Re: [OpenSIPS-Users] how do you determine first registration of a device

2009-09-26 Thread Uwe Kastens
Hi Alex,
 
 Can anyone provide advice on how to determine first registration of a
 phone coming back online in another way?
 

Depends on the phone I would say. From my point of view I would say,
that is very hard to track, since some phones acts very strange in that
case. Maybe short expire would help?

Can you tell more about your setup?

BR

Uwe

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Re: [OpenSIPS-Users] howto = mediaproxy on lenny

2009-09-26 Thread Uwe Kastens
Hi Euge,

That might be an option in a testing environment. I don't like the idea
to upgrade production systems to debian unstable To much pkg are
changing in short period.

BR

Uwe




Euge Serrano schrieb:
 Hello Uwe,
 
 I recommend to upgrade your Lenny to Squeeze, after that you will be able to 
 install it without problems
 
 You can follow those easy steps
 
 http://www.go2linux.org/how-to-upgrade-from-debian-lenny-to-squeeze
 
 Euge
 
 
 
 On 25/09/09 17:31, Uwe Kastens ki...@kiste.org wrote:
 
 Hello,
 
 I was wondering why I was able to build mediaproxy packages on debian
 lenny (stable) but been unable to install them.
 
 It looks like, one needs only to build the python-application in the
 correct version as dpk from source.
 
 BR
 
 Uwe
 
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Re: [OpenSIPS-Users] how do you determine first registration of a device

2009-09-26 Thread Alex G
i have a 60 second expire on my location table, but some phones still will
reboot in less time than that. especially soft phones like xlite. i need to
know when this first registration is in order to trigger a script that send
mwi notification if that device has new messages. none of my devices
subscribe for it, therefore i have to send notifications to them.

On Sat, Sep 26, 2009 at 3:44 AM, Uwe Kastens ki...@kiste.org wrote:

 Hi Alex,
 
  Can anyone provide advice on how to determine first registration of a
  phone coming back online in another way?
 

 Depends on the phone I would say. From my point of view I would say,
 that is very hard to track, since some phones acts very strange in that
 case. Maybe short expire would help?

 Can you tell more about your setup?

 BR

 Uwe

 --

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[OpenSIPS-Users] OpenSIPS presence server error [presence:get_stored_info:]

2009-09-26 Thread Jawad hussain
Hello,

I am using OpenSIPS presence server. Server starts well and handle
subscriptions and notifications but after some time its give following
errors. Can some one helps in solving this ..

Sep 26 17:06:04 [13800] INFO:presence:send_notify_request: NOTIFY
sip:al...@open-ims.test via sip:m...@scscf.open-ims.test:6060;lr on behalf of
sip:b...@open-ims.test for event presence
Sep 26 17:07:04 [13803] ERROR:presence:get_stored_info: record not found in
hash_table
Sep 26 17:07:04 [13803] ERROR:presence:handle_subscribe: getting stored info
Sep 26 17:07:04 [13803] ERROR:presence:get_stored_info: record not found in
hash_table
Sep 26 17:07:04 [13803] ERROR:presence:handle_subscribe: getting stored info
Sep 26 17:08:32 [13808] INFO:presence:send_notify_request: NOTIFY
sip:b...@open-ims.test via sip:m...@scscf.open-ims.test:6060;lr on behalf of
sip:al...@open-ims.test for event presence

OPENSIPS cfg


#
# $Id: opensips.cfg 5503 2009-03-22 16:22:32Z bogdan_iancu $
#
# OpenSIPS basic configuration script
# by Anca Vamanu a...@voice-system.ro
#
# Please refer to the Core CookBook at:
#  http://www.opensips.org/index.php?n=Resources.DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


### Global Parameters #

debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
log_stderror=yes

/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#opensipsdbctl create
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* uncomment the next line to disable the auto discovery of local aliases
   based on revers DNS on IPs (default on) */
#auto_aliases=no

/* uncomment the following lines to enable TLS support  (default off) */
#disable_tls = no
#listen = tls:your_IP:5061
#tls_verify_server = 1
#tls_verify_client = 1
#tls_require_client_certificate = 0
#tls_method = TLSv1
#tls_certificate = /usr/local/etc/opensips/tls/user/user-cert.pem
#tls_private_key = /usr/local/etc/opensips/tls/user/user-privkey.pem
#tls_ca_list = /usr/local/etc/opensips/tls/user/user-calist.pem


port=5065

/* uncomment and configure the following line if you want opensips to
   bind on a specific interface/port/proto (default bind on all available)
*/
listen=udp:127.0.0.1:5065


### Modules Section 

#set module pathopensipsdbctl create
mpath=/usr/local/lib/opensips/modules/

/* uncomment next line for MySQL DB support */
loadmodule db_mysql.so
loadmodule signaling.so
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
loadmodule registrar.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri_db.so
loadmodule uri.so
loadmodule xlog.so
loadmodule acc.so
/* uncomment next lines for MySQL based authentication support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule auth.so
loadmodule auth_db.so
/* uncomment next line for aliases support
   NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule alias_db.soopensipsdbctl create
/* uncomment next line for multi-domain support
   NOTE: a DB (like db_mysql) module must be also loaded
   NOTE: be sure and enable multi-domain support in all used modules
 (see multi-module params section ) */
#loadmodule domain.so
/* uncomment the next two lines for presence server support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule presence.so
loadmodule presence_xml.so
loadmodule presence_mwi.so
loadmodule pua.so


# - setting module-specific parameters ---


# - mi_fifo params -
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)


# - rr params -
# add value to ;lr param to cope with most of the UAs
modparam(rr, enable_full_lr, 1)
# do not append from tag to the RR (no need for this script)
modparam(rr, append_fromtag, 0)


# - registrar params -
modparam(registrar, method_filtering, 1)
/* uncomment the next line to disable parallel forking via location */
# modparam(registrar, append_branches, 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam(registrar, max_contacts, 10)


# - usrloc params -
#modparam(usrloc, db_mode,   0)
/* uncomment the following lines if you want to enable DB persistency
   for location entries */
modparam(usrloc, db_mode,   2)
modparam(usrloc, db_url,
mysql://opensips:opensip...@localhost/opensips)


# - uri_db params -
/* by default we disable the DB support in the module as we do not need it
   in this configuration */
#modparam(uri_db, use_uri_table, 0)
#modparam(uri_db, db_url, )


# - acc params -
/* what sepcial events should be accounted ? */

Re: [OpenSIPS-Users] Load Balance Module URI Limit

2009-09-26 Thread Bogdan-Andrei Iancu
Hi Russell,

I tried to reproduce this report (have 35 records in the same group, 
with one resource), but with no success. The load_balance() function 
returns true.

Let's try the followings:
1) set full debug (debug=4) and send me the logs for the balanced 
request.
2) if you can sent me the DB content you use and the LB script 
command, it will also help.

Regards,
Bogdan


Russell Bierschbach wrote:
 1.6

 version: opensips 1.6.0dev0-notls (x86_64/linux)

 -Original Message-
 From: users-boun...@lists.opensips.org
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Friday, September 25, 2009 10:39 AM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Load Balance Module URI Limit

 OK - I will try to reproduce it. BTW, trunk or 1.5 ?

 Regards,
 Bogdan

 Russell Bierschbach wrote:
   
 The load_balance() call returns false when there are 34 or more records in
 the table.

 -Original Message-
 From: users-boun...@lists.opensips.org
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Friday, September 25, 2009 2:18 AM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] Load Balance Module URI Limit

 Hi Russell,

 So, if there are more than 34 records in DB, the LB will fails? by 
 failing what you mean ? it does not start or load_balance() returns 
 false all the time?

 Regards,
 Bogdan

 Russell Bierschbach wrote:
   
 
 The load_balance() call always fails if there 34 or more entries in 
 the load_balancer table.  I have confirmed that it's not a problem 
 with the entries themselves (I've tried other entries that all work 
 when there are 33 or less records).  I have also changed the shared 
 memory from the default of 32 to 64, but that didn't seem to help.  Is 
 there some other configuration I'm missing, or is this a limitation of 
 the load balancer module... is there any place I can change it?

  

 version: opensips 1.6.0dev0-notls (x86_64/linux)

  

 Russ

 

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Re: [OpenSIPS-Users] how do you determine first registration of a device

2009-09-26 Thread Alex Balashov
As Inaki said, what you're trying to achieve is a little ridiculous.

But, if you must do it, I recommend using a database capable of 
advanced triggers and decent stored procedures (PostgreSQL) and using 
usrloc and registrar with full database backing and no caching or 
synchronisation (changes immediately reflected in database).  Then put 
a trigger on your 'location' table.  Trying to do this from the route 
script is silly.

Alex G wrote:

 I have a scenario where I'm trying to fire a script based on the first 
 registration of a device. In normal circumstances I can check the 
 location table using registered() function and if there is an entry i 
 know it's not the first registration. My problem is when a phone does a 
 quick reboot., it sometimes comes back on line before the phone's entry 
 is removed from the location table. In this case, the above solution 
 does not work.
 
 Can anyone provide advice on how to determine first registration of a 
 phone coming back online in another way?
 
 
 
 
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Re: [OpenSIPS-Users] Presence Issue

2009-09-26 Thread Saúl Ibarra
How are you loading the presnce modules? Please paste your config
file's relevant section.


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Re: [OpenSIPS-Users] unix domain MI

2009-09-26 Thread Bogdan-Andrei Iancu
Hi Richard,

I found some bug related to flushing (of output MI tree) when using the 
Datagram backend.

Please update from SVN (1.5 branch) and try again (or see rev #6191).

Thanks and regards,
Bogdan

Richard Revels wrote:
 I'm using unix domain socket for MI on opensips-1.5.2 and am getting  
 an error when trying to do opensipsctl ul show.  Is this something  
 that was fixed in a newer version?

 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:  
 ERROR:mi_datagram:mi_datagram_flush_tree: failed to write - reason too  
 long!
 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:  
 ERROR:mi_datagram:mi_datagram_flush_tree: failed to write - reason too  
 long!
 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:  
 ERROR:mi_datagram:datagram_recur_write_tree: failed to write -line too  
 long!!!
 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:  
 ERROR:mi_datagram:mi_datagram_server: failed to build the response

 Richard Revels




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Re: [OpenSIPS-Users] how do you determine first registration of a device

2009-09-26 Thread Alex G
this still does not solve the issue of a reboot that occurs in under 60
seconds. if a phone does not unregister on reboot, the record remains in the
db for those 60 seconds. if it comes back up in under 60 seconds, then to
opensips it looks like a reregistration.

is there someway to discern this scenario from the info in the location
table that is already there? maybe the callid or cseq or something?

On Sat, Sep 26, 2009 at 1:43 PM, Alex Balashov abalas...@evaristesys.comwrote:

 As Inaki said, what you're trying to achieve is a little ridiculous.

 But, if you must do it, I recommend using a database capable of
 advanced triggers and decent stored procedures (PostgreSQL) and using
 usrloc and registrar with full database backing and no caching or
 synchronisation (changes immediately reflected in database).  Then put
 a trigger on your 'location' table.  Trying to do this from the route
 script is silly.

 Alex G wrote:

  I have a scenario where I'm trying to fire a script based on the first
  registration of a device. In normal circumstances I can check the
  location table using registered() function and if there is an entry i
  know it's not the first registration. My problem is when a phone does a
  quick reboot., it sometimes comes back on line before the phone's entry
  is removed from the location table. In this case, the above solution
  does not work.
 
  Can anyone provide advice on how to determine first registration of a
  phone coming back online in another way?
 
 
  
 
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Re: [OpenSIPS-Users] how do you determine first registration of a device

2009-09-26 Thread Alex Balashov
A new REGISTER request will certainly have a different Call-ID.

However, I think the correct way to handle this is to just live with 
it.  If the phone did not unregister, then it will be subject to the 
consequences of that, and it's okay.

Kamailio 1.5.x's registrar module has a function called 
reg_free_contacts() which you could use if a save() fails due to a 
contact already existing in memory or database and then call save(). 
Perhaps OpenSIPS will implement something like this at some point.

Alex G wrote:

 this still does not solve the issue of a reboot that occurs in under 60 
 seconds. if a phone does not unregister on reboot, the record remains in 
 the db for those 60 seconds. if it comes back up in under 60 seconds, 
 then to opensips it looks like a reregistration.
 
 is there someway to discern this scenario from the info in the location 
 table that is already there? maybe the callid or cseq or something?
 
 On Sat, Sep 26, 2009 at 1:43 PM, Alex Balashov 
 abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
 
 As Inaki said, what you're trying to achieve is a little ridiculous.
 
 But, if you must do it, I recommend using a database capable of
 advanced triggers and decent stored procedures (PostgreSQL) and using
 usrloc and registrar with full database backing and no caching or
 synchronisation (changes immediately reflected in database).  Then put
 a trigger on your 'location' table.  Trying to do this from the route
 script is silly.
 
 Alex G wrote:
 
   I have a scenario where I'm trying to fire a script based on the
 first
   registration of a device. In normal circumstances I can check the
   location table using registered() function and if there is an entry i
   know it's not the first registration. My problem is when a phone
 does a
   quick reboot., it sometimes comes back on line before the phone's
 entry
   is removed from the location table. In this case, the above solution
   does not work.
  
   Can anyone provide advice on how to determine first registration of a
   phone coming back online in another way?
  
  
  
 
  
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Re: [OpenSIPS-Users] how do you determine first registration of a device

2009-09-26 Thread Alex Balashov
Alex Balashov wrote:
 A new REGISTER request will certainly have a different Call-ID.
 
 However, I think the correct way to handle this is to just live with 
 it.  If the phone did not unregister, then it will be subject to the 
 consequences of that, and it's okay.
 
 Kamailio 1.5.x's registrar module has a function called 
 reg_free_contacts() which you could use if a save() fails due to a 
 contact already existing in memory or database and then call save(). 
 Perhaps OpenSIPS will implement something like this at some point.

It actually has an unregister() function as well.

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Re: [OpenSIPS-Users] unix domain MI

2009-09-26 Thread Richard Revels
Thank you for looking into this.  Didn't mean to cause anyone work on  
the weekend.  I'll update from SVN tonight or tomorrow.


Richard Revels


On Sep 26, 2009, at 1:56 PM, Bogdan-Andrei Iancu wrote:

 Hi Richard,

 I found some bug related to flushing (of output MI tree) when using  
 the
 Datagram backend.

 Please update from SVN (1.5 branch) and try again (or see rev #6191).

 Thanks and regards,
 Bogdan

 Richard Revels wrote:
 I'm using unix domain socket for MI on opensips-1.5.2 and am getting
 an error when trying to do opensipsctl ul show.  Is this something
 that was fixed in a newer version?

 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:
 ERROR:mi_datagram:mi_datagram_flush_tree: failed to write - reason  
 too
 long!
 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:
 ERROR:mi_datagram:mi_datagram_flush_tree: failed to write - reason  
 too
 long!
 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:
 ERROR:mi_datagram:datagram_recur_write_tree: failed to write -line  
 too
 long!!!
 Sep 25 19:36:21 wtrunking /usr/local/opensips/sbin/opensips[10316]:
 ERROR:mi_datagram:mi_datagram_server: failed to build the response

 Richard Revels




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[OpenSIPS-Users] Polycom 650 can't Register behind NAT

2009-09-26 Thread osiris123d

I am sure this is not a OpenSIPs issue, but I was just wondering if people
have run into this and what they have done to resolve this.  I currently
have been testing my OpenSIPs config with Counterpaths Bria client and also
the SJphone.  They don't have any issues registering behind NAT.  I just
purchases a Polycom 650 phone and I cannot get it to register.  When I do an
ngrep I see that the phone is wanting OpenSIPS to talk back to it via port
5060.  When I do an ngrep with Bria or SJphone I see that when OpenSIPS
replies back to the softphone it is not on port 5060.  

Here is the ngrep showing a polycom trying to register

U 75.65.x.x:5060 - 66.80.x.x:5060
REGISTER sip:sip..com SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK484090eaC6F4185B.
From: DL sip:d...@sip..com;tag=2AED749F-64CCEB50.
To: sip:d...@sip..com.
CSeq: 1 REGISTER.
Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
Contact: sip:d...@192.168.0.100;methods=INVITE, ACK, BYE, CANCEL, OPTIONS,
INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPD
ATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
Accept-Language: en.
Max-Forwards: 70.
Expires: 3600.
Content-Length: 0.
.


U 66.80.x.x:5060 - 75.65.x.x:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
192.168.0.100;branch=z9hG4bK484090eaC6F4185B;received=75.65.x.x.
From: DL sip:d...@sip..com;tag=2AED749F-64CCEB50.
To: sip:d...@sip..com;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.e526.
CSeq: 1 REGISTER.
Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
Server:  SIP Proxy.
Content-Length: 0.
.


U 75.65.x.x:5060 - 66.80.x.x:5060
REGISTER sip:sip..com SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK8e1a28414313B72.
From: DL sip:d...@sip..com;tag=2AED749F-64CCEB50.
To: sip:d...@sip..com.
CSeq: 2 REGISTER.
Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
Contact: sip:d...@192.168.0.100;methods=INVITE, ACK, BYE, CANCEL, OPTIONS,
INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPD
ATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
Accept-Language: en.
Max-Forwards: 70.
Expires: 3600.
Content-Length: 0.

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Re: [OpenSIPS-Users] Polycom 650 can't Register behind NAT

2009-09-26 Thread Norman Brandinger
Try changing the port that the Polycom is listening on from 5060 to 
something else.

Regards,
Norm

osiris123d wrote:
 I am sure this is not a OpenSIPs issue, but I was just wondering if people
 have run into this and what they have done to resolve this.  I currently
 have been testing my OpenSIPs config with Counterpaths Bria client and also
 the SJphone.  They don't have any issues registering behind NAT.  I just
 purchases a Polycom 650 phone and I cannot get it to register.  When I do an
 ngrep I see that the phone is wanting OpenSIPS to talk back to it via port
 5060.  When I do an ngrep with Bria or SJphone I see that when OpenSIPS
 replies back to the softphone it is not on port 5060.  

 Here is the ngrep showing a polycom trying to register

 U 75.65.x.x:5060 - 66.80.x.x:5060
 REGISTER sip:sip..com SIP/2.0.
 Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK484090eaC6F4185B.
 From: DL sip:d...@sip..com;tag=2AED749F-64CCEB50.
 To: sip:d...@sip..com.
 CSeq: 1 REGISTER.
 Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
 Contact: sip:d...@192.168.0.100;methods=INVITE, ACK, BYE, CANCEL, OPTIONS,
 INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPD
 ATE, REFER.
 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
 Accept-Language: en.
 Max-Forwards: 70.
 Expires: 3600.
 Content-Length: 0.
 .


 U 66.80.x.x:5060 - 75.65.x.x:5060
 SIP/2.0 401 Unauthorized.
 Via: SIP/2.0/UDP
 192.168.0.100;branch=z9hG4bK484090eaC6F4185B;received=75.65.x.x.
 From: DL sip:d...@sip..com;tag=2AED749F-64CCEB50.
 To: sip:d...@sip..com;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.e526.
 CSeq: 1 REGISTER.
 Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
 Server:  SIP Proxy.
 Content-Length: 0.
 .


 U 75.65.x.x:5060 - 66.80.x.x:5060
 REGISTER sip:sip..com SIP/2.0.
 Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK8e1a28414313B72.
 From: DL sip:d...@sip..com;tag=2AED749F-64CCEB50.
 To: sip:d...@sip..com.
 CSeq: 2 REGISTER.
 Call-ID: 86e5160c-a73b99fd-e4c3b...@192.168.0.100.
 Contact: sip:d...@192.168.0.100;methods=INVITE, ACK, BYE, CANCEL, OPTIONS,
 INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPD
 ATE, REFER.
 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
 Accept-Language: en.
 Max-Forwards: 70.
 Expires: 3600.
 Content-Length: 0.

   


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Re: [OpenSIPS-Users] Polycom 650 can't Register behind NAT

2009-09-26 Thread osiris123d

So I went to the web GUI clicked on SIP and then clicked on Local Settings
and set the Local SIP Port to be 60111, but the polycom is still not
registering.  Here is the ngrep output now.


U 75.65.x.x:60111 - 66.80.x.x:5060
REGISTER sip:sip..com SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.100:60111;branch=z9hG4bK74ec705c1135DC1.
From: DL sip:d...@sip..com sip%3...@sip..com
;tag=D602822D-9C60EFAE.
To: sip:d...@sip..com sip%3...@sip..com.
CSeq: 1 REGISTER.
Call-ID: e2d7e872-d4ef43bf-4036e...@192.168.0.100.
Contact: sip:d...@192.168.0.100:60111;methods=INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRAC
K, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
Accept-Language: en.
Max-Forwards: 70.
Expires: 3600.
Content-Length: 0.
.

U 66.80.x.x:5060 - 75.65.x.x:60111
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.0.100:60111
;branch=z9hG4bK74ec705c1135DC1;received=75.65.x.x.
From: DL sip:d...@sip..com sip%3...@sip..com
;tag=D602822D-9C60EFAE.
To: sip:d...@sip..com sip%3...@sip..com
;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.df7d.
CSeq: 1 REGISTER.
Call-ID: e2d7e872-d4ef43bf-4036e...@192.168.0.100.
Server: s SIP Proxy.
Content-Length: 0.
.

U 75.65.x.x:60111 - 66.80.x.x:5060
REGISTER sip:sip..com SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.100:60111;branch=z9hG4bKb735008bA458A5D4.
From: DL sip:d...@sip..com sip%3...@sip..com
;tag=D602822D-9C60EFAE.
To: sip:d...@sip..com sip%3...@sip..com.
CSeq: 2 REGISTER.
Call-ID: e2d7e872-d4ef43bf-4036e...@192.168.0.100.
Contact: sip:d...@192.168.0.100:60111;methods=INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRAC
K, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
Accept-Language: en.
Max-Forwards: 70.
Expires: 3600.
Content-Length: 0.
.

U 66.80.x.x:5060 - 75.65.x.x:60111
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.0.100:60111
;branch=z9hG4bKb735008bA458A5D4;received=75.65.x.x.
From: DL sip:d...@sip..com sip%3...@sip..com
;tag=D602822D-9C60EFAE.
To: sip:d...@sip..com sip%3...@sip..com
;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.7ddc.
CSeq: 2 REGISTER.
Call-ID: e2d7e872-d4ef43bf-4036e...@192.168.0.100.
Server: s SIP Proxy.
Content-Length: 0.



On Sat, Sep 26, 2009 at 11:04 PM, Norman Brandinger (via Nabble) 
ml-user+123738-149177...@n2.nabble.comml-user%2b123738-149177...@n2.nabble.com
 wrote:

 Try changing the port that the Polycom is listening on from 5060 to
 something else.

 Regards,
 Norm

 osiris123d wrote:
  I am sure this is not a OpenSIPs issue, but I was just wondering if
 people
  have run into this and what they have done to resolve this.  I currently
  have been testing my OpenSIPs config with Counterpaths Bria client and
 also
  the SJphone.  They don't have any issues registering behind NAT.  I just
  purchases a Polycom 650 phone and I cannot get it to register.  When I do
 an
  ngrep I see that the phone is wanting OpenSIPS to talk back to it via
 port
  5060.  When I do an ngrep with Bria or SJphone I see that when OpenSIPS
  replies back to the softphone it is not on port 5060.
 
  Here is the ngrep showing a polycom trying to register
 
  U 75.65.x.x:5060 - 66.80.x.x:5060
  REGISTER sip:sip..com SIP/2.0.
  Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK484090eaC6F4185B.
  From: DL sip:[hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=0;tag=2AED749F-64CCEB50.

  To: sip:[hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=1.

  CSeq: 1 REGISTER.
  Call-ID: [hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=2.

  Contact: sip:[hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=3;methods=INVITE,
 ACK, BYE, CANCEL, OPTIONS,
  INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPD
  ATE, REFER.
  User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.2.0392.
  Accept-Language: en.
  Max-Forwards: 70.
  Expires: 3600.
  Content-Length: 0.
  .
 
 
  U 66.80.x.x:5060 - 75.65.x.x:5060
  SIP/2.0 401 Unauthorized.
  Via: SIP/2.0/UDP
  192.168.0.100;branch=z9hG4bK484090eaC6F4185B;received=75.65.x.x.
  From: DL sip:[hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=4;tag=2AED749F-64CCEB50.

  To: sip:[hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=5;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.e526.

  CSeq: 1 REGISTER.
  Call-ID: [hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=6.

  Server:  SIP Proxy.
  Content-Length: 0.
  .
 
 
  U 75.65.x.x:5060 - 66.80.x.x:5060
  REGISTER sip:sip..com SIP/2.0.
  Via: SIP/2.0/UDP 192.168.0.100;branch=z9hG4bK8e1a28414313B72.
  From: DL sip:[hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=7;tag=2AED749F-64CCEB50.

  To: sip:[hidden 
  email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3719554i=8.

  CSeq: 2 REGISTER.
  Call-ID: [hidden