If it were me I would start over with a sub domain if that is your ultimate
goal. It sounds as *though* you have inconsistencies in your setup and will
have issues until you rebuild it.
On Jun 17, 2012 9:53 AM, "[email protected]" <[email protected]> wrote:
> Fixed now. I just ran through sipxecs-setup again, set the domain for
> sipx.hmherbs.com, then ran the freeswitch.sh --configtest and --config.
> It's all good now but the voicemail/operator calls still fail.****
>
> ** **
>
> [root@sipx ~]# sipxproc --state****
>
> {"PageServer"=>"Running",****
>
> "PresenceServer"=>"Disabled",****
>
> "ConfigAgent"=>"Disabled",****
>
> "sipXrecording"=>"Running",****
>
> "SharedAppearanceAgent"=>"Running",****
>
> "sipXivr"=>"Running",****
>
> "SIPStatus"=>"Running",****
>
> "ACDServer"=>"Disabled",****
>
> "sipXmrtg"=>"Running",****
>
> "ParkServer"=>"Running",****
>
> "SipXopenfire"=>"Running",****
>
> "CallResolver"=>"Running",****
>
> "sipXacccode"=>"Running",****
>
> "sipXimbot"=>"Running",****
>
> "SipXrest"=>"Running",****
>
> "ResourceListServer"=>"Running",****
>
> "SIPRegistrar"=>"Running",****
>
> "SIPXProxy"=>"Running",****
>
> "sipXprovision"=>"Running",****
>
> "SipXrelay"=>"Running",****
>
> "CallResolver-Agent"=>"Disabled",****
>
> "SipXbridge"=>"Running",****
>
> "FreeSWITCH"=>"Running",****
>
> "ConfigServer"=>"Running"}****
>
> ** **
>
> ** **
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Tony Graziano
> *Sent:* Sunday, June 17, 2012 9:30 AM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
>
> ** **
>
> Your freeswitch connection test failed. Freeswicth s the media engine for
> voicemail and auto attendant. Put the log level to debug for media server
> and restart it and then place a failed call and look at the freswitch logs
> for why!****
>
> On Jun 17, 2012 9:07 AM, "[email protected]" <[email protected]> wrote:***
> *
>
> Yes it's every call to voicemail (101), operator (0), or if I call an
> extension and let it go to voicemail. As you've guessed already this is the
> initial dev/test installation, so if it would help our troubleshooting I'd
> be happy to set up an extension for you. The SIP trunks haven't been
> terminated to it yet so it could only be strictly extension to extension
> dialing. I couldn't remember if I had sent the server profiles so I've just
> done that, added intranet domain alias *.sipx.hmherbs.com, *.
> voice.hmherbs.com, and restarted the box. Here's the output you asked for
> :****
>
> ****
>
> [root@sipx sipxpbx]# sipxproc --state****
>
> {"FreeSWITCH"=>"ConfigurationTestFailed",****
>
> "ConfigServer"=>"Running",****
>
> "SipXbridge"=>"Running",****
>
> "ConfigAgent"=>"Disabled",****
>
> "sipXprovision"=>"Running",****
>
> "SipXopenfire"=>"Running",****
>
> "PresenceServer"=>"Disabled",****
>
> "SipXrelay"=>"Running",****
>
> "sipXimbot"=>"Running",****
>
> "sipXmrtg"=>"Running",****
>
> "SIPRegistrar"=>"Running",****
>
> "CallResolver-Agent"=>"Disabled",****
>
> "ACDServer"=>"Disabled",****
>
> "sipXivr"=>"Running",****
>
> "PageServer"=>"Running",****
>
> "sipXacccode"=>"Running",****
>
> "ParkServer"=>"Running",****
>
> "ResourceListServer"=>"Running",****
>
> "SharedAppearanceAgent"=>"Running",****
>
> "SIPStatus"=>"Running",****
>
> "sipXrecording"=>"Running",****
>
> "CallResolver"=>"Running",****
>
> "SipXrest"=>"Running",****
>
> "SIPXProxy"=>"Running"}****
>
> ****
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Tony Graziano
> *Sent:* Sunday, June 17, 2012 6:12 AM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] voicemail and autoattendant 408 timeout****
>
> ****
>
> What is the output of
> sipxproc --state
> Does this happen on every call to the voicemail? When you set the system
> up did you send the server its profile?****
>
> If it were me I'd kick the grand streams to the curb. Since you have them
> you might be better off to remove the other codecs except g711 from them. I
> still think something is wring with the domain aspect of it.****
>
> Ensure you have a domain alias of the "hostname" and restart services as
> prompted. When you decide to do a sub domain in production, wipe and
> rebuild the system from bare metal, don't try to reconfigure it.****
>
> Also, check top to make sure the system is not using swap.****
>
> On Jun 16, 2012 9:22 PM, "[email protected]" <[email protected]> wrote:***
> *
>
> 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.comset,
> 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] <[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 SRV _sip._tcp.hmherbs.com****
>
> ****
>
> ; <<>> DiG 9.7.3-P3-RedHat-9.7.3-8.P3.el6_2.3 <<>> @70.88.18.153 SRV
> _sip._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. IN SRV****
>
> ****
>
> ;; ANSWER SECTION:****
>
> _sip._tcp.hmherbs.com. 1800 IN SRV 1 0 5060 sipx.hmherbs.com.
> ****
>
> ****
>
> ;; AUTHORITY SECTION:****
>
> hmherbs.com. 1800 IN NS sipx.hmherbs.com.****
>
> ****
>
> ;; ADDITIONAL SECTION:****
>
> 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]] *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] <[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]] *On Behalf Of *[email protected]
> *Sent:* Saturday, June 16, 2012 9:39 AM
> *To:* [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]
> List Archive: http://list.sipfoundry.org/archive/sipx-users/****
>
>
>
> ****
>
> ****
>
> --
> ~~~~~~~~~~~~~~~~~~
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: [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]****
>
> ****
>
> Helpdesk Customers: http://myhelp.myitdepartment.net****
>
> Blog: http://blog.myitdepartment.net****
>
>
> _______________________________________________
> 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: 434.465.6833
> ~~~~~~~~~~~~~~~~~~
> 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]****
>
> ****
>
> Helpdesk Customers: http://myhelp.myitdepartment.net****
>
> Blog: http://blog.myitdepartment.net****
>
>
> _______________________________________________
> sipx-users mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipx-users/****
>
> ****
>
> LAN/Telephony/Security and Control Systems Helpdesk:****
>
> Telephone: 434.984.8426****
>
> sip: [email protected]****
>
> ****
>
> Helpdesk Customers: http://myhelp.myitdepartment.net****
>
> Blog: http://blog.myitdepartment.net****
>
>
> _______________________________________________
> sipx-users mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipx-users/****
>
> ** **
>
> LAN/Telephony/Security and Control Systems Helpdesk:****
>
> Telephone: 434.984.8426****
>
> sip: [email protected]****
>
> ** **
>
> Helpdesk Customers: http://myhelp.myitdepartment.net****
>
> Blog: http://blog.myitdepartment.net****
>
> _______________________________________________
> sipx-users mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]
Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
