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]<mailto:[email protected]>"
<[email protected]<mailto:[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<http://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<http://sipx.hmherbs.com>", with an alias for
voice.hmherbs.com<http://voice.hmherbs.com> and hmherbs.com<http://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<http://192.168.1.0/24>,
192.168.2.0/24<http://192.168.2.0/24>, and 10.1.10.0/24<http://10.1.10.0/24> as
Intranet subnets and *.hmherbs.com<http://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]>
[mailto:[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
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~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip:
[email protected]<mailto:[email protected]>
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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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
_______________________________________________
sipx-users mailing list
[email protected]<mailto:[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]<mailto:[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/
