vicidial.org

I have setup message broadcasting on vicidial and use xlite to record the message, yet for some odd reason when recording, then playing back the message on xlite, the call drops. and when I run the broadcast and vicidial calls the numbers the phone rings but I do not hear the message then it says goodbye?

here is the cli when the call drops from playing back the message in xlite:

[Aug 27 00:35:08] -- Registered SIP '102' at 71.203.248.167 port 26840
[Aug 27 00:35:08] -- Saved useragent "X-Lite release 1104o stamp 56125" for peer 102
[Aug 27 00:35:08] -- Unregistered SIP '102'
[Aug 27 00:35:08] -- Registered SIP '102' at 71.203.248.167 port 26840
[Aug 27 00:35:08] -- Saved useragent "X-Lite release 1104o stamp 56125" for peer 102
[Aug 27 00:35:26] -- Executing [8168@default:1] Answer("SIP/102-000000a7", "") in new stack
[Aug 27 00:35:26] -- Executing [8168@default:2] AGI("SIP/102-000000a7", "agi-record_prompts.agi|gsm-----7 20000") in new stack
[Aug 27 00:35:26] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-record_prompts.agi
[Aug 27 00:35:26] -- Playing 'ld_welcome_pin_number' (escape_digits=123456789) (sample_offset 0)
[Aug 27 00:35:33] -- Playing 'beep' (escape_digits=) (sample_offset 0)
[Aug 27 00:35:33] -- Playing 'vm-rec-generic' (escape_digits=) (sample_offset 0)
[Aug 27 00:35:37] -- Playing 'beep' (escape_digits=) (sample_offset 0)
[Aug 27 00:35:43] -- Playing 'beep' (escape_digits=) (sample_offset 0)
[Aug 27 00:35:43] -- Playing 'vm-review' (escape_digits=123456789) (sample_offset 0)
[Aug 27 00:35:46] WARNING[3730]: chan_sip.c:2015 retrans_pkt: Maximum retries exceeded on transmission NWMxYj ZmZGZjZDY1YTBmODRlM2RjNjZlMDk1NGQ4Mzg. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
[Aug 27 00:35:46] WARNING[3730]: chan_sip.c:2037 retrans_pkt: Hanging up call NWMxYjZmZGZjZDY1YTBmODRlM2RjNjZ lMDk1NGQ4Mzg. - no reply to our critical packet (see doc/sip-retransmit.txt).
[Aug 27 00:35:46] -- Executing [h@default:1] DeadAGI("SIP/102-000000a7", "agi://127.0.0.1:4577/call_log-- HVcauses--PRI-----NODEBUG-----18---------------") in new stack
[Aug 27 00:35:46] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- ----- completed, returning 0
[Aug 27 00:36:01] == Parsing '/etc/asterisk/manager.conf': [Aug 27 00:36:01] Found
[Aug 27 00:36:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 27 00:36:01] == Parsing '/etc/asterisk/manager.conf': [Aug 27 00:36:01] Found
[Aug 27 00:36:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 27 00:36:01] == Parsing '/etc/asterisk/manager.conf': [Aug 27 00:36:01] Found
[Aug 27 00:36:01] == Manager 'sendcron' logged on from 127.0.0.1



I am not sure if it has to be because of this other issue I am having with the agent login showing no available sessions refer here:

http://www.vicidial.org/VICIDIALforum/v ... hp?t=13075

I pinged the server with a 1000 ping and I am not getting packet loss, so Where I am going wrong....

Any help is greatly appreciated

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)

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 |

have you successfully made a phone call with this sip phone? with sound in both directions?

I can dial into asterisk to record a message but it wont make a call directly from xlite.

when I do try to play back the message i record it cuts off about 20 seconds into making the call to asterisk,

does your x-lite phone "register" successfully?

can you call into a conference in asterisk and hear "you are the only person in this conference?"

is your phone server on the same physical network and subnet as your x-lite phone?

yes it registers only when I use the password test for 102, if I change 102's password in vicidial to something like newtest, then put that password in xlite, it gives me a 403, its like xlite is not reading the new password givin, but I used the password test, and it registered, and I dialed the conference extension 8600001 and it says nothing

the ip address in the domain field of xlite is the same as the server ip address in vicidial

williamconley wrote:

williamconley wrote:is your phone server on the same physical network and subnet as your x-lite phone?

the question does not pertain to "what it's set to" to try to determine if you put in the correct ip. the question is to determine the topography of your network and the location of your x-lite phone (when geographically compared to your vicidial server) to see if you have a firewall or other possible network complications to your x-lite interaction with your vicidial.

"is your phone server on the same physical network..."?

no, this is a dedicated server colocated at a another destination, I turned my firewall off and it still does not login with the new credentials i set for the password.

irregardless, I will be doing another install tomorrow of vicidial like you instructed earlier to see if this resolves any issues.

chrisnetronix wrote:I turned my firewall off

just to be clear ... when one "turns off" one's firewall, one normally has NO access to the internet.

if you mean you now have no firewall, then are you using an IP on the WWW instead of a "local" IP? (as in: your IP address no longer begins with 192.168 or 10?) for BOTH your x-lite running computer AND your server?

if that is NOT what you mean, then you may have turned off the security portion of your firewall, but you have not turned off Network Address Translation (NAT), which is likely at least one of the problems which you are experiencing. SIP does not appreciate NAT, and has very little patience for double NAT (when there is NAT at both the server and client locations). IAX, however, does not seem to mind so much. (KIAX, Zoiper are both handy products for this reason).

FYI

I am using xlite on a personal computer from home, and yes the internet works if I turn it off.

I do have a router installed here.

Whats your thoughts.

chrisnetronix wrote:I am using xlite on a personal computer from home, and yes the internet works if I turn it off.

I do have a router installed here

what is the "it" you turned off? (be very specific)

my firewall, I use norton

that's a mere security device. not a "firewall". your firewall is in your router. it remaps data packets and STOPS access to your local network from outside.

it is often referred to as a "personal firewall" as a sales tactic, which is nice, but is really only necessary if you have a laptop and take it outside your local network.

while ON your local network, your router is also your firewall. noone can get past it ... it creates an internal network (192.168.x.x) completely separate from the world wide web. It performs network translation of data packets and decides who on the inside ASKED for each packet (since the web site you asked for this information has no idea that your local IP address is 192.168.10.5, the ROUTER, with the built-in firewall, figures all this out).

That's likely to be one of your issues.

Next time you do your installation ... use the manual to set up the system one step at a time, and STOP the moment something isn't perfect and ask for help. That way you don't end up with a completely built system but the basics not taken care of.

You could even go back NOW and try this.

Before your SIP phone and SIP trunk work, ANYTHING else you do is likely a waste of your time.

is their any type on configuration needed for my router to allow xlite to work correctly?

does your server also have a router? (does it have an IP address that is both internal AND on the world wide web, or is the IP address of your server a "local" ip address? enter "ifconfig" and get the IP, remove the last two digits and post it if you are not sure).

i say this because if it has a router, you are double NAT'ing and it's about to get complex. (so get yourself a copy of Zoiper and use IAX instead)

eth0 Link encap:Ethernet HWaddr 00:25:90:06:B5:86
inet addr:66.71.249.1xx Bcast:66.71.249.1xx Mask:255.255.255.2xx
inet6 addr: fe80::225:90ff:fe06:b586/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:152048 errors:0 dropped:0 overruns:0 frame:0
TX packets:149989 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:100
RX bytes:43661418 (41.6 Mb) TX bytes:53922165 (51.4 Mb)
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:5191211 errors:0 dropped:0 overruns:0 frame:0
TX packets:5191211 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:384543367 (366.7 Mb) TX bytes:384543367 (366.7 Mb)

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

http://www.vicidial.org/VICIDIALforum/v ... 2849#52849

please do not double post