > Does anyone know how to make this work correctly? > > > You need an ITSP record for sipxbridge to work. The domain part of the INVITE request that is sent to sipxbridge is used to pick the ITSP account. There are step by step instructions on the sipxrbidge wiki page. Did you take a look at those? Regards Ranga. > > > Mike Burden > > Lynk Systems, Inc > > e-mail: [email protected] > > Phone: 616-532-4985 > > > > > > > > From: [email protected] > [mailto:[email protected]] On Behalf Of Burden, Mike > Sent: Friday, November 06, 2009 9:01 AM > To: [email protected] > Subject: [sipx-users] sipXbridge returning 404 (was: RE: Multiple gateways?) > > > > OK, I'm down to one issue... I have one Gateway that is not working > correctly. It doesn't work correctly regardless of whether any other > gateways are configured. > > > > The sipXbridge is returning "SIP/2.0 404 No record of ITSP. Check > configuration", so we're not even getting far enough for the format that the > ITSP expects the phone number to be to come into play. > > > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- M. Ranganathan _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
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