Home » Forums » Community Edition » Help

Registrations Timing out, Peers unreachable... I'm going to lose my mind!

mikecentola
Posts: 34
 Member Since:
2007-07-09
Submitted by mikecentola on Thu, 02/25/2010 - 3:40pm.

This has been driving me crazy for over a week now. For some reason my phone server had stopped working properly and I started getting these unreachable problems. I installed 2.8 with no luck, reinstalled 2.6 and still had issues, so now I'm back on 2.8 and I'm still having the issues. I don't want to reinstall for a fourth time now. I've set the external ips. I've followed what seems any piece of advice on this forums or all over the internet, and I'm still having the issues. When the peers are all up, everything works great, but this seems to happen every few minutes causing all kinds of issues.

At first I thought it was a network issue, but I've checked and double checked all kinds of settings. I'm using bonded ethernet (802.3ad) for redundancy, and I thought that was the problem, but it still did it with just using a single port. I thought it was my firewall even after enabling TCP/UDP 5060 forwarded to the server. I even removed the firewall but it still did it. If it was the firewall, then the local extensions wouldn't be losing their registration also.

Could it be something with my SIP configurations? I'm not sure what I might be doing wrong. I would really REALLY appreciate any help that you all can give. I'm losing my mind and it's starting to hurt my businesses because I don't have phone connectivity. I could even offer some payment to someone who might be able to help me out with this.

Thanks in advance,
Mike

‹ Speaking clock sayings hour and minutes then hangs up What files should i back up ›


mikecentola
Posts: 34
 Member Since:
2007-07-09
other people i have talked
Fri, 02/26/2010 - 4:58am

other people i have talked to say it is probably a network problem, but i'm not experiencing any other issues with SSH or the web browser at all. I tried manually setting the registration time out in the phones' config to 3600, but I think that caused even more problems. Do you think I'd be better off setting it to something small like 360?

please help! going crazy :(



aerodesliza
Posts: 328
 Member Since:
2008-08-10
Try this, conect your Ip
Fri, 02/26/2010 - 5:31am

Try this, conect your Ip Phone to the trixbox server just using a path cord (ethernet cable), that will erase any network issue.



dickson
Posts: 1831
 Member Since:
2006-06-02
Explain your network
Fri, 02/26/2010 - 6:39am

Explain your network topology a bit.
Where is your server located in logical relation to your phones. You say you had a firewall, so your server is in a completely separate network?
Static or Dynamic IP? If dynamic, how often does it change?
Is it possible your ISP is filtering these ports? As aerodesliza suggested, if you plug a phone into the same local segment as your PBX with no firewalls or routers between the phone and your PBX, does the problem occur?

What phones are you using it with?
What phones have you tested it with?

Are your phones registering from different geographical locations to your server? Or are they all located in the same office/network

Do phones, once a connected and a call established, do those phone stay connected? So if you do a *43 echo test, does that call stay established indefinitely? If you have an established call, do your phones have this same issue? Or is it only occurring when there is no activity.

Does the same problem occur if you use an IAX softphone instead?

Just some questions to help try and figure your issue out



mikecentola
Posts: 34
 Member Since:
2007-07-09
Sorry, here is some more
Fri, 02/26/2010 - 7:42am

Sorry, here is some more info about my set up :)

INTERNET -> 2811 -> ASA5505 -> 3550-12T -> SRW2024P

So basically my core network switch is the 3550-12T. It does the DHCP, and Layer 3 routing. The SRW2024P is a PoE gigE layer 2 switch for the phones and phone server. They are all on the same network (ie 192.168.1.X) and in the same VLAN. Our external IP is static.

We are using polycom 501's and 601's. They are all in the same office, and they were working fine just a few weeks ago, when something started to act up, and I'm not sure what its from. Maybe from a yum update, I'm not sure.

It's funny that it seems to drop it every couple of minutes...

Here is some info from the logs:

[Feb 26 03:39:02] NOTICE[4422] chan_sip.c: Peer 'VP-SIPJFKA' is now Reachable. (97ms / 2000ms)
[Feb 26 03:39:02] NOTICE[4422] chan_sip.c: Peer 'VP-SIPJFKB' is now Reachable. (104ms / 2000ms)
[Feb 26 03:41:27] NOTICE[4422] chan_sip.c: -- Registration for 'XXXXXXXXXX@chi-reg.voipstreet.com' timed out, trying again (Attempt #1)
[Feb 26 03:41:27] NOTICE[4422] chan_sip.c: -- Registration for 'XXXXXXXXXX@jfk-primary.voicepulse.com' timed out, trying again (Attempt #1)
[Feb 26 03:41:32] NOTICE[4422] chan_sip.c: Peer '110' is now UNREACHABLE! Last qualify: 47
[Feb 26 03:41:32] NOTICE[4422] chan_sip.c: Peer '103' is now UNREACHABLE! Last qualify: 47
[Feb 26 03:41:32] NOTICE[4422] chan_sip.c: Peer '102' is now UNREACHABLE! Last qualify: 47
[Feb 26 03:41:32] NOTICE[4422] chan_sip.c: Peer '100' is now UNREACHABLE! Last qualify: 46
[Feb 26 03:41:32] NOTICE[4422] chan_sip.c: Peer '101' is now UNREACHABLE! Last qualify: 48
[Feb 26 03:41:32] NOTICE[4422] chan_sip.c: Peer 'VP-SIPJFKA' is now UNREACHABLE! Last qualify: 68
[Feb 26 03:41:32] NOTICE[4422] chan_sip.c: Peer 'VP-SIPJFKB' is now UNREACHABLE! Last qualify: 111
[Feb 26 03:41:32] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:32] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:32] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:32] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:32] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:42] NOTICE[4422] chan_sip.c: Peer '103' is now Reachable. (47ms / 2000ms)
[Feb 26 03:41:42] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:42] NOTICE[4422] chan_sip.c: Peer '110' is now Reachable. (48ms / 2000ms)
[Feb 26 03:41:42] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:42] NOTICE[4422] chan_sip.c: Peer '102' is now Reachable. (50ms / 2000ms)
[Feb 26 03:41:42] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:42] NOTICE[4422] chan_sip.c: Peer '101' is now Reachable. (52ms / 2000ms)
[Feb 26 03:41:42] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:42] NOTICE[4422] chan_sip.c: Peer '100' is now Reachable. (54ms / 2000ms)
[Feb 26 03:41:42] DEBUG[4389] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Feb 26 03:41:42] NOTICE[4422] chan_sip.c: Peer 'VP-SIPJFKA' is now Reachable. (56ms / 2000ms)
[Feb 26 03:41:42] NOTICE[4422] chan_sip.c: Peer 'VP-SIPJFKB' is now Reachable. (94ms / 2000ms)
[Feb 26 03:55:31] NOTICE[4422] chan_sip.c: -- Registration for 'XXXXXXXXXX@chi-reg.voipstreet.com' timed out, trying again (Attempt #1)
[Feb 26 04:01:44] NOTICE[4422] chan_sip.c: -- Registration for 'XXXXXXXXXX@chi-reg.voipstreet.com' timed out, trying again (Attempt #1)
[Feb 26 04:01:44] NOTICE[4422] chan_sip.c: -- Registration for 'XXXXXXXXX@jfk-primary.voicepulse.com' timed out, trying again (Attempt #1)



mikecentola
Posts: 34
 Member Since:
2007-07-09
I am running SIP debug right
Fri, 02/26/2010 - 8:46am

I am running SIP debug right now, looking for any problems.... Could this be part of the problem??? (I bolded it)

[Feb 26 10:40:53] VERBOSE[4422] logger.c: Retransmitting #1 (NAT) to 64.136.174.24:5060:
REGISTER sip:chi-reg.voipstreet.com SIP/2.0
Via: SIP/2.0/UDP 24.97.84.17:5060;branch=z9hG4bK6b3bd84f;rport
Max-Forwards: 70
From: ;tag=as01caf7b5
To:
Call-ID: 05c04c6c486325ef79b6ab9b42da23ef@127.0.0.1
CSeq: 1262 REGISTER
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Authorization: Digest username="XXXXXXXXX", realm="asterisk", algorithm=MD5, uri="sip:chi-reg.voipstreet.com", nonce="6672ce00", response="abee2b13e6d1d635b9aad155959a04cd"
Expires: 120
Contact:
Event: registration
Content-Length: 0

But then I see these:

[Feb 26 10:42:58] VERBOSE[4422] logger.c:
UDP://67.108.9.165:5060 --->
SIP/2.0 200 OK to keepalive
Via: SIP/2.0/UDP 10.41.10.1:5060;branch=z9hG4bK44b0bb66;rport=5060
From: "Unknown" ;tag=as5412f35b
To: ;tag=f84de4042d96a5aa76310e059e0d6884.190e
Call-ID: 3006c30707373f177bdab5511523d11c@24.97.84.17
CSeq: 102 OPTIONS
Server: OpenSIPS (1.6.0-notls (i386/linux))
Content-Length: 0

Which seems right since its receiving keepalives. The funny thing is seems to re-register fine now. Am I being paranoid?



SkykingOH
Posts: 9677
 Member Since:
2007-12-17
The SIP traces are from a
Fri, 02/26/2010 - 10:34am

The SIP traces are from a public IP. What IP's re on the phones?

Clearly the trixbox is loosing connectivity with the phones.

Have you checked the ports for errors as I have requested of you?

You can install mtr (yum install mtr) and then run 'mtr ip.address.of.phone' and watch for dropped packets.

--

Scott

aka "Skyking"



mikecentola
Posts: 34
 Member Since:
2007-07-09
The server is 10.41.10.1 and
Fri, 02/26/2010 - 12:47pm

The server is 10.41.10.1 and the phones are 10.41.10.1XX. I ran 'mtr 10.41.10.106' which happens to be my phone, and I called my cell phone. It went through and showed no errors. However when I hung up the phone, my cell phone kept ringing, so it didn't disconnect the far end or whatever.

here's the mtr output....

                                                My traceroute  [v0.72]
calypso.voip.hq.technoticmedia.com (0.0.0.0)                                                   Fri Feb 26 14:46:23 2010
Keys:  Help   Display mode   Restart statistics   Order of fields   quit
                                                                               Packets               Pings
 Host                                                                        Loss%   Snt   Last   Avg  Best  Wrst StDev
 1. 10.41.10.106                                                              0.0%   111    1.1   1.1   1.0   1.4   0.1

I'm hoping this is just a configuration issue here. Any other configs I should post?



mikecentola
Posts: 34
 Member Since:
2007-07-09
I thought I'd post this as
Fri, 02/26/2010 - 4:24pm

I thought I'd post this as well....

[calypso.voip.hq.technoticmedia.com ~]# ifconfig
eth0 Link encap:Ethernet HWaddr 00:0E:A6:F1:EE:2E
inet addr:10.41.10.1 Bcast:10.41.10.255 Mask:255.255.255.0
inet6 addr: fe80::20e:a6ff:fef1:ee2e/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:181902 errors:0 dropped:0 overruns:0 frame:0
TX packets:190844 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:39716712 (37.8 MiB) TX bytes:72291445 (68.9 MiB)
Interrupt:169

Doesn't show any network errors :(



SkykingOH
Posts: 9677
 Member Since:
2007-12-17
Mike, When you get a chance
Fri, 02/26/2010 - 11:55pm

Mike,

When you get a chance read the "how to ask for help" sticky. Notice I fixed your post, you can actually read the output now.

You have an odd problem, it is not a configuration issue as Asterisk is loosing it's ability to contact the phone. The lack of a disconnect clearly shows a dropped packed.

Leave MTR up overnight.

--

Scott

aka "Skyking"



mikecentola
Posts: 34
 Member Since:
2007-07-09
Here's a longer MTR
Mon, 03/01/2010 - 8:55am

Here's a longer MTR

                                                 My traceroute  [v0.72]
calypso.voip.hq.technoticmedia.com (0.0.0.0)                                                   Mon Mar  1 10:50:56 2010
Keys:  Help   Display mode   Restart statistics   Order of fields   quit
                                                                               Packets               Pings
 Host                                                                        Loss%   Snt   Last   Avg  Best  Wrst StDev
 1. 10.41.10.106                                                              0.0% 84362    1.1   1.0   1.0   4.6   0.2

I'm working on installing trixbox into a VM with VMWare on my mac. It's on a different subnet and different switch all together. I know it won't eliminate much, but it may eliminate my config, or point a finger to it. I'll post back shortly.



mikecentola
Posts: 34
 Member Since:
2007-07-09
Ok so, fresh install Trixbox
Tue, 03/02/2010 - 8:52am

Ok so, fresh install Trixbox 2.6 in a VM. On that same computer, I loaded up X-Lite softphone and registered to the server. Sure enough, I got a Peer is now UNREACHABLE error. I didn't even run yum update or anything. This has to be a configuration problem or maybe a bug. I even removed my firewall from the picture and I was still getting time outs.

I can't believe that no one else is having problems like this. :(



415eric
Posts: 416
 Member Since:
2009-10-29
I doubt it is a trixbox bug.
Tue, 03/02/2010 - 9:41am

I doubt it is a trixbox bug. A lot of us here have installed trixbox umpteen times without experiencing the same issues. For those who did it was either a network error or user error. That being said IF you want help you need to provide the information we need to help. The standard garbage in garbage out.

The fact you are not able to get a softphone on the same box to work leads me to believe your configurations are off. We need to see your config files to see what you have configured wrong. Files like sip_additional.conf, there is a post with a list of all the files you should post. Be sure to places x's in the sensitive areas.

--


mikecentola
Posts: 34
 Member Since:
2007-07-09
Lots of info!
Tue, 03/02/2010 - 10:05am

Ok no problem :) Here is as much info as possible!

sip show peer on the soft phone

* Name       : 120
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-internal
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 120@default
  VM Extension : *97
  LastMsgsSent : 0/0
  Call limit   : 50
  Dynamic      : Yes
  Callerid     : "device" <120>
  MaxCallBR    : 384 kbps
  Expire       : 2416
  Insecure     : no
  Nat          : Always
  ACL          : Yes
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 
  Addr->IP     : 10.41.15.22 Port 40590
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 120
  SIP Options  : (none)
  Codecs       : 0x28000c (ulaw|alaw|h263|h264)
  Codec Order  : (ulaw:20,alaw:20)
  Auto-Framing:  No 
  Status       : OK (4 ms)
  Useragent    : X-Lite Beta release 4.0 v3 stamp 56227
  Reg. Contact : sip:120@10.41.15.22:40590;rinstance=80abfcef53aed5f6

sip show registry

Host                            Username       Refresh State                Reg.Time                 
jfk-backup.voicepulse.com:5060  UFg4CHK79E         105 Registered           Tue, 02 Mar 2010 11:53:50
jfk-primary.voicepulse.com:506  UFg4CHK79E         105 Registered           Tue, 02 Mar 2010 11:53:50

sip_additional.conf

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: <a href="http://freepbx.org/configuration_files" title="http://freepbx.org/configuration_files">http://freepbx.org/configuration_files</a>       ;
;--------------------------------------------------------------------------------;
;

[101]
deny=0.0.0.0/0.0.0.0
type=friend
secret=XXXXXXX
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=101@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/101
context=from-internal
canreinvite=no
callgroup=
callerid=device <101>
accountcode=
call-limit=50

[120]
deny=0.0.0.0/0.0.0.0
type=friend
secret=XXXXXXX
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=120@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/120
context=from-internal
canreinvite=no
callgroup=
callerid=device <120>
accountcode=
call-limit=50

[VP-SIPJFKA]
disallow=all
type=peer
context=from-pstn
username=XXXXXXXXXX
secret=XXXXXXXXXXX
host=jfk-primary.voicepulse.com
qualify=yes
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes

[VP-SIPJFKB]
disallow=all
type=peer
context=from-pstn
username=XXXXXXXXXXXX
secret=XXXXXXXXXX
host=jfk-backup.voicepulse.com
qualify=yes
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes

sip_nat.conf

externip=24.97.84.17
localnet=10.41.0.0/255.255.0.0
nat=yes

sip debug output

REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 64.61.93.190:5060:
REGISTER sip:jfk-primary.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 24.97.84.17:5060;branch=z9hG4bK67d3ff7c;rport
From: <sip:XXXXXXXXXXX@jfk-primary.voicepulse.com>;tag=as551237ec
To: <sip:XXXXXXXXXXX@jfk-primary.voicepulse.com>
Call-ID: 230baea57789a4bc77d34d520c5d0d7f@127.0.0.1
CSeq: 135 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="XXXXXXXXXXXXX", realm="jfk-primary.voicepulse.com", algorithm=MD5, uri="sip:jfk-primary.voicepulse.com", nonce="4b8d4506b01fef34eadf706dd25b42e29f760226", response="c020bde7e39d1291cac24ea32a001d2a", qop=auth, cnonce="1a1ce9bc", nc=00000002
Expires: 120
Contact: <sip:s@24.97.84.17>
Event: registration
Content-Length: 0


---
Reliably Transmitting (NAT) to 10.41.15.22:40590:
OPTIONS sip:120@10.41.15.22:40590;rinstance=80abfcef53aed5f6 SIP/2.0
Via: SIP/2.0/UDP 10.41.15.107:5060;branch=z9hG4bK00578ece;rport
From: "Unknown" <sip:Unknown@10.41.15.107>;tag=as28bf6435
To: <sip:120@10.41.15.22:40590;rinstance=80abfcef53aed5f6>
Contact: <sip:Unknown@10.41.15.107>
Call-ID: 714c246e75cdc9b45161780e7372cd6d@10.41.15.107
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Mar 2010 17:01:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Reliably Transmitting (NAT) to 64.61.93.190:5060:
OPTIONS sip:jfk-primary.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 24.97.84.17:5060;branch=z9hG4bK16f1e31f;rport
From: "Unknown" <sip:Unknown@24.97.84.17>;tag=as5ad3fea5
To: <sip:jfk-primary.voicepulse.com>
Contact: <sip:Unknown@24.97.84.17>
Call-ID: 6d1c14f904b603df6e7b7acf34ad2169@24.97.84.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Mar 2010 17:01:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Reliably Transmitting (NAT) to 67.108.9.165:5060:
OPTIONS sip:jfk-backup.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 24.97.84.17:5060;branch=z9hG4bK2dbbad82;rport
From: "Unknown" <sip:Unknown@24.97.84.17>;tag=as031f77a6
To: <sip:jfk-backup.voicepulse.com>
Contact: <sip:Unknown@24.97.84.17>
Call-ID: 22d406ce539661d73dccbb995bcd3743@24.97.84.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Mar 2010 17:01:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 10.41.15.22:40590 --->



<------------->
trixbox1*CLI> 
<--- SIP read from 10.41.15.22:40590 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.41.15.107:5060;branch=z9hG4bK00578ece;rport=5060
Contact: <sip:10.41.15.22:40590>
To: <sip:120@10.41.15.22:40590;rinstance=80abfcef53aed5f6>;tag=3a797429
From: "Unknown"<sip:Unknown@10.41.15.107>;tag=as28bf6435
Call-ID: 714c246e75cdc9b45161780e7372cd6d@10.41.15.107
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite Beta release 4.0 v3 stamp 56227
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '714c246e75cdc9b45161780e7372cd6d@10.41.15.107' Method: OPTIONS

<--- SIP read from 64.61.93.190:5060 --->
SIP/2.0 100 Trying Registration
Via: SIP/2.0/UDP 10.41.10.1:5060;branch=z9hG4bK67d3ff7c;rport=1229
From: <sip:XXXXXXXXXX@jfk-primary.voicepulse.com>;tag=as551237ec
To: <sip:XXXXXXXXXXXX@jfk-primary.voicepulse.com>
Call-ID: 230baea57789a4bc77d34d520c5d0d7f@127.0.0.1
CSeq: 135 REGISTER
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from 64.61.93.190:5060 --->
SIP/2.0 200 OK to keepalive
Via: SIP/2.0/UDP 10.41.10.1:5060;branch=z9hG4bK16f1e31f;rport=1229
From: "Unknown" <sip:Unknown@10.41.10.1>;tag=as5ad3fea5
To: <sip:jfk-primary.voicepulse.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.03ac
Call-ID: 6d1c14f904b603df6e7b7acf34ad2169@24.97.84.17
CSeq: 102 OPTIONS
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '6d1c14f904b603df6e7b7acf34ad2169@24.97.84.17' Method: OPTIONS
trixbox1*CLI> 
<--- SIP read from 64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.41.10.1:5060;branch=z9hG4bK67d3ff7c;rport=1229
From: <sip:XXXXXXXXXX@jfk-primary.voicepulse.com>;tag=as551237ec
To: <sip:XXXXXXXXXX@jfk-primary.voicepulse.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.7300
Call-ID: 230baea57789a4bc77d34d520c5d0d7f@127.0.0.1
CSeq: 135 REGISTER
Contact: <sip:s@24.97.84.17:1229>;expires=120;received="sip:24.97.84.17:1229"
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '230baea57789a4bc77d34d520c5d0d7f@127.0.0.1' in 32000 ms (Method: REGISTER)

<--- SIP read from 67.108.9.165:5060 --->
SIP/2.0 200 OK to keepalive
Via: SIP/2.0/UDP 10.41.10.1:5060;branch=z9hG4bK2dbbad82;rport=1223
From: "Unknown" <sip:Unknown@10.41.10.1>;tag=as031f77a6
To: <sip:jfk-backup.voicepulse.com>;tag=f84de4042d96a5aa76310e059e0d6884.523c
Call-ID: 22d406ce539661d73dccbb995bcd3743@24.97.84.17
CSeq: 102 OPTIONS
Server: OpenSIPS (1.6.0-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '22d406ce539661d73dccbb995bcd3743@24.97.84.17' Method: OPTIONS

Hopefully this is enough to help you guys help me! :)



A.Salah
Posts: 99
 Member Since:
2011-02-16
so this problem has no
Mon, 01/09/2012 - 5:17am

so this problem has no solution ??
"pbx.c: FONALITY: This thread has already held the conlock, skip locking"



Comment viewing options

Select your preferred way to display the comments and click "Save settings" to activate your changes.