For some reason our replies seem to get stuck somewhere. Here is another try.
We did numerous test and the issue boils down to NAT. - Flowroute (our SIP Service Provider) provides NAT and it cannot be turned off. - Incoming calls work until they are transferred for a second time e.g. Attendant -> Hunt Group -> Other Attendant -> Mailbox - sipX uses a re-invite for the second transfer which is not supported by Flowroute. - if we move the sipX outside the internal network everything works as intended. - we also tried a second sipX outside the network but were not able to route incoming calls to the internal sipX - we tried several routers and it is not a firewall issue. - we also tried using two NICs, one with public IP and one with internal IP. But there registration issues and it is not officially supported. - Flowroute suggested to set the contact header from the internal IP to the public IP. Not sure how to do that??? There are two reasons why we would like to keep the sipX internal: - auto provisioning of the phones works only if sipX is internal - we are much more vulnerable to attacks if the sipX is external Does anybody have any suggestions on how to get around this NAT issue. Any help is greatly appreciated. Thanks, Marcello -----Original Message----- From: Marcel Manzardo [mailto:[email protected]] Sent: Friday, June 24, 2011 6:28 AM To: 'Discussion list for users of sipXecs software' Subject: RE: [sipx-users] Incoming external call does not complete successfully Sorry for the delay. Our reply got held by the moderator because the attachment was too big... Here is the rest of the info including the siptrace: - Our firewall type is NATed with outbound ports set to static so no port manipulation. - Additional info about the failed call: The call is a one way/no audio call. Incoming external caller still has an active call until caller hangs up. Any help is greatly appreciated Thanks Marcello -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Marcel Manzardo Sent: Tuesday, June 21, 2011 3:24 PM To: 'Discussion list for users of sipXecs software' Subject: Re: [sipx-users] Incoming external call does not complete successfully Thank you very much for all the feedback. We will provide a SIP trace as soon as possible. Here is some additional info: - The only difference between a failed call flow and a successful call flow is that in the case of the successful call, the caller is an internal user calling the autoattendant. In the failed call an incoming "external" call is directed to the same autoattendant via the DID and the call fails after the second leg. - We are using SIP trunks from Flowroute. - We are connected via a pfSense firewall with QoS enabled; we have a static IP (I will be able to provide more info shortly) Thanks Marcello -----Original Message----- From: Tony Graziano [mailto:[email protected]] Sent: Tuesday, June 21, 2011 1:29 PM To: Discussion list for users of sipXecs software Cc: [email protected] Subject: Re: [sipx-users] Incoming external call does not complete successfully that pcap is useless. it shows an ip "248" and another "222", there are only two packets and there is a registration attempt. You can't tell if it is successful, and none of the packets are from sipx itself. What we all would like to see if a siptrace as George pointed out. In the interim, please make sure you describe your call flow according to the failed call. I don't think an internal "successful" call would be much help here. Also, please describe your gateway (type and schema). I.e. Grandstream HT386 FXO, Audicocodes, Patton, etc. or siptrunk through provider "abc" with firewall configured as "type"... On Tue, Jun 21, 2011 at 4:21 PM, George Niculae <[email protected]> wrote: > > > On Tue, Jun 21, 2011 at 10:50 PM, Marcel Manzardo > <[email protected]> > wrote: >> >> We are running SIPx ver 4.4. Internal calls follow call flow >> successfully but external call does not. We might be missing a >> parameter? Here are the call flows: >> >> Internal Call Scenario: >> >> 1. USER Extension 401(BEN) calls General Auto Attendant(x380) 2. >> General Auto Attendant picks up call and USER chooses option 2(Sales >> Hunt Group x 352) 3. Sales Hunt Group rings extensions 400,401,404 4. >> No extensions answers and Sales Hunt Group and falls back to >> extension 382(Sales Auto Attendant) 5. Sales Auto Attendant answers >> call and user chooses option 4(General Mailbox x393) 6. Call is >> successful USER is able to leave voicemail >> >> External Call Scenario: >> 1. External caller calls General Auto Attendant(DID 18888850205 >> ext380) 2. General Auto Attendant picks up call and USER chooses >> option 2(Sales Hunt Group x 352) 3. Sales Hunt Group rings internal >> extensions 400,401,404 4. No internal extensions answers and Sales >> Hunt Group and falls back to extension 382(Sales Auto Attendant) 5. >> Sales Auto Attendant answers call and external caller chooses option >> 4(General Mailboxx393) 6. Call is Unsuccessful and External caller is >> still connected but dead air. >> >> tcpdumps are attached. > > Can you provide also a snapshot or a call flow to be analyzed with > sipviewer (http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+S ipviewer)? > George > _______________________________________________ > 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 _______________________________________________ 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/
