It's both sides with this problem. The polycom 321s behind the server are pulling programming/provisioning from sipxecs (dhcp and tftp). The grandstream I have here at home I've configured manually. Again, I can successfully make calls to/from all extensions, audio is working bidirectionally, and I see registrations for all of them. The hostname (sipx.hmherbs.com) will probably be a full subdomain soon but at the moment I've just got it set on a A record and the clients pointing to sipxecs for DNS. Under System -> Domain, the domain name is set to "sipx.hmherbs.com", with an alias for voice.hmherbs.com and hmherbs.com set, depending on what the customer wants to use when we're finished. Under System -> Internet Calling, I've got 192.168.1.0/24, 192.168.2.0/24, and 10.1.10.0/24 as Intranet subnets and *.hmherbs.com as Intranet domains.
I'm thinking it has to do with the " OsSocket::write() returned -1, errno = 32 " and broken pipe messages... but I didn't know for sure what's causing it. Thanks, Matt From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: Saturday, June 16, 2012 5:51 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout its a grandstream having the problem or both? Can you tell me how the phones were programmed? I think I see the hostname being called "instead of" the domain name. Why would that be? Did you properly populate the intranet subnets? On Sat, Jun 16, 2012 at 3:07 PM, [email protected]<mailto:[email protected]> <[email protected]<mailto:[email protected]>> wrote: Hi Tony, thanks for the quick response. There are phones in both locations. I'm testing using a Grandstream GXV3000 remotely from home but there are two Polycom 321s also registered on the premises. SRV records are there, you've just got to look in the right spot. The grandstream here is pointed to the DNS there, the Polycoms on site use the sipXecs server for DHCP/DNS and gateway : ~]$ dig @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com> ; <<>> DiG 9.7.3-P3-RedHat-9.7.3-8.P3.el6_2.3 <<>> @70.88.18.153<http://70.88.18.153> SRV _sip._tcp.hmherbs.com<http://tcp.hmherbs.com> ; (1 server found) ;; global options: +cmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 7630 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 1, ADDITIONAL: 1 ;; QUESTION SECTION: ;_sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. IN SRV ;; ANSWER SECTION: _sip._tcp.hmherbs.com<http://tcp.hmherbs.com>. 1800 IN SRV 1 0 5060 sipx.hmherbs.com<http://sipx.hmherbs.com>. ;; AUTHORITY SECTION: hmherbs.com<http://hmherbs.com>. 1800 IN NS sipx.hmherbs.com<http://sipx.hmherbs.com>. ;; ADDITIONAL SECTION: sipx.hmherbs.com<http://sipx.hmherbs.com>. 1800 IN A 70.88.18.153 ;; Query time: 48 msec ;; SERVER: 70.88.18.153#53(70.88.18.153) ;; WHEN: Sat Jun 16 14:52:03 2012 ;; MSG SIZE rcvd: 105 I've attached the siptrace of the call to 101 I attached in the earlier logs. Thanks again, Matt From: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] On Behalf Of Tony Graziano Sent: Saturday, June 16, 2012 11:24 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] voicemail and autoattendant 408 timeout You are not really providing enough information... Is the phone local or remote? If it is remote, did you change any of the phone parameters manually (I ask because the SRV records are missing for the domain name and that will always cause issues)... It would really help if you explained the call flow and provided a siptrace. snippets of logs tell not much... http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer On Sat, Jun 16, 2012 at 10:37 AM, [email protected]<mailto:[email protected]> <[email protected]<mailto:[email protected]>> wrote: Whoops, left the hostname there! Oh well... I also found a bug report about this here : http://track.sipfoundry.org/browse/XX-6675?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel . Per the request of the developer on that page I've attached a INFO level log of it. From: [email protected]<mailto:[email protected]> [mailto:[email protected]<mailto:[email protected]>] On Behalf Of [email protected]<mailto:[email protected]> Sent: Saturday, June 16, 2012 9:39 AM To: [email protected]<mailto:[email protected]> Subject: [sipx-users] voicemail and autoattendant 408 timeout Hello, I'm having the same problem as this user : http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331 I can make calls from extension to extension, however if I let it go to voicemail the system returns a "408 Timeout" to the phone. Calls directly to 0 or 101 also get the fast busy on polycom 321s, the grandstream gxv3000 will return "408 Timeout" on the display screen. The CDRs also show the 408 and a failed status. In sipXproxy.log, I see (I've sensored the IP with xxx.xxx.xxx.xxx) : "2012-06-16T13:18:50.088967Z":904:SIP:WARNING:sipx.hmherbs.com:SipRouter-15:42305940:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction" "2012-06-16T13:18:50.093375Z":905:KERNEL:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"OsSocket::write 22 (xxx.xxx.xxx.xxx:15060 :-1) send returned -1, errno=32 'Broken pipe'" "2012-06-16T13:18:50.093442Z":906:SIP:ERR:sipx.hmherbs.com:SipClientTcp-196:40F5F940:SipXProxy:"SipClientWriteBuffer[SipClientTcp-196]::writeMore OsSocket::write() returned -1, errno = 32" "2012-06-16T13:18:50.133730Z":907:SIP:ERR:sipx.hmherbs.com:SipUserAgent-2:42204940:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction" Thanks, Matt _______________________________________________ sipx-users mailing list [email protected]<mailto:[email protected]> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected]<mailto:[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! ~~~~~~~~~~~~~~~~~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected]<mailto:[email protected]> Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net _______________________________________________ sipx-users mailing list [email protected]<mailto:[email protected]> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected]<mailto:[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! ~~~~~~~~~~~~~~~~~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected]<mailto:[email protected]> Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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