vicidial.org

by **chrisnetronix** » Thu Aug 26, 2010 4:32 pm

I can get xlite to login, but when I try to log in with the agent interface I get no sessions available.

here is the cli before I set qualify to no in sip.conf

[Aug 24 21:23:04] NOTICE[3751]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/102-0000000c

the congesting stopped after i changed qualify to no, then reloaded via asterisk, but I still get no sessions available, it was suggested that this may indicate a severe time delay when communicating with your x-lite from the server.

But I have no idea how to fix this, and I NEED to get this resolved.

any help would be great!

by **williamconley** » Sat Aug 28, 2010 3:02 pm

when you post, please post your entire configuration including (but not limited to) your installation method and OS with kernel or version, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

by **chrisnetronix** » Sun Aug 29, 2010 2:11 am

by **williamconley** » Sun Aug 29, 2010 10:13 am

the two most obvious possibilities (throughout the history of this response on this forum, which has come up many times ...):

1) have you changed the IP address of the server? (if so, did you run the update server ip script?)

2) are there vicidial_conferences under the Admin->Conferences->Show Vicidial Conferences menu item (And are those conferences also actually in the meetme.conf file so they really exist?)

by **chrisnetronix** » Sun Aug 29, 2010 9:00 pm

yes i ran the update script for the ip

the conferences area in admin shows all the conference numbers and the ability to modify them.

I took the first number and put in my ip and the extension 7777

meetme.conf looks like this:

[rooms]
conf => 8600
conf => 8601,1234

#include meetme-vicidial.conf

meetme-vicidial.conf looks like this:

; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
; ViciDial Conferences:

; Conferences:
conf => 8600001

; END OF FILE

and I still get the no sessions error on the agent end.

also whats weird is when I try to edit the phone extension 102's password, it changes in vicidial but it wont allow the change in x lite it says 403 forbidden.

by **williamconley** » Sun Aug 29, 2010 9:41 pm

your conferences are missing. your install is bad. you should have many more conferences than that.

you may seriously want to consider just reinstalling and watching for errors (lost internet during the installation?)

there is a sql script that runs to install the "conferences" in the GUI, but those aren't the "real" conferences, they are merely references to the conferences in the meetme.conf file (or an included file). if they are in the GUI but not in the meetme.conf, it won't work.

the question is why aren't they there?

in this case: why is there not a meetme.conf file (or included entry) for these? is it because the sample file wasn't written during installation or is your version supposed to "generate" them in the meetme-vicidial.conf"? if it is supposed to generate them ... why isn't it? is "generate conf files" set to yes for this server in admin->servers?

i've never had an installation missing a component like that "work" after just adding the component. it always turns out LOTS of things are missing and i should have just reinstalled.

experience. if it isn't just "change a no to a yes" for the server, if the problem is deeper than that ... REINSTALL. watch for errors.

by **chrisnetronix** » Sun Aug 29, 2010 9:44 pm

well it is set to yes, I have already scheduled a reinstall tomorrow, I will keep you informed on how it goes.

by **williamconley** » Sun Aug 29, 2010 11:40 pm

after your installation, before making any changes to the system, ensure that your vicidial configurations are being written after you make a change to a sip phone or sip carrier (assuming you are using sip).

if you change a conf password for a sip phone, when the server's clock starts a new minute the vicidial script should generate a new sip-vicidial.conf file and "reload" asterisk.

this is one of the most basic "it works" tests in vicidial. it's also a good indicator that you have a 'good' install.

two more are (these are the initial tests we use to see if install was successful):

Code: Select all: screen -list

which should show all the screens running, up to 9 of them depending on your install. if it only shows a few (or none) the install is bad. start over.

Code: Select all: asterisk -R

should bring up an Asterisk console and inform you of the Asterisk version running. No errors.

Ordinarily, if both of those two tests are "good" the install is successful and VICIdial should work. It's fairly rare for those two to be good and yet NOT have a successful installation, unless something is damaged AFTER that moment.

by **chrisnetronix** » Sat Sep 04, 2010 8:25 am

ok, I have reinstalled everything went smoothly and the agent interface logs in but x lite does not ring, what ports or config should zi use for my router.

aslo when doing a play message cmpaign from remote agent once i make it active cli says this:

[Sep 4 09:19:41] NOTICE[9607]: chan_local.c:508 local_call: No such extension/context 12xxxxxxx@default while calling Local channel
[Sep 4 09:19:41] NOTICE[9607]: channel.c:3291 __ast_request_and_dial: Unable to call channel Local/127xxxxxxx@default

what does it mean by no such extension exist

by **williamconley** » Sat Sep 04, 2010 8:54 am

1) shall i assume that you once again have installed this:

ViciBox_Redux.i686-2.0.2 from .iso | Vicidial 2.2.1-237 Build 100510-2015 | Asterisk 1.4.21.2 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation |

2) what set of directions are you using for your installation? and does it contain a section to register a phone (x-lite for instance)?
3) since you do not have a functional system yet, continuing with other configurations and tests is putting the cart ahead of the horse (besides which I did not understand "doing a play message campaign from remote agent"). How about we successfully register a phone and log in an agent and DIAL OUT from a campaign with that logged in agent before we attempt to build a nuclear power plant with it

?

by **chrisnetronix** » Sat Sep 04, 2010 8:57 am

sounds good my friend, so basically the agent area logs in now, but x lite does not ring, what configuration should I make my router?

by **williamconley** » Sat Sep 04, 2010 8:58 am

what set of directions are you using for your installation? and does it contain a section to register a phone (x-lite for instance)?

by **williamconley** » Sat Sep 04, 2010 9:00 am

http://www.vicidial.org/VICIDIALforum/v ... 2850#52850

please do not double post

by **chrisnetronix** » Sat Sep 04, 2010 9:02 am

I am using the managers manual latest version,

I used it to do carrier, campaign, load leads, make user, setup remote agent

and this is the other info you needed.

the version and build is different.

ViciBox_Redux.i686-2.0.2 from .iso | VERSION: 2.4-278
BUILD: 100901-2055 | Asterisk 1.4.21.2 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation |

by **williamconley** » Sat Sep 04, 2010 9:06 am

never use the phrase "latest version". always include the version you are using. latest can change daily and confuse anyone reading this post tomorrow.

and what if i have a later version of the manual than you do?

you still do not answer:

and does it contain a section to register a phone (x-lite for instance)?

by **chrisnetronix** » Sat Sep 04, 2010 9:13 am

I am using managers manual 220, but the only place I see xlite mentioned is here:

A. Add a new SIP or IAX phone to the system
This tutorial goes over the steps needed to create a phone account entry in the system that you can
configure a soft-phone(computer based phone) or SIP-based hard-phone(a separate physical phone) to
use.
For Soft-phones, we usually recommend the IAX soft-phones Zoiper(http://www.zoiper.com) and
KIAX(https://sourceforge.net/projects/kiax/) because IAX is a native protocol to ViciDial and it can go
through firewalls easier than SIP and it uses less bandwidth than SIP. Some SIP soft-phones that will
work are Xlite and Eyebeam. As for hard-phones: Polycom and Snom are recommended, but
Linksys/Cisco and Grandstream should also work.
1. Go to the ViciDial administration page, go to the ADMIN section, click on PHONES and click on
the ADD A NEW PHONE link
2. For this tutorial we will use the following values for the fields on the ADD A NEW PHONE form:
phone extension: 201
dialplan number: 201
voicemail box: 201
outbound callerid: 7275551212
<we will leave phone and computer IP address fields blank>
server ip: 10.10.10.15 (set this to your server ip)
login: 201
password: test
status: ACTIVE
active account: Y
phone type: SIP
full name: sip201
<we will leave company and picture fields blank>
client protocol: SIP
local gmt: -5 (GMT timezone, you should NOT include adjusting for Daylight Savings Time)
(0 is the UK, +1 is most of Europe, -5 is EST in the USA, -8 is PST in the USA)
Template ID: SIP_generic
3. Click the submit button to create this phone record. After one minute, the phone will be active in
the system.
NOTE: if you are using a phone outside of a local network, we would strongly recommend changing
the “Conf File Secret” field for enhanced security. This field is what you would use as the phone secret
or password when you register it on your soft-phone or hard-phone.
4. Go to the Agent's phone and configure the server(or proxy) for the same IP address as you selected
above in step 2 for Server-IP, and the login(username) is the Extension as defined above in step 2 and
password(secret) is defined as the “Conf File Secret” field in the Phones entry, then click to “Register”
if that option is available, or restart the phone or application.
5. Have the agent log into the vicidial.php script with the phone login(Login) of "201" and phone
2010-05-10 version 11 ©2010 Matt Florell
password(Password) of "test".
6. Within a few seconds the agent's phone should ring and they will be logged into ViciDial and can
proceed as normal to take calls.
NOTES:
- You can also use phones entries to live monitor through the Real-time report. An example of this is
given in Tutorial P.
- If you need to add Channelbank or FXO/FXS phones, you will need to coordinate with your systems
administrator to set up the zaptel/dahdi conf files properly as well as adding the dialplan entries to
connect to the channelbank agent phones. There are sample zaptel.conf and zapata.conf files included
with ViciDial for these kinds of connections, as well as sample dialplan entries in the extensions.conf
sample files.

by **williamconley** » Sat Sep 04, 2010 9:22 am

ok, now does your x-lite phone say "ready" or "Registration error" or ...?

by **chrisnetronix** » Sat Sep 04, 2010 9:23 am

it says ready: your username is 201

by **williamconley** » Sat Sep 04, 2010 9:39 am

ok. and you enter the phone logon/pass on the first screen, the user logon/pass and campaign on the second screen ... and then you submit and get the agent screen?

but your phone does not ring?

by **chrisnetronix** » Sat Sep 04, 2010 11:18 am

correct

by **williamconley** » Sat Sep 04, 2010 12:16 pm

Asterisk CLI output during this theoretically attempted call from asterisk to your soft phone?

If it doesn't show anything obvious, try using sip debug to get more information

by **chrisnetronix** » Sat Sep 04, 2010 1:03 pm

here is the cli, logged in to agent interface, with xlite running:

14:02:02] Found
[Sep 4 14:02:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 4 14:02:02] Found
[Sep 4 14:02:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 4 14:02:02] == Parsing '/etc/asterisk/manager.conf': [Sep 4 14:02:02] Found
[Sep 4 14:02:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 4 14:02:02] == Parsing '/etc/asterisk/manager.conf': [Sep 4 14:02:02] Found
[Sep 4 14:02:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 4 14:02:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 4 14:02:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 4 14:02:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 4 14:02:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 4 14:02:07] == Parsing '/etc/asterisk/manager.conf': [Sep 4 14:02:07] Found
[Sep 4 14:02:07] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 4 14:02:07] == Parsing '/etc/asterisk/manager.conf': [Sep 4 14:02:07] Found
[Sep 4 14:02:07] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 4 14:02:07] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 4 14:02:07] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 4 14:02:08] NOTICE[3764]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/201-00000004
[Sep 4 14:02:08] > Channel SIP/201-00000004 was never answered.
[Sep 4 14:02:10] == Manager 'sendcron' logged off from 127.0.0.1

by **williamconley** » Sat Sep 04, 2010 8:01 pm

1) you need to post ONLY the output during the attempted logon (and state that is what you posted). Otherwise, of course, generic output for some other timespan really doesn't help (and if we don't know whether it is from that time, we'd have to take wild pot-shot guesses at what's wrong, not the best technician's method)
2) auto-congesting: is your x-lite on the same physical network as your server? set "qualify=no" in your SIP phone definition to turn off auto-congesting under some circumstances; also, are BOTH your server and your soft-phone behind firewalls and not on the same subnet? (if so, change to Zoiper or KIAX and use IAX instead of SIP)

so ...
(a) show sip debug output which RESULTS in the "auto-congesting" result.
(b) post the sip/debug output from an attempted logon moment, stating that is what it is.

by **chrisnetronix** » Sat Sep 04, 2010 8:15 pm

this is the cli / sip debug WITHOUT qualify set to NO:

the setup is the same as before, server colocated, I am on a computer at home logging into x lite, with antivirus / personal firewall and router.

my servers ifconfig is this:

eth0 Link encap:Ethernet HWaddr 00:25:90:06:B6:0C
inet addr:66.71.xxx.xxx Bcast:66.71.xxx.xxx Mask:255.255.255.248
inet6 addr: fe80::225:90ff:fe06:b60c/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:153300 errors:0 dropped:0 overruns:0 frame:0
TX packets:164049 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:100
RX bytes:40676771 (38.7 Mb) TX bytes:63262658 (60.3 Mb)
Memory:fb5e0000-fb600000

eth0:6671 Link encap:Ethernet HWaddr 00:25:90:06:B6:0C
inet addr:66.71.xxx.xxx Bcast:66.71.xxx.xxxx Mask:255.255.255.248
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Memory:fb5e0000-fb600000

eth0:6671 Link encap:Ethernet HWaddr 00:25:90:06:B6:0C
inet addr:66.71.xxx.xxx Bcast:66.71.xxx.xxx Mask:255.255.255.248
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Memory:fb5e0000-fb600000

eth0:6671 Link encap:Ethernet HWaddr 00:25:90:06:B6:0C
inet addr:66.71.xxx.xxxx Bcast:66.71.xxx.xxxx Mask:255.255.255.248
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Memory:fb5e0000-fb600000

eth0:6671 Link encap:Ethernet HWaddr 00:25:90:06:B6:0C
inet addr:66.71.xxx.xxx Bcast:66.71.xxxx.xxx Mask:255.255.255.248
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
Memory:fb5e0000-fb600000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:725798 errors:0 dropped:0 overruns:0 frame:0
TX packets:725798 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:53266442 (50.7 Mb) TX bytes:53266442 (50.7 Mb)

this is the cli readout right when i try to log in as agent, and it tries to dial.
cli =

[Sep 4 21:09:11]
<--- SIP read from 71.203.248.167:59824 --->

<------------->
[Sep 4 21:09:11] Retransmitting #6 (NAT) to 71.203.248.167:59824:
INVITE sip:201@71.203.248.167:59824;rinstance=2bb838d7d39f2e3e;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.170:5060;branch=z9hG4bK79416fac;rport
From: "S1009042109058600051" <sip:0000000000@66.71.249.170>;tag=as72ed169f
To: <sip:201@71.203.248.167:59824;rinstance=2bb838d7d39f2e3e;cpd=on>
Contact: <sip:0000000000@66.71.249.170>
Call-ID: 67ca36d72b3e6fcc4a2e60c623b07886@66.71.249.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1009042109058600051" <sip:0000000000@66.71.249.170>;privacy=off;screen=no
Date: Sun, 05 Sep 2010 01:09:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 3696 3696 IN IP4 66.71.249.170
s=session
c=IN IP4 66.71.249.170
t=0 0
m=audio 19450 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep 4 21:09:13] Really destroying SIP dialog '44383aa64c4147d84416963818f958e3@66.71.249.170' Method: NOTIFY
[Sep 4 21:09:13] NOTICE[3764]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/201-00000005
[Sep 4 21:09:13] > Channel SIP/201-00000005 was never answered.
[Sep 4 21:09:13] Scheduling destruction of SIP dialog '67ca36d72b3e6fcc4a2e60c623b07886@66.71.249.170' in 6400 ms (Method: INVITE)
[Sep 4 21:09:13] Really destroying SIP dialog 'ZTgyZGM3MzkzZTY3NTQ5ODI5YjE3ODdiYjE0YjRjYjI.' Method: REGISTER
[Sep 4 21:09:15] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 4 21:09:41]

by **williamconley** » Sat Sep 04, 2010 8:45 pm

why does your server have 5 eth0's?

by **chrisnetronix** » Sun Sep 05, 2010 2:20 pm

there are 5 ip adresses, assigned to the server, this is the way it is setup , should it be a certain way

by **williamconley** » Sun Sep 05, 2010 3:00 pm

turn off 4 of them. you can only have ONE ethX (of a certain number) without having issues. when you get 5 ips you are expected to have 5 different computers or 5 NIC cards.

do you NEED to use 5 different IPs for some special reason?

this can (and likely is) cause a bit of networking confusion.

unless you have a special need, you should ONLY use 1 IP address on 1 Network Card for this system.

AFTER you have the system up and running, experiment with other IPs, although I still don't know why you'd need 'em.

by **chrisnetronix** » Sun Sep 05, 2010 4:58 pm

once I have only 1 set, do I need to do a reinstall or what?

by **williamconley** » Sun Sep 05, 2010 8:49 pm

depends on whether that clears up your interference?!

by **chrisnetronix** » Sun Sep 05, 2010 8:50 pm

ok, I will take them out, what should I do after that.

by **williamconley** » Sun Sep 05, 2010 9:17 pm

do you know anyone who can set up a standard internet connection for a linux server? (of course, i'd test the one that is left over when you have it running ... but you'll need to be sure you have a single valid internet connection for your server)

by **chrisnetronix** » Sun Sep 05, 2010 9:19 pm

the server is colocated in a data center, but I have the admin on IM, what do you need him to do?

we have 1 ip setup, and I also use the ip to reach the admin / agent interface.

by **williamconley** » Sun Sep 05, 2010 10:22 pm

we need a single IP set up so that all network traffic will get to and from this server on that IP no matter what the destination (such as carriers and phones).

as if it had been set up with a single IP from the beginning.

then we need to test the ability to register a phone

then we need to test the ability to RING that phone when the agent logs in (as in: when asterisk attempts to call the phone, it succeeds in ringing the phone)

after that we can move on. but we need to get there first.

see how far you can get in calling the phone from asterisk (including looking at the asterisk CLI during attempts with SIP debug on if necessary) by logging in as an agent (which would generate a call to the agent) and/or by entering "dial XXX@default" at the asterisk command line (where XXX is the phone's extension)

by **chrisnetronix** » Tue Sep 07, 2010 12:32 am

here is the xlite logged in via cli with sip debug:

Also: qualifyy is set at no

<------------>
[Sep 7 01:28:14] Reliably Transmitting (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:14] Retransmitting #3 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:14] Retransmitting #1 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:14] Retransmitting #2 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:14] Retransmitting #3 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:15] Retransmitting #4 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:15] Really destroying SIP dialog 'Zjg2MjcyMTM4ZTJhMGYwZThmM2Q3ODY2MTdhMWVjNTM.' Method: REGISTE R
[Sep 7 01:28:15] Retransmitting #4 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:16] Retransmitting #5 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:17] Retransmitting #5 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:19] Retransmitting #6 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:20] Retransmitting #6 (NAT) to 71.203.248.167:12100:
NOTIFY sip:201@71.203.248.167:12100;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK1f1ad9ae;rport
Route: <sip:201@71.203.248.167:12100>
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as5ba6684d
To: <sip:201@71.203.248.167:12100;cpd=on>;tag=045ce871
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg.
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@66.71.249.xxx
Voice-Message: 0/0 (0/0)

---
[Sep 7 01:28:23] WARNING[3764]: chan_sip.c:2015 retrans_pkt: Maximum retries exceeded on transmission NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg. for seqno 102 (Non-critical Request) -- See doc/sip-retransmit.txt.
[Sep 7 01:28:24] WARNING[3764]: chan_sip.c:2015 retrans_pkt: Maximum retries exceeded on transmission NDU2NWU1ZmY3NzQ1MjEwNGU1Mjg2NmE0NzhmYzhmNjg. for seqno 103 (Non-critical Request) -- See doc/sip-retransmit.txt.

Here is when the agent logs in: still no x -lite ring:

[Sep 7 01:31:01] == Parsing '/etc/asterisk/manager.conf': [Sep 7 01:31:01] Found
[Sep 7 01:31:01] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 01:31:02] == Parsing '/etc/asterisk/manager.conf': [Sep 7 01:31:02] Found
[Sep 7 01:31:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 01:31:02] == Parsing '/etc/asterisk/manager.conf': [Sep 7 01:31:02] Found
[Sep 7 01:31:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 01:31:02] == Parsing '/etc/asterisk/manager.conf': [Sep 7 01:31:02] Found
[Sep 7 01:31:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 01:31:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 01:31:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 01:31:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 01:31:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 01:31:02] Retransmitting #4 (NAT) to 71.203.248.167:12100:
INVITE sip:201@71.203.248.167:12100;rinstance=fd0181fdb1d56521;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK48f9ed49;rport
From: "S1009070130598600051" <sip:0000000000@66.71.249.xxx>;tag=as5b06aee9
To: <sip:201@71.203.248.167:12100;rinstance=fd0181fdb1d56521;cpd=on>
Contact: <sip:0000000000@66.71.249.xxx>
Call-ID: 2c55d3b601d73809171e79e43113c30b@66.71.249.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1009070130598600051" <sip:0000000000@66.71.249.xxx>;privacy=off;screen=no
Date: Tue, 07 Sep 2010 05:30:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 3696 3696 IN IP4 66.71.249.xxx
s=session
c=IN IP4 66.71.249.xxx
t=0 0
m=audio 17054 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep 7 01:31:05] Retransmitting #5 (NAT) to 71.203.248.167:12100:
INVITE sip:201@71.203.248.167:12100;rinstance=fd0181fdb1d56521;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK48f9ed49;rport
From: "S1009070130598600051" <sip:0000000000@66.71.249.xxx>;tag=as5b06aee9
To: <sip:201@71.203.248.167:12100;rinstance=fd0181fdb1d56521;cpd=on>
Contact: <sip:0000000000@66.71.249.xxx>
Call-ID: 2c55d3b601d73809171e79e43113c30b@66.71.249.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1009070130598600051" <sip:0000000000@66.71.249.xxx>;privacy=off;screen=no
Date: Tue, 07 Sep 2010 05:30:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 3696 3696 IN IP4 66.71.249.xxx
s=session
c=IN IP4 66.71.249.xxx
t=0 0
m=audio 17054 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep 7 01:31:07] == Parsing '/etc/asterisk/manager.conf': [Sep 7 01:31:07] Found
[Sep 7 01:31:07] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 01:31:07] == Parsing '/etc/asterisk/manager.conf': [Sep 7 01:31:07] Found
[Sep 7 01:31:07] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 01:31:07] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 01:31:07] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 01:31:07] NOTICE[3764]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/201-00000006
[Sep 7 01:31:07] > Channel SIP/201-00000006 was never answered.
[Sep 7 01:31:07] Scheduling destruction of SIP dialog '2c55d3b601d73809171e79e43113c30b@66.71.249.xxx' in 10752 ms (Method: INVITE)
[Sep 7 01:31:09] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 01:31:12]
<--- SIP read from 71.203.248.167:12100 --->

<------------->
[Sep 7 01:31:13] Reliably Transmitting (NAT) to 71.203.248.167:12100:
OPTIONS sip:201@71.203.248.167:12100;rinstance=fd0181fdb1d56521;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK33caa1e3;rport
From: "asterisk" <sip:asterisk@66.71.249.xxx>;tag=as2686b562
To: <sip:201@71.203.248.167:12100;rinstance=fd0181fdb1d56521;cpd=on>
Contact: <sip:asterisk@66.71.249.xxx>
Call-ID: 5179fac90984ddd32de8db296919ffdd@66.71.249.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 07 Sep 2010 05:31:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 7 01:31:13]
<--- SIP read from 71.203.248.167:12100 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.71.249.xxx:5060;branch=z9hG4bK33caa1e3;rport=5060
Contact: <sip:192.168.1.9:12100>
To: <sip:201@71.203.248.167:12100;rinstance=fd0181fdb1d56521;cpd=on>;tag=65000032
From: "asterisk"<sip:asterisk@66.71.249.xxx>;tag=as2686b562
Call-ID: 5179fac90984ddd32de8db296919ffdd@66.71.249.xxx
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
[Sep 7 01:31:13] --- (12 headers 0 lines) ---
[Sep 7 01:31:13] Really destroying SIP dialog '5179fac90984ddd32de8db296919ffdd@66.71.249.xxx' Method: OPTIONS

by **williamconley** » Tue Sep 07, 2010 7:43 am

does the phone register?

can the server ping the phone?

by **chrisnetronix** » Tue Sep 07, 2010 8:30 am

the xlite phone registers, and I can ping the server

how can you make the server ping the phone

by **williamconley** » Tue Sep 07, 2010 9:14 am

ping --help

by **chrisnetronix** » Tue Sep 07, 2010 4:05 pm

ok, I bypassed my router and vonage, and got the xlite to work, when I click dial the next number on the agent interface it gives this error:

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.

Here is the CLI

<------------>
[Sep 7 16:58:19] Scheduling destruction of SIP dialog '740769e2-fbc29c8-782e923@70.167.153.136' in 32000 ms (Method: OPTIONS)
[Sep 7 16:58:20] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 16:58:23] == Parsing '/etc/asterisk/manager.conf': [Sep 7 16:58:23] Found
[Sep 7 16:58:23] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 16:58:23] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-538e,2", "8600051|F") in new stack
[Sep 7 16:58:23] > Channel Local/8600051@default-538e,1 was answered.
[Sep 7 16:58:23] -- Executing [91xxxxxx7027@default:1] AGI("Local/8600051@default-538e,1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 7 16:58:23] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 7 16:58:23] -- Executing [91xxxxxx7027@default:2] GotoIf("Local/8600051@default-538e,1", "1?10") in new stack
[Sep 7 16:58:23] -- Goto (default,91xxxxxx7027,10)
[Sep 7 16:58:23] -- Executing [91xxxxxx7027@default:10] Set("Local/8600051@default-538e,1", "GROUP()=sgw") in new stack
[Sep 7 16:58:23] -- Executing [91xxxxxx7027@default:11] Dial("Local/8600051@default-538e,1", "SIP/sgw1/12768077027||To") in new stack
[Sep 7 16:58:23] WARNING[3416]: chan_sip.c:3095 create_addr: No such host: sgw1
[Sep 7 16:58:23] Really destroying SIP dialog '0a4e8586436a8ca96426780130518eac@127.0.0.2' Method: INVITE
[Sep 7 16:58:23] WARNING[3416]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Sep 7 16:58:23] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 7 16:58:23] -- Executing [91xxxxxx7027@default:12] GotoIf("Local/8600051@default-538e,1", "1?20") in new stack
[Sep 7 16:58:23] -- Goto (default,91xxxxxx7027,20)
[Sep 7 16:58:23] -- Executing [91xxxxxx7027@default:20] Dial("Local/8600051@default-538e,1", "SIP/sgw2/12768077027||To") in new stack
[Sep 7 16:58:23] WARNING[3416]: chan_sip.c:3095 create_addr: No such host: sgw2
[Sep 7 16:58:23] Really destroying SIP dialog '0e557f8b78661c2b631b79365c7bd9ac@127.0.0.2' Method: INVITE
[Sep 7 16:58:23] WARNING[3416]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Sep 7 16:58:23] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 7 16:58:23] -- Executing [91xxxxxx7027@default:21] Hangup("Local/8600051@default-538e,1", "") in new stack
[Sep 7 16:58:23] == Spawn extension (default, 91xxxxxx7027, 21) exited non-zero on 'Local/8600051@default-538e,1'
[Sep 7 16:58:23] -- Executing [h@default:1] DeadAGI("Local/8600051@default-538e,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Sep 7 16:58:23] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 7 16:58:23] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-538e,2'
[Sep 7 16:58:23] -- Executing [h@default:1] DeadAGI("Local/8600051@default-538e,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 7 16:58:23] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 7 16:58:25] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 16:58:25] Really destroying SIP dialog '887d2e83-2fb8c786-83199b4@216.115.69.131' Method: OPTIONS
[Sep 7 16:58:35] Really destroying SIP dialog '1ad010db36c1513c505e74390226782f@127.0.0.2' Method: REGISTER
[Sep 7 16:58:42] == Parsing '/etc/asterisk/manager.conf': [Sep 7 16:58:42] Found
[Sep 7 16:58:42] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 16:58:44] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 16:58:44]
<--- SIP read from 71.228.160.169:55468 --->

with sip debug on

<------------->
[Sep 7 16:59:48] --- (8 headers 0 lines) ---
[Sep 7 16:59:48] Scheduling destruction of SIP dialog '1ad010db36c1513c505e74390226782f@127.0.0.2' in 32000 ms (Method: REGISTER)
[Sep 7 16:59:48] NOTICE[3764]: chan_sip.c:13365 handle_response_register: Outbound Registration: Expiry for sip.flowroute.com is 120 sec (Scheduling reregistration in 105 s)
[Sep 7 16:59:48] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 16:59:53] == Parsing '/etc/asterisk/manager.conf': [Sep 7 16:59:53] Found
[Sep 7 16:59:53] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 16:59:53] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-80d2,2", "8600051|F") in new stack
[Sep 7 16:59:53] > Channel Local/8600051@default-80d2,1 was answered.
[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:1] AGI("Local/8600051@default-80d2,1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 7 16:59:53] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:2] GotoIf("Local/8600051@default-80d2,1", "1?10") in new stack
[Sep 7 16:59:53] -- Goto (default,91xxxxxx7027,10)
[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:10] Set("Local/8600051@default-80d2,1", "GROUP()=sgw") in new stack
[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:11] Dial("Local/8600051@default-80d2,1", "SIP/sgw1/12768077027||To") in new stack
[Sep 7 16:59:53] WARNING[3587]: chan_sip.c:3095 create_addr: No such host: sgw1
[Sep 7 16:59:53] Really destroying SIP dialog '00a64cf833b57bfd24ae780030d40d4f@127.0.0.2' Method: INVITE
[Sep 7 16:59:53] WARNING[3587]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Sep 7 16:59:53] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:12] GotoIf("Local/8600051@default-80d2,1", "1?20") in new stack
[Sep 7 16:59:53] -- Goto (default,91xxxxxx7027,20)
[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:20] Dial("Local/8600051@default-80d2,1", "SIP/sgw2/12768077027||To") in new stack
[Sep 7 16:59:53] WARNING[3587]: chan_sip.c:3095 create_addr: No such host: sgw2
[Sep 7 16:59:53] Really destroying SIP dialog '1882c22d501f916259730c7e4556f732@127.0.0.2' Method: INVITE
[Sep 7 16:59:53] WARNING[3587]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Sep 7 16:59:53] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:21] Hangup("Local/8600051@default-80d2,1", "") in new stack
[Sep 7 16:59:53] == Spawn extension (default, 91xxxxxx7027, 21) exited non-zero on 'Local/8600051@default-80d2,1'
[Sep 7 16:59:53] -- Executing [h@default:1] DeadAGI("Local/8600051@default-80d2,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Sep 7 16:59:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 7 16:59:53] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-80d2,2'
[Sep 7 16:59:53] -- Executing [h@default:1] DeadAGI("Local/8600051@default-80d2,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 7 16:59:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 7 16:59:55] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 17:00:02] == Parsing '/etc/asterisk/manager.conf': [Sep 7 17:00:02] Found
[Sep 7 17:00:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 17:00:02] == Parsing '/etc/asterisk/manager.conf': [Sep 7 17:00:02] Found
[Sep 7 17:00:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 17:00:02] == Parsing '/etc/asterisk/manager.conf': [Sep 7 17:00:02] Found
[Sep 7 17:00:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 17:00:02] == Parsing '/etc/asterisk/manager.conf': [Sep 7 17:00:02] Found
[Sep 7 17:00:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 17:00:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 17:00:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 17:00:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 17:00:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 17:00:03] Reliably Transmitting (NAT) to 70.167.153.130:5060:
OPTIONS sip:sip.flowroute.com;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.xxx.xxx:5060;branch=z9hG4bK7dd0d6af;rport
From: "asterisk" <sip:asterisk@66.71.xxx.xxx>;tag=as033a47a0
To: <sip:sip.flowroute.com;cpd=on>
Contact: <sip:asterisk@66.71.xxx.xxx>
Call-ID: 6b06047e3f7442cf03b0c06415034065@66.71.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 07 Sep 2010 21:00:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 7 17:00:03]
<--- SIP read from 70.167.153.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.71.xxx.xxx:5060;branch=z9hG4bK7dd0d6af;rport=5060
From: "asterisk" <sip:asterisk@66.71.xxx.xxx>;tag=as033a47a0
To: <sip:sip.flowroute.com;cpd=on>;tag=e9901e48385c1aec25ff857f21344e5a.d647
Call-ID: 6b06047e3f7442cf03b0c06415034065@66.71.xxx.xxx
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0

<------------->
[Sep 7 17:00:03] --- (11 headers 0 lines) ---
[Sep 7 17:00:03] Reliably Transmitting (NAT) to 71.228.160.169:55468:
OPTIONS sip:201@71.228.160.169:55468;rinstance=a155489e4f9ab65c;cpd=on SIP/2.0
Via: SIP/2.0/UDP 66.71.xxx.xxx:5060;branch=z9hG4bK349b6c33;rport
From: "asterisk" <sip:asterisk@66.71.xxx.xxx>;tag=as4f165535
To: <sip:201@71.228.160.169:55468;rinstance=a155489e4f9ab65c;cpd=on>
Contact: <sip:asterisk@66.71.xxx.xxx>
Call-ID: 0acf094633767a6100e2dc4d4372a342@66.71.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 07 Sep 2010 21:00:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 7 17:00:03] Really destroying SIP dialog '6b06047e3f7442cf03b0c06415034065@66.71.xxx.xxx' Method: OPTIONS
[Sep 7 17:00:03]
<--- SIP read from 71.228.160.169:55468 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.71.xxx.xxx:5060;branch=z9hG4bK349b6c33;rport=5060
Contact: <sip:71.228.160.169:55468>
To: <sip:201@71.228.160.169:55468;rinstance=a155489e4f9ab65c;cpd=on>;tag=f4450c3d
From: "asterisk"<sip:asterisk@66.71.xxx.xxx>;tag=as4f165535
Call-ID: 0acf094633767a6100e2dc4d4372a342@66.71.xxx.xxx
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
[Sep 7 17:00:03] --- (12 headers 0 lines) ---
[Sep 7 17:00:03] Really destroying SIP dialog '0acf094633767a6100e2dc4d4372a342@66.71.xxx.xxx' Method: OPTIONS
[Sep 7 17:00:07] == Parsing '/etc/asterisk/manager.conf': [Sep 7 17:00:07] Found
[Sep 7 17:00:07] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 17:00:07] == Parsing '/etc/asterisk/manager.conf': [Sep 7 17:00:07] Found
[Sep 7 17:00:07] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 7 17:00:07] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 17:00:07] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 7 17:00:09]
<--- SIP read from 70.167.153.130:5060 --->
OPTIONS sip:66.71.xxx.xxx:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:70.167.153.130;lr>
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bK72c2.4cc6997608371aa1685528ac0f50b3a1.0
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
Route: <sip:70.167.153.130;lr;received="sip:66.71.xxx.xxx:5060">
From: sip:ping@invalid;tag=850eda74
To: sip:66.71.xxx.xxx:5060
Call-ID: 740769e2-3ba39c8-5f2e923@70.167.153.136
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
[Sep 7 17:00:09] --- (11 headers 0 lines) ---
[Sep 7 17:00:09] Looking for s in trunkinbound (domain 66.71.xxx.xxx)
[Sep 7 17:00:09]
<--- Transmitting (NAT) to 70.167.153.130:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bK72c2.4cc6997608371aa1685528ac0f50b3a1.0;received=70.167.153.130
Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0
From: sip:ping@invalid;tag=850eda74
To: sip:66.71.xxx.xxx:5060;tag=as0610ab0e
Call-ID: 740769e2-3ba39c8-5f2e923@70.167.153.136
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0

Not sure what is happening any help here would be great

by **williamconley** » Tue Sep 07, 2010 4:33 pm

[Sep 7 16:59:53] -- Executing [91xxxxxx7027@default:11] Dial("Local/8600051@default-80d2,1", "SIP/sgw1/12768077027||To") in new stack
[Sep 7 16:59:53] WARNING[3587]: chan_sip.c:3095 create_addr: No such host: sgw1

did you see this portion of your post?

what do you have in "account entry" for your carrier?

by **chrisnetronix** » Tue Sep 07, 2010 4:57 pm

Flowroute carrier info:

Carrier ID: Flowroute
Carrier Name: flowroute
Carrier Description: test
Registration String: register => myusername:mypassowrd@sip.flowroute.com
Template ID:
Account Entry:

[flowroute]
type=friend
secret=xxxxxxxxx
username=xxxxxxx
host=sip.flowroute.com
dtmfmode=inband
rfc2833compensate=yes
context=inbound
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=sip.flowroute.com

Protocol: Sip
Globals String: FRSIPTRUNK = SIP/testcarrier
Dialplan Entry:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@flowroute,55,tTo)
exten => _91NXXNXXXXXX,3,Hangup

Server IP: 66.71.xxx.xxx

vicidial.org

Xlite runs but agent cannot login due to no sessions

Xlite runs but agent cannot login due to no sessions

Who is online