Re: [sipx-users] Call forward fails to external number
It will depend entirely upon the itsp or telco provider. Some itsp's do not actually support hair pinned calls. I ran into an instance recently where the outbound call (forward) was a local call but we had to use a 10 digit number instead of 7 only for the forward. A siptrace would be helpful. -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! On Sep 18, 2012 10:24 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Hello, What must one do in 4.6 to allow a local user to forward a call to an external number? Here is what I have now: User A can dial a 10-digit outside number and it routes out the gateway correctly. User A can include the outside number in his call forwarding configuration, and if User B calls User A, the call forward works correctly. But if User A receives a call from the outside, the outside caller hits User A's voicemail instead of forwarding to the outside number. All other inbound and outbound calling through the gateway seems to work okay. As far as I can tell all my permissions and dial-plans are configured and enabled correctly. sipXproxy.log isn't helping much. What might I check next? - Jeff ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] MP3 creation by IVR on 4.6
On Wed, Sep 19, 2012 at 5:17 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Life is good. A 64 kbps MP3 @ 16 kHz and a 32 kbps MP3 @ 8 kHz are a quarter the size of their WAV equivalents with a quality high enough most folks can't hear any compression artifacts at all. Works for me. Glad it worked. If you feel we should expose these as settings in UI please raise a JIRA and will schedule for next releases George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] new patch for XX-10177
Andrewe any news on the patch i sent you offlist? On 09/13/2012 03:19 AM, andrewpit...@comcast.net wrote: Joegen, We recently upgraded a bunch of our servers to 4.4.0 update #18, and while this issue doesn't occur as frequently, we are still experiencing it. I was going to try running a version that has the original patch (any combination of CR-LFs in a keepalive message) and see if that gets us anywhere, but not being able to produce a hang on demand kind of limits our testing options for this. Not being totally convinced it is the keepalive processing, are there any other places we could look for the root cause of this problem? Thanks, Andy *From: *Joegen Baclor jbac...@ezuce.com *To: *andrewpit...@comcast.net *Cc: *Discussion list for users of sipXecs software sipx-users@list.sipfoundry.org *Sent: *Wednesday, August 22, 2012 5:32:34 PM *Subject: *Re: [sipx-users] new patch for XX-10177 Ooops, typo. Its not PUBLISH but UPDATE. On 08/23/2012 01:58 AM, andrewpit...@comcast.net wrote: Joegen, Actually, it looks like we have the keepalive.sessionTimers option turned off for all our phones anyway. I looked in the pcap we have from a hang just this morning and I didn't see any messages matching the PUBLISH method. I can send you these off list if you'd like. -Andy *From: *Joegen Baclor jbac...@ezuce.com *To: *Discussion list for users of sipXecs software sipx-users@list.sipfoundry.org *Cc: *andrewpit...@comcast.net *Sent: *Tuesday, August 21, 2012 10:16:20 PM *Subject: *Re: [sipx-users] new patch for XX-10177 I've taken a quick look at the NAT traversal plugin and it seems to be not accounting PUBLISH as a potential stream modifier. But it is not clear to me how it would cause a hang. Can you disable session-timers in the phones and see if that somehow lessens the occurence of this incident? On 08/22/2012 01:06 AM, andrewpit...@comcast.net wrote: They're all Polycom phones. Mostly 450s, with a couple of 550s, a 670 and a SoundStation 6000 in the mix. Firmware 3.2.6 with bootrom 4.3.1. -Andy *From: *Tony Graziano tgrazi...@myitdepartment.net *To: *Discussion list for users of sipXecs software sipx-users@list.sipfoundry.org *Sent: *Tuesday, August 21, 2012 1:01:03 PM *Subject: *Re: [sipx-users] new patch for XX-10177 can you elaborate what the UA's are that are involved in that transaction? On Tue, Aug 21, 2012 at 5:59 AM, andrewpitman andrewpit...@comcast.net mailto:andrewpit...@comcast.net wrote: Joegen, George, I noticed some messages in my sipXproxy log from a hang on a customer system just today, which may or may not be pertinent. Besides the parsing errors, I also see messages such as this immediately before the hang: 2012-08-21T15:40:56.031862Z :269589:NAT:WARNING:pbx1.sipdomain.com:SipUserAgent-2:B5EBAB 90:SipXProxy: '83226727-F4CE51E8': Received unexpected event InviteRequest while in state 'WaitingForMediaOffer' 2012-08-21T15:40:56.233883Z :269590:SIP:WARNING:pbx1.sipdomain.com:SipRouter-15:B5DB9B90 :SipXProxy: SipUserAgent::send INVITE request matches existing transaction 2012-08-21T15:40:56.323512Z :269591:SIP:ERR:pbx1.sipdomain.com:SipUserAgent-2:B5EBAB90:S ipXProxy: SipUserAgent::handleMessage SIP message timeout expired with no matching transaction 2012-08-21T15:40:56.414306Z :269592:NAT:WARNING:pbx1.sipdomain.com:SipRouter-15:B5DB9B90 :SipXProxy: '5FHSHt9cQ2XBS':Received unexpected event UpdateRequest while in state 'WaitingForInvite' 2012-08-21T15:40:56.516706Z :269593:NAT:WARNING:pbx1.sipdomain.com:SipUserAgent-2:B5EBAB 90:SipXProxy: '5FHSHt9cQ2XBS': Received unexpected event UpdateRequest while in state 'WaitingForInvite' 2012-08-21T15:40:56.523583Z :269594:NAT:WARNING:pbx1.sipdomain.com:SipClientTcp-4212:B19 FAB90:SipXProxy: '5FHSHt9cQ2XBS': Received unexpected event SuccessfulResponse while in state 'WaitingForInvite' Since this has been happening to some of our customers more than once within a 24 hour period, in order to keep them from cancelling with us we've had to resort to a watchdog script which sends OPTIONS messages to the servers periodically and restarts sipXproxy if it fails to respond. Obviously this is just a
Re: [sipx-users] fax attachments as pdf by default in 4.4
Please disregard this. The behavior is correct (4.4 sends PDF by default, UNLESS the call is not completed, in which case the attachment is a TIFF file, which is inconsequential since the testing was incorrectly reviewed). On Tue, Sep 18, 2012 at 1:54 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: I installed 4.4 and updated it. When I sent a fax it was still tiff. It has all the latest patches from sipfoundry and centos save the latest bind postgres updates which I don't think should matter for this. Reading http://track.sipfoundry.org/browse/XX-9776 tells me it was verified but must have reverted in a later patch. Is there a way to conform this? Do we need a JIRA? -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! http://sipxcolab2013.eventbrite.com/?discount=tony2013 -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! http://sipxcolab2013.eventbrite.com/?discount=tony2013 -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] MP3 creation by IVR on 4.6
On Wed, Sep 19, 2012 at 3:04 AM, George Niculae geo...@ezuce.com wrote: Glad it worked. If you feel we should expose these as settings in UI please raise a JIRA and will schedule for next releases Done. http://track.sipfoundry.org/browse/XTRN-1073 - Jeff ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
what kind of gateway/who is the telco? -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! On Sep 19, 2012 9:50 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Tony, For this testing, no ITSP, just a local PRI gateway. Although we're not that far yet - sipX never sends the INVITE to gateway for the outbound leg of the forward, so no SIP trace. This is day 3 a new install atop Centos 6.3 (not the ISO). Very little fiddling so far. - Jeff On Wed, Sep 19, 2012 at 2:46 AM, Tony Graziano tgrazi...@myitdepartment.net wrote: It will depend entirely upon the itsp or telco provider. Some itsp's do not actually support hair pinned calls. I ran into an instance recently where the outbound call (forward) was a local call but we had to use a 10 digit number instead of 7 only for the forward. A siptrace would be helpful. -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! On Sep 18, 2012 10:24 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Hello, What must one do in 4.6 to allow a local user to forward a call to an external number? Here is what I have now: User A can dial a 10-digit outside number and it routes out the gateway correctly. User A can include the outside number in his call forwarding configuration, and if User B calls User A, the call forward works correctly. But if User A receives a call from the outside, the outside caller hits User A's voicemail instead of forwarding to the outside number. All other inbound and outbound calling through the gateway seems to work okay. As far as I can tell all my permissions and dial-plans are configured and enabled correctly. sipXproxy.log isn't helping much. What might I check next? - Jeff ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
Adtran TA908e. I manage it. CLEC (network) PRI behind it.
The user's extension is set to ring first for 4 seconds, then forward to a
10-digit PSTN number. The PRI on the gateway will accept 7 or 10-digit
local number and the gateway is set to pass through whatever it receives.
I can dial a 7 or 10-digit local number just fine from a local user,
including the one I'm using to test this forwarding. If I call this local
user from another local user, the forward works correctly. Just not if an
outside user calls in.
Here is the tshark outbound of the entire call flow from the perspective of
that gateway, starting with the inbound call from the PRI sent to sipX.
192.168.53.11 is the gateway; 192.168.54.46 is the sipX system.
216000 is not the real DID.
0.00 192.168.53.11 - 192.168.54.46 SIP/SDP Request: INVITE
sip:216...@sipx46.dtcle.fvd.local:5060, with session description
0.002137 192.168.54.46 - 192.168.53.11 SIP Status: 100 Trying
0.141508 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.181202 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.205498 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.300094 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
4.154583 192.168.54.46 - 192.168.53.11 SIP/SDP Status: 200 OK, with
session description
4.171045 192.168.53.11 - 192.168.54.46 SIP Request: ACK
sip:2166821821@192.168.54.46:15060;transport=udp
-{ caller hears VM system }-
7.896108 192.168.53.11 - 192.168.54.46 SIP Request: BYE
sip:2166821821@192.168.54.46:15060;transport=udp
7.909495 192.168.54.46 - 192.168.53.11 SIP Status: 200 OK
There are 4 registered devices on the called user, hence the 4x 180 Ringing
messages.
I would expect to see an INVITE or REFER sent to the gateway at
call-forward time instead of the 200 OK of the VM system. It seems like
something is preventing the system from even trying to send the call.
- Jeff
On Wed, Sep 19, 2012 at 9:56 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
what kind of gateway/who is the telco?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Sep 19, 2012 9:50 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hi Tony,
For this testing, no ITSP, just a local PRI gateway. Although we're not
that far yet - sipX never sends the INVITE to gateway for the outbound leg
of the forward, so no SIP trace.
This is day 3 a new install atop Centos 6.3 (not the ISO). Very little
fiddling so far.
- Jeff
On Wed, Sep 19, 2012 at 2:46 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
It will depend entirely upon the itsp or telco provider.
Some itsp's do not actually support hair pinned calls.
I ran into an instance recently where the outbound call (forward) was a
local call but we had to use a 10 digit number instead of 7 only for the
forward.
A siptrace would be helpful.
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Sep 18, 2012 10:24 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hello,
What must one do in 4.6 to allow a local user to forward a call to an
external number?
Here is what I have now: User A can dial a 10-digit outside number and
it routes out the gateway correctly. User A can include the outside number
in his call forwarding configuration, and if User B calls User A, the call
forward works correctly. But if User A receives a call from the outside,
the outside caller hits User A's voicemail instead of forwarding to the
outside number.
All other inbound and outbound calling through the gateway seems to
work okay.
As far as I can tell all my permissions and dial-plans are configured
and enabled correctly. sipXproxy.log isn't helping much. What might I
check next?
- Jeff
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net
Helpdesk Customers:
http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] MP3 creation by IVR on 4.6
On Wed, Sep 19, 2012 at 4:43 PM, Jeff Pyle jp...@fidelityvoice.com wrote: On Wed, Sep 19, 2012 at 3:04 AM, George Niculae geo...@ezuce.com wrote: Glad it worked. If you feel we should expose these as settings in UI please raise a JIRA and will schedule for next releases Done. http://track.sipfoundry.org/browse/XTRN-1073 Thanks, moved to proper project http://track.sipfoundry.org/browse/XX-10452 (XTRN is for external components) George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
Have you tried removing all the phones but one on that called user to ensure
its not a bad behaving endpoint?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Pyle
Sent: Wednesday, September 19, 2012 7:16 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Call forward fails to external number
Adtran TA908e. I manage it. CLEC (network) PRI behind it.
The user's extension is set to ring first for 4 seconds, then forward to a
10-digit PSTN number. The PRI on the gateway will accept 7 or 10-digit
local number and the gateway is set to pass through whatever it receives. I
can dial a 7 or 10-digit local number just fine from a local user, including
the one I'm using to test this forwarding. If I call this local user from
another local user, the forward works correctly. Just not if an outside
user calls in.
Here is the tshark outbound of the entire call flow from the perspective of
that gateway, starting with the inbound call from the PRI sent to sipX.
192.168.53.11 is the gateway; 192.168.54.46 is the sipX system. 216000
is not the real DID.
0.00 192.168.53.11 - 192.168.54.46 SIP/SDP Request: INVITE
sip:216...@sipx46.dtcle.fvd.local:5060, with session description
0.002137 192.168.54.46 - 192.168.53.11 SIP Status: 100 Trying
0.141508 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.181202 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.205498 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.300094 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
4.154583 192.168.54.46 - 192.168.53.11 SIP/SDP Status: 200 OK, with
session description
4.171045 192.168.53.11 - 192.168.54.46 SIP Request: ACK
sip:2166821821@192.168.54.46:15060;transport=udp
-{ caller hears VM system }-
7.896108 192.168.53.11 - 192.168.54.46 SIP Request: BYE
sip:2166821821@192.168.54.46:15060;transport=udp
7.909495 192.168.54.46 - 192.168.53.11 SIP Status: 200 OK
There are 4 registered devices on the called user, hence the 4x 180 Ringing
messages.
I would expect to see an INVITE or REFER sent to the gateway at call-forward
time instead of the 200 OK of the VM system. It seems like something is
preventing the system from even trying to send the call.
- Jeff
On Wed, Sep 19, 2012 at 9:56 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
what kind of gateway/who is the telco?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Sep 19, 2012 9:50 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hi Tony,
For this testing, no ITSP, just a local PRI gateway. Although we're not
that far yet - sipX never sends the INVITE to gateway for the outbound leg
of the forward, so no SIP trace.
This is day 3 a new install atop Centos 6.3 (not the ISO). Very little
fiddling so far.
- Jeff
On Wed, Sep 19, 2012 at 2:46 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
It will depend entirely upon the itsp or telco provider.
Some itsp's do not actually support hair pinned calls.
I ran into an instance recently where the outbound call (forward) was a
local call but we had to use a 10 digit number instead of 7 only for the
forward.
A siptrace would be helpful.
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Sep 18, 2012 10:24 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hello,
What must one do in 4.6 to allow a local user to forward a call to an
external number?
Here is what I have now: User A can dial a 10-digit outside number and it
routes out the gateway correctly. User A can include the outside number in
his call forwarding configuration, and if User B calls User A, the call
forward works correctly. But if User A receives a call from the outside,
the outside caller hits User A's voicemail instead of forwarding to the
outside number.
All other inbound and outbound calling through the gateway seems to work
okay.
As far as I can tell all my permissions and dial-plans are configured and
enabled correctly. sipXproxy.log isn't helping much. What might I check
next?
- Jeff
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
LAN/Telephony/Security and Control Systems Helpdesk:
Re: [sipx-users] Call forward fails to external number
Scenario is now one Polycom 550 registered to sipX, receiving a call from
the outside. No change in the gateway-to-sipX messaging (call still
answered by VM instead of forwarded). Gateway is .53.11, sipX is .45.46.
0.00 192.168.53.11 - 192.168.54.46 SIP/SDP Request: INVITE
sip:2166821...@sipx46.dtcle.fvd.local:5060, with session description
0.001530 192.168.54.46 - 192.168.53.11 SIP Status: 100 Trying
0.269556 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
4.136557 192.168.54.46 - 192.168.53.11 SIP/SDP Status: 200 OK, with
session description
4.152206 192.168.53.11 - 192.168.54.46 SIP Request: ACK
sip:2166821821@192.168.54.46:15060;transport=udp
6.856499 192.168.53.11 - 192.168.54.46 SIP Request: BYE
sip:2166821821@192.168.54.46:15060;transport=udp
6.864558 192.168.54.46 - 192.168.53.11 SIP Status: 200 OK
The sipX-to-Poly messaging, Poly is .201.60. TCP ACKs removed for clarity.
0.21 192.168.54.46 - 172.21.201.60 SIP/SDP Request: INVITE
sip:1821@172.21.201.60;x-sipX-nonat, with session description
0.064324 172.21.201.60 - 192.168.54.46 SIP Status: 100 Trying
0.230543 172.21.201.60 - 192.168.54.46 SIP Status: 180 Ringing
3.993589 192.168.54.46 - 172.21.201.60 SIP Request: CANCEL
sip:1821@172.21.201.60;x-sipX-nonat
4.045358 172.21.201.60 - 192.168.54.46 SIP Status: 200 OK
4.079853 172.21.201.60 - 192.168.54.46 SIP Status: 487 Request Cancelled
4.080990 192.168.54.46 - 172.21.201.60 SIP Request: ACK
sip:1821@172.21.201.60;x-sipX-nonat
I'm new to sipX, but not to SIP. All this messaging looks appropriate to
me.
The only call forward scenario that doesn't work for me is when an external
caller hits a user set to forward to an external number - the call goes to
the user's VM instead. Every other internal/external caller/callee
combination works as expected.
- Jeff
On Wed, Sep 19, 2012 at 11:59 AM, Todd Hodgen thod...@frontier.com wrote:
Have you tried removing all the phones but one on that called user to
ensure its not a bad behaving endpoint?
** **
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Jeff Pyle
*Sent:* Wednesday, September 19, 2012 7:16 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] Call forward fails to external number
** **
Adtran TA908e. I manage it. CLEC (network) PRI behind it.
** **
The user's extension is set to ring first for 4 seconds, then forward to a
10-digit PSTN number. The PRI on the gateway will accept 7 or 10-digit
local number and the gateway is set to pass through whatever it receives.
I can dial a 7 or 10-digit local number just fine from a local user,
including the one I'm using to test this forwarding. If I call this local
user from another local user, the forward works correctly. Just not if an
outside user calls in.
** **
Here is the tshark outbound of the entire call flow from the perspective
of that gateway, starting with the inbound call from the PRI sent to sipX.
192.168.53.11 is the gateway; 192.168.54.46 is the sipX system.
216000 is not the real DID.
** **
0.00 192.168.53.11 - 192.168.54.46 SIP/SDP Request: INVITE
sip:216...@sipx46.dtcle.fvd.local:5060, with session description
0.002137 192.168.54.46 - 192.168.53.11 SIP Status: 100 Trying
0.141508 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.181202 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.205498 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.300094 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
4.154583 192.168.54.46 - 192.168.53.11 SIP/SDP Status: 200 OK, with
session description
4.171045 192.168.53.11 - 192.168.54.46 SIP Request: ACK
sip:2166821821@192.168.54.46:15060;transport=udp
** **
-{ caller hears VM system }-
** **
7.896108 192.168.53.11 - 192.168.54.46 SIP Request: BYE
sip:2166821821@192.168.54.46:15060;transport=udp
7.909495 192.168.54.46 - 192.168.53.11 SIP Status: 200 OK
** **
There are 4 registered devices on the called user, hence the 4x 180
Ringing messages.
** **
I would expect to see an INVITE or REFER sent to the gateway at
call-forward time instead of the 200 OK of the VM system. It seems like
something is preventing the system from even trying to send the call.
** **
** **
- Jeff
** **
** **
On Wed, Sep 19, 2012 at 9:56 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
what kind of gateway/who is the telco?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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On Sep
Re: [sipx-users] Yealink plugin 4.4
On Wed, Sep 19, 2012 at 10:21 AM, Daniel Peinado Lopez daniel.pein...@iant.de wrote: Hello, I want to try some Yealink Telephones in SipX 4.4. I have downloaded the Plugin in https://github.com/siplabs/sipXyealink and I have it in my CentOS system. The problem is that I don`t know exactly how can I install this. I need to install this plugin, because I want to add these telephones in the SipX interface. Can you tell me how to install step by step? I don`t want to break my system for one stupid error. Thank you very much And SIP Labs... i'd be interested in your interest to package this as rpm for sipxecs 4.6. Get more adoption. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
It is more likely a provider issue. I have seen this also but in my case
the provided wanted 10 digits instead of 7 even though 7 was valid.
I'd call the telco and troubleshoot with them. They are getting the digits
but they need to tell you what is wrong with the digits. really. Call the
telco.
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
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2013!
On Sep 19, 2012 10:15 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Adtran TA908e. I manage it. CLEC (network) PRI behind it.
The user's extension is set to ring first for 4 seconds, then forward to a
10-digit PSTN number. The PRI on the gateway will accept 7 or 10-digit
local number and the gateway is set to pass through whatever it receives.
I can dial a 7 or 10-digit local number just fine from a local user,
including the one I'm using to test this forwarding. If I call this local
user from another local user, the forward works correctly. Just not if an
outside user calls in.
Here is the tshark outbound of the entire call flow from the perspective
of that gateway, starting with the inbound call from the PRI sent to sipX.
192.168.53.11 is the gateway; 192.168.54.46 is the sipX system.
216000 is not the real DID.
0.00 192.168.53.11 - 192.168.54.46 SIP/SDP Request: INVITE
sip:216...@sipx46.dtcle.fvd.local:5060, with session description
0.002137 192.168.54.46 - 192.168.53.11 SIP Status: 100 Trying
0.141508 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.181202 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.205498 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.300094 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
4.154583 192.168.54.46 - 192.168.53.11 SIP/SDP Status: 200 OK, with
session description
4.171045 192.168.53.11 - 192.168.54.46 SIP Request: ACK
sip:2166821821@192.168.54.46:15060;transport=udp
-{ caller hears VM system }-
7.896108 192.168.53.11 - 192.168.54.46 SIP Request: BYE
sip:2166821821@192.168.54.46:15060;transport=udp
7.909495 192.168.54.46 - 192.168.53.11 SIP Status: 200 OK
There are 4 registered devices on the called user, hence the 4x 180
Ringing messages.
I would expect to see an INVITE or REFER sent to the gateway at
call-forward time instead of the 200 OK of the VM system. It seems like
something is preventing the system from even trying to send the call.
- Jeff
On Wed, Sep 19, 2012 at 9:56 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
what kind of gateway/who is the telco?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
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2013!
On Sep 19, 2012 9:50 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hi Tony,
For this testing, no ITSP, just a local PRI gateway. Although we're not
that far yet - sipX never sends the INVITE to gateway for the outbound leg
of the forward, so no SIP trace.
This is day 3 a new install atop Centos 6.3 (not the ISO). Very little
fiddling so far.
- Jeff
On Wed, Sep 19, 2012 at 2:46 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
It will depend entirely upon the itsp or telco provider.
Some itsp's do not actually support hair pinned calls.
I ran into an instance recently where the outbound call (forward) was a
local call but we had to use a 10 digit number instead of 7 only for the
forward.
A siptrace would be helpful.
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about
sipX-CoLab 2013!
On Sep 18, 2012 10:24 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hello,
What must one do in 4.6 to allow a local user to forward a call to an
external number?
Here is what I have now: User A can dial a 10-digit outside number
and it routes out the gateway correctly. User A can include the outside
number in his call forwarding configuration, and if User B calls User A,
the call forward works correctly. But if User A receives a call from the
outside, the outside caller hits User A's voicemail instead of forwarding
to the outside number.
All other inbound and outbound calling through the gateway seems to
work okay.
As far as I can tell all my permissions and
Re: [sipx-users] Call forward fails to external number
Hi Tony,
In the case of an internal caller, all is well. The called-user's call
forward definitions work as one would expect. SipX sends the 7 or 10 digit
INVITE to the gateway when it should, and the telco handles the call. I
happen to be using 10 at the moment.
In the case of an external caller, sipX does not send the INVITE to the
gateway. Instead, that step in the call forward seems to be skipped and
the call goes to the user's VM. That's not telco's fault.
SipX does not send the INVITE for the outbound leg of the call forward to
the gateway if the caller is external. Any thoughts on what might cause
that behavior within sipX? The Forwarding and Transferring Wiki
pagehttp://wiki.sipfoundry.org/display/sipXecs/Forwarding+and+Transferring
mentions,
If you can call [an external] number from your SIP phone, you should be
able to forward or transfer to it ... enter the forwarding number in
exactly the same way in the forwarding target box. I can indeed call the
number (with 7 or 10 digits) but the forward doesn't seem to be working.
No on-phone forwarding is used (signaling verified in a previous post).
Something else seems to be going on.
The How to configure User Call Forwarding
pagehttp://wiki.sipfoundry.org/display/sipXecs/How+to+configure+User+Call+Forwardingmentions
a user and group permission named Forward Calls External that
needs to be enabled. But this seems to be old info, as XX-422 (redirected
from XECS-200) removed this requirement in 2007 (ver 3.10.0).
So by all indications this should just work. But it doesn't. Hmm. Time
to find a bigger hammer.
- Jeff
On Wed, Sep 19, 2012 at 2:07 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
It is more likely a provider issue. I have seen this also but in my case
the provided wanted 10 digits instead of 7 even though 7 was valid.
I'd call the telco and troubleshoot with them. They are getting the digits
but they need to tell you what is wrong with the digits. really. Call the
telco.
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Sep 19, 2012 10:15 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Adtran TA908e. I manage it. CLEC (network) PRI behind it.
The user's extension is set to ring first for 4 seconds, then forward to
a 10-digit PSTN number. The PRI on the gateway will accept 7 or 10-digit
local number and the gateway is set to pass through whatever it receives.
I can dial a 7 or 10-digit local number just fine from a local user,
including the one I'm using to test this forwarding. If I call this local
user from another local user, the forward works correctly. Just not if an
outside user calls in.
Here is the tshark outbound of the entire call flow from the perspective
of that gateway, starting with the inbound call from the PRI sent to sipX.
192.168.53.11 is the gateway; 192.168.54.46 is the sipX system.
216000 is not the real DID.
0.00 192.168.53.11 - 192.168.54.46 SIP/SDP Request: INVITE
sip:216...@sipx46.dtcle.fvd.local:5060, with session description
0.002137 192.168.54.46 - 192.168.53.11 SIP Status: 100 Trying
0.141508 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.181202 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.205498 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
0.300094 192.168.54.46 - 192.168.53.11 SIP Status: 180 Ringing
4.154583 192.168.54.46 - 192.168.53.11 SIP/SDP Status: 200 OK, with
session description
4.171045 192.168.53.11 - 192.168.54.46 SIP Request: ACK
sip:2166821821@192.168.54.46:15060;transport=udp
-{ caller hears VM system }-
7.896108 192.168.53.11 - 192.168.54.46 SIP Request: BYE
sip:2166821821@192.168.54.46:15060;transport=udp
7.909495 192.168.54.46 - 192.168.53.11 SIP Status: 200 OK
There are 4 registered devices on the called user, hence the 4x 180
Ringing messages.
I would expect to see an INVITE or REFER sent to the gateway at
call-forward time instead of the 200 OK of the VM system. It seems like
something is preventing the system from even trying to send the call.
- Jeff
On Wed, Sep 19, 2012 at 9:56 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
what kind of gateway/who is the telco?
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!
On Sep 19, 2012 9:50 AM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hi Tony,
For this testing,
Re: [sipx-users] Call forward fails to external number
Hi Jeff, Just to ensure I understand the scenario: • Calling party originates via PRI • Calling party is routed to called party. • Called party is unavailable, rule is set to forward to 10 digit destination number via PRI • Calling party actually arrives at Free Switch VM box for called party. • The ISDN-PRI gateway is an ADTRAN 908e First, could you please share what version of AOS your TA is running? Second, how many PRIs are actually terminating on the 908e? Best, Mark ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
Hi Mark, Your understand is spot on. A4.11. One PRI to a Lucent 5ESS CO. It's a lab setup so there is next to no traffic. Not that it's related to this case but the 908e does handle REFERs correctly from sipX when necessary. That got crossed off the list yesterday. Back to the sipX box. Capturing traffic on localhost 5060 I found the INVITE that represented the outbound leg of the call. It gets shot down with a 403 Requires by Server: sipXecs/4.6.0 sipXecs/sipXproxy (Linux), never making it to the gateway. - Jeff On Wed, Sep 19, 2012 at 5:02 PM, Mark A. Smith masm...@bcslive.biz wrote: Hi Jeff, Just to ensure I understand the scenario: • Calling party originates via PRI • Calling party is routed to called party. • Called party is unavailable, rule is set to forward to 10 digit destination number via PRI • Calling party actually arrives at Free Switch VM box for called party. • The ISDN-PRI gateway is an ADTRAN 908e First, could you please share what version of AOS your TA is running? Second, how many PRIs are actually terminating on the 908e? Best, Mark ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
Hi Jeff, Thanks for the reply. My knee jerk reaction when I first saw your post is that you may have had multiple PRIs across multiple TDM groups. I'm glad you've dug into this a bit. By default, inter B-channel transfer (hair pinning) is not allowed via AOS switchboard. However, I was running on a number of assumptions. Best, Mark ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
On Thu, Sep 20, 2012 at 12:34 AM, Mark A. Smith masm...@bcslive.biz wrote: Hi Jeff, Thanks for the reply. My knee jerk reaction when I first saw your post is that you may have had multiple PRIs across multiple TDM groups. I'm glad you've dug into this a bit. By default, inter B-channel transfer (hair pinning) is not allowed via AOS switchboard. However, I was running on a number of assumptions. Best, Mark Looks like time to recreate it with proxy and registrar on debug and take log snapshot (Diagnostics Snapshot) George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
On Thu, Sep 20, 2012 at 12:47 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Mark, thanks anyway. I appreciate the replies. We haven't gotten far enough to try TBCT yet! I like you're thinking, though. George, I think I'm onto something. I've recreated this scenario with two local users, 1821 and 4821. 1821 is set to forward to 2169311212 after 4 seconds. 4821 calls 1821. 1821 doesn't answer, and the call forwards as expected. But, if I remove 4821's Local Dialing privilege, 1821 is no longer able to forward 4821 to the PSTN. Straight to voicemail - just like my external testing. It appears sipXproxy is using the privileges of the original caller rather than those of the one doing the forwarding. There is some discussion about this in XX-422. I'm new enough to sipX this is the first time I've attempted a snapshot. Seems simple enough. Is it just as simple to tail the last 1000 lines of sipXproxy.log and sipregistrar.log? Regardless of the approach, do I reply with them as an attachment, or is there another preference? It's not about tailing logs only but it also captures other config from your system. However in your case tailing proxy and reg logsand post them back here would suffice. George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Call forward fails to external number
This is a long standing issue and is all about 302 redirects not able to grant permissions of the called number to the caller. On top of this, branches will also be enforced if you have set one. The caller will not inherent the branch where the callee is located. On 09/20/2012 10:04 AM, Jeff Pyle wrote: On Wed, Sep 19, 2012 at 5:56 PM, George Niculae geo...@ezuce.com mailto:geo...@ezuce.com wrote: It's not about tailing logs only but it also captures other config from your system. However in your case tailing proxy and reg logsand post them back here would suffice. The proxy and registrar logs are attached. User 1821 is set to ring for 4 seconds, then forward to 2169311212. User 1821 can call 2169311212 with no problem (has Local permission). In this test, User 4821 called user 1821, but since User 4821 does not have the Local permission, the call went to 1821's VM box instead of to 2169311212. If I re-enable Local permission for 4821, the call forward on 1821 succeeds. - Jeff ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
