Problem after install Elastix 1.1

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#1
I went to TB2.6 to Elastix 1.1 and ho surprise.!

I have informed all extensions SIP (as before), and I can not call that the extensions Xlite ....
All other extensions as linksys (SPA3102 PAP2), do not work. (occupied)

I have not changed my setup

Besides, I have the option FR in sip.conf, but the sounds are still EN!

Result ... not Elastix not working properly.

I'm disappointed, because before it was operating under TB26.<br><br>Post edited by: danardf, at: 2008/06/22 21:54
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#2
hmmm.
For information, this problem is for exemple:
phone 102 (Xlite) calling 103 (Linksys)
I have this message:
The..nanana..is not aviable, please try again.

On 103 extension there is not VM.
If VM is enabled, my call is hangup.

problem into dialplan? :unsure:

Help me please. :(
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#3
danardf said:
Besides, I have the option FR in sip.conf, but the sounds are still EN!
I reload the french voices into sounds root diretory (var/lib/asterisk/sounds) is ok.
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#4
With sip debug peeer 103:

-- Executing [s@macro-dial:3] AGI("SIP/102-09e371a8", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Franck Distant' number is '102'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 103 to extension map
-- dialparties.agi: Extension 103 cf is disabled
-- dialparties.agi: Extension 103 do not disturb is disabled
> dialparties.agi: extnum 103 has: cw: 1; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
-- dialparties.agi: dbset CALLTRACE/103 to 102
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:10] Dial("SIP/102-09e371a8", "SIP/103||tr") in new stack
-- Couldn't call 103
Scheduling destruction of SIP dialog '6c98af621996e1086adf671131656e16@193.107.20.38' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dial:11] Set("SIP/102-09e371a8", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:10] Set("SIP/102-09e371a8", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/102-09e371a8", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/102-09e371a8", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:13] Set("SIP/102-09e371a8", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:14] NoOp("SIP/102-09e371a8", "Voicemail is novm") in new stack
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/102-09e371a8", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] PlayTones("SIP/102-09e371a8", "congestion") in new stack
 

Bob

Joined
Nov 4, 2007
Messages
2,400
Likes
1
Points
36
#5
danardf said:
I went to TB2.6 to Elastix 1.1 and ho surprise.!
Danardf,

How did you migrate??

Was it a completely fresh install of Elastix 1.1 with everything put back by hand in the GUI??
OR
Did you restore the Freepbx backup from TB2.6 to the new Elastix 1.1 install?

Regards

Bob<br><br>Post edited by: Bob, at: 2008/06/23 02:00
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#6
yes, it's a fresh install with everything put back by hand in the GUI.

Can i update asterisk version 1.4.21.
Recompiling option "make sample"?

If you have some solution....

Thanks
 

lek

Guest
#7
danardf,

Are you sure you typed "sip debug..."?. I can't see the sip packages in your text.

Additionally, please post the output of the command "show hints" on the CLI.
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#8
i'm very sure!
sip debug peer 103

show hints :
host*CLI> show hints
host*CLI>
-= Registered Asterisk Dial Plan Hints =-
105@ext-local : SIP/105 State:Idle Watchers 0
104@ext-local : SIP/104 State:Idle Watchers 0
103@ext-local : SIP/103 State:Idle Watchers 0
102@ext-local : SIP/102 State:Idle Watchers 0
101@ext-local : SIP/101 State:Idle Watchers 0
100@ext-local : SIP/100 State:Unavailable Watchers 0
----------------
- 6 hints registered
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#9
edgar said:
danardf,
I can't see the sip packages in your text.
ha yes.... excuse me.
it's a part of sip debug.

the complete trace is:

host*CLI> sip debug peer 103
SIP Debugging Enabled for IP: 193.107.20.60:5062
host*CLI>
<--- SIP read from 193.107.20.60:5062 --->
NOTIFY sip:193.107.20.38 SIP/2.0
Via: SIP/2.0/UDP 193.107.20.60:5062;branch=z9hG4bK-89fa5a7c
From: Central <sip:103@193.107.20.38>;tag=6dc9b7f2fa248bddo0
To: <sip:193.107.20.38>
Call-ID: ce867a06-32690d9@193.107.20.60
CSeq: 626 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 193.107.20.60 : 5062 (NAT)

<--- Transmitting (NAT) to 193.107.20.60:5062 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 193.107.20.60:5062;branch=z9hG4bK-89fa5a7c;received=193.107.20.60
From: Central <sip:103@193.107.20.38>;tag=6dc9b7f2fa248bddo0
To: <sip:193.107.20.38>;tag=as18ec76fe
Call-ID: ce867a06-32690d9@193.107.20.60
CSeq: 626 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
-- Executing [103@from-internal:1] Macro("SIP/102-b7706680", "exten-vm|novm|103") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/102-b7706680", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/102-b7706680", "user-callerid: device 102") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/102-b7706680", "AMPUSER=102") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/102-b7706680", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] GotoIf("SIP/102-b7706680", "0?start") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/102-b7706680", "REALCALLERIDNUM=102") in new stack
-- Executing [s@macro-user-callerid:6] NoOp("SIP/102-b7706680", "REALCALLERIDNUM is 102") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/102-b7706680", "AMPUSER=102") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/102-b7706680", "AMPUSERCIDNAME=Franck Distant") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/102-b7706680", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/102-b7706680", "AMPUSERCID=102") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/102-b7706680", "CALLERID(all)="Franck Distant" <102>") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/102-b7706680", "REALCALLERIDNUM=102") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/102-b7706680", "TTL: ARG1: novm") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/102-b7706680", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/102-b7706680", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/102-b7706680", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/102-b7706680", "Using CallerID "Franck Distant" <102>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/102-b7706680", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/102-b7706680", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/102-b7706680", "EXTTOCALL=103") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/102-b7706680", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/102-b7706680", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/102-b7706680", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/102-b7706680", "record-enable|103|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/102-b7706680", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/102-b7706680", "recordingcheck|20080623-094035|1214206835.42") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080623-094035|1214206835.42: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/102-b7706680", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/102-b7706680", "dial||tr|103") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/102-b7706680", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/102-b7706680", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Franck Distant' number is '102'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 103 to extension map
-- dialparties.agi: Extension 103 cf is disabled
-- dialparties.agi: Extension 103 do not disturb is disabled
> dialparties.agi: extnum 103 has: cw: 1; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
-- dialparties.agi: dbset CALLTRACE/103 to 102
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:10] Dial("SIP/102-b7706680", "SIP/103||tr") in new stack
-- Couldn't call 103
Scheduling destruction of SIP dialog '5e12469c4448091e34afcfe03f8ba8c1@193.107.20.38' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dial:11] Set("SIP/102-b7706680", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:10] Set("SIP/102-b7706680", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/102-b7706680", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/102-b7706680", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:13] Set("SIP/102-b7706680", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:14] NoOp("SIP/102-b7706680", "Voicemail is novm") in new stack
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/102-b7706680", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] PlayTones("SIP/102-b7706680", "congestion") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 1) exited non-zero on 'SIP/102-b7706680' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 1) exited non-zero on 'SIP/102-b7706680'
host*CLI>
oupss. :unsure:
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#10
i have to make an asterisk update to 1.4.21 + addons.
Still the same problem.
I had to make a backup Elastix and after update, i make a restore the last configuration.

wath's append?

I do not want to return to TB2.6
 

CleveJ

Joined
Nov 12, 2007
Messages
100
Likes
0
Points
0
#11
danardf said:
i have to make an asterisk update to 1.4.21 + addons.
Still the same problem.
I had to make a backup Elastix and after update, i make a restore the last configuration.

wath's append?

I do not want to return to TB2.6
Hi Danardf,

I do understand how you feel, but I can assure you when you have it all up and running you will be so pleased and happy you will think of nothing else but Elastix, so please don't be upset we here will assist you one way or another.

I am doing a fresh install of Elastix 1.1 Stable right now to test it as I go. I do not have any of the problems you have at all it works without any problems.

So lets start!!

Can you please answer the following.

1. With your fresh install did you update FreePBX? if so what did you update it to.
2. From a extension can you dial *60 and see if it will repeat the time.

Cheers<br><br>Post edited by: CleveJ, at: 2008/06/23 18:10
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#12
ok thanks for your help ;)
The problem is that, i don't have the phone now because the Elastix is down.
But..... i think this gonna be ok, with your help. :)

i try to looking for by google and i think that the problem become chan_sip.so like this link:

http://bugs.digium.com/view.php?id=9329

For answer at your question, yes after installed Elastix, i had to make an >yum update. and make also freepbx.

after execute the debug methode by digium, i have this part:

[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: AGI
-- Executing [s@macro-dial:10] Dial("SIP/102-082ec560", "SIP/105||tr") in new stack
[Jun 23 13:43:50] DEBUG[3324]: chan_sip.c:3287 update_call_counter: Call to peer '105' is 1 out of 50
[Jun 23 13:43:50] WARNING[3324]: chan_sip.c:3024 sip_call: No audio format found to offer. Cancelling call to 105
-- Couldn't call 105
[Jun 23 13:43:50] DEBUG[3324]: chan_sip.c:3261 update_call_counter: Call to peer '105' removed from call limit 50
Scheduling destruction of SIP dialog '4970d3fc7a0501e24c6f1fc026428bab@193.107.20.38' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: Dial
-- Executing [s@macro-dial:11] Set("SIP/102-082ec560", "DIALSTATUS=CHANUNAVAIL") in new stack
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: Set
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: Macro
-- Executing [s@macro-exten-vm:10] Set("SIP/102-082ec560", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: Set
-- Executing [s@macro-exten-vm:11] GosubIf("SIP/102-082ec560", "0?docfu|1") in new stack
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: GosubIf
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/102-082ec560", "0?docfb|1") in new stack
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: GosubIf
-- Executing [s@macro-exten-vm:13] Set("SIP/102-082ec560", "DIALSTATUS=CHANUNAVAIL") in new stack
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: Set
-- Executing [s@macro-exten-vm:14] NoOp("SIP/102-082ec560", "Voicemail is novm") in new stack
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: NoOp
-- Executing [s@macro-exten-vm:15] GotoIf("SIP/102-082ec560", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
[Jun 23 13:43:50] DEBUG[3324]: app_macro.c:337 _macro_exec: Executed application: GotoIf
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] PlayTones("SIP/102-082ec560", "congestion") in new stack
[Jun 23 13:43:50] WARNING[3324]: channel.c:2779 set_format: Unable to find a codec translation path from ilbc to slin
[Jun 23 13:43:50] WARNING[3324]: indications.c:121 playtones_alloc: Unable to set 'SIP/102-082ec560' to signed linear format (write)
[Jun 23 13:43:50] NOTICE[3324]: res_indications.c:212 handle_playtones: Unable to start playtones
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 1) exited non-zero on 'SIP/102-082ec560' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 1) exited non-zero on 'SIP/102-082ec560'
[Jun 23 13:43:50] DEBUG[3324]: chan_sip.c:3261 update_call_counter: Call from peer '102' removed from call limit 50
[Jun 23 13:43:51] WARNING[3297]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol Packet
[Jun 23 13:43:51] WARNING[3297]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol Packet
[Jun 23 13:43:51] WARNING[3297]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol Packet
[Jun 23 13:43:54] WARNING[3297]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol Packet
[Jun 23 13:43:54] WARNING[3297]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol Packet
[Jun 23 13:43:55] WARNING[3297]: chan_sip.c:6797 determine_firstline_parts: Bad request protocol Packet
Really destroying SIP dialog '4970d3fc7a0501e24c6f1fc026428bab@193.107.20.38' Method: INVITE
I must to reinstall quickly elastix 1.1 and not updating system.
From a extension can you dial *60 and see if it will repeat the time.
And, the same time, i try this also.
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#13
hmmmm whooooo:

I have a way....

I try to change some codecs by extension and now is ok.
But...but....

In my exemple (befor)
(102) - codec = ilbc, gsm only
(103) - codec = alaw only.

the calls is ok. with old configuration.

I must to put alaw into (102) for the call is up.

funy no?

Normaly, call 102 to 103.
102 (ilbc)---> asterisk translate ilbc to alaw and ----> 103 (Alaw)
So 2 codecs alive. NO?
 

CleveJ

Joined
Nov 12, 2007
Messages
100
Likes
0
Points
0
#14
danardf said:
hmmmm whooooo:

I have a way....

I try to change some codecs by extension and now is ok.
But...but....

In my exemple (befor)
(102) - codec = ilbc, gsm only
(103) - codec = alaw only.

the calls is ok. with old configuration.

I must to put alaw into (102) for the call is up.

funy no?

Normaly, call 102 to 103.
102 (ilbc)---> asterisk translate ilbc to alaw and ----> 103 (Alaw)
So 2 codecs alive. NO?
Good to see you have it fixed, you must remember that the SPA device does not support the ilbc or gsm codec, xlite has the ilbc codec, so the best way to do this is to have all you LAN extensions using g711 u or a as they are all local you should be ok.
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#15
i'm ok with you, but befor (with the last configuration on TB26), i had not this problem.

Asterisk make a twin compression. (ILBC<--->ALAW).

I want to find this

For exemple, if i have some interanet extension on the LAN ans one extension on the WAN with a little band switch. I must to use only one codec gsm on ilbc.

I must to do make a call from (102 / gsm) to (SPA3102 / Alaw), .... no? otherise it's unthinkable.
 

CleveJ

Joined
Nov 12, 2007
Messages
100
Likes
0
Points
0
#16
danardf said:
i'm ok with you, but befor (with the last configuration on TB26), i had not this problem.

Asterisk make a twin compression. (ILBC<--->ALAW).

I want to find this

For exemple, if i have some interanet extension on the LAN ans one extension on the WAN with a little band switch. I must to use only one codec gsm on ilbc.

I must to do make a call from (102 / gsm) to (SPA3102 / Alaw), .... no? otherise it's unthinkable.
Hi Danardf,

I have found your problem, Elastix 1.1 Stable does not have the ilbc codec so this is your problem. What you have to do is install the codec and reboot your Elastix server and it should work. To make sure at Cli do "core show translation" you will see tha ilbc has all dashes (-).

Please post back soon as I have to go to bed soon.

Cheers
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#17
i replace all ilbc codec by gsm and speex.
All is ok now.... :)

pfffiuuuuu.

One more try, outing by SPA. (Trunk SPA3102)

Ok thanks very very much.:woohoo:

I didn't think that ilbc was not installed. :(

info. the result core:

host*CLI> core show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 3 - 15 - 2 -
ulaw - 2 - 1 2 2 1 3 - 15 - 2 -
alaw - 2 1 - 2 2 1 3 - 15 - 2 -
g726aal2 - 2 2 2 - 2 1 3 - 15 - 1 -
adpcm - 2 2 2 2 - 1 3 - 15 - 2 -
slin - 1 1 1 1 1 - 2 - 14 - 1 -
lpc10 - 2 2 2 2 2 1 - - 15 - 2 -
g729 - - - - - - - - - - - - -
speex - 3 3 3 3 3 2 4 - - - 3 -
ilbc - - - - - - - - - - - - -
g726 - 2 2 2 1 2 1 3 - 15 - - -
g722 - - - - - - - - - - - - -
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#18
and now.....
I can make outgoing call by my SPA3102, but i can't take incoming call. :angry:

with trace debug, like this:

CLI>sip debug peer freebox
SIP Debugging Enabled for IP: 193.107.20.62:5060
host*CLI>
<--- SIP read from 193.107.20.62:5060 --->
INVITE sip:pstn@193.107.20.38 SIP/2.0
Via: SIP/2.0/UDP 193.107.20.62:5060;branch=z9hG4bK-43b2ada0;rport
From: pstn <sip:0611582914@193.107.20.38>;tag=ea5994ad33d05118o1
To: <sip:pstn@193.107.20.38>
Call-ID: 1d82e811-4067504@193.107.20.62
CSeq: 101 INVITE
Max-Forwards: 70
Contact: pstn <sip:0611582914@193.107.20.62:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 127842 127842 IN IP4 193.107.20.62
s=-
c=IN IP4 193.107.20.62
t=0 0
m=audio 12570 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 11 lines) ---
Sending to 193.107.20.62 : 5060 (NAT)
Using INVITE request as basis request - 1d82e811-4067504@193.107.20.62
Found peer 'Freebox'
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 193.107.20.62:12570
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 193.107.20.62:12570
Looking for pstn in from-trunk (domain 193.107.20.38)

<--- Reliably Transmitting (no NAT) to 193.107.20.62:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 193.107.20.62:5060;branch=z9hG4bK-43b2ada0;received=193.107.20.62;rport=5060
From: pstn <sip:0611582914@193.107.20.38>;tag=ea5994ad33d05118o1
To: <sip:pstn@193.107.20.38>;tag=as22517aaf
Call-ID: 1d82e811-4067504@193.107.20.62
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1d82e811-4067504@193.107.20.62' in 7744 ms (Method: INVITE)
host*CLI>
<--- SIP read from 193.107.20.62:5060 --->
ACK sip:pstn@193.107.20.38 SIP/2.0
Via: SIP/2.0/UDP 193.107.20.62:5060;branch=z9hG4bK-43b2ada0;rport
From: pstn <sip:0611582914@193.107.20.38>;tag=ea5994ad33d05118o1
To: <sip:pstn@193.107.20.38>;tag=as22517aaf
Call-ID: 1d82e811-4067504@193.107.20.62
CSeq: 101 ACK
Max-Forwards: 70
Contact: pstn <sip:0611582914@193.107.20.62:5060>
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1d82e811-4067504@193.107.20.62' Method: ACK
in my trunk freebox, i had insecure=very, i change by invite, but no result. :unsure:

i don't understand that's append :unsure:
there is a "SIP/2.0 404 Not Found" into the trace SIP.!!!
 

Bob

Joined
Nov 4, 2007
Messages
2,400
Likes
1
Points
36
#19
I have found your problem, Elastix 1.1 Stable does not have the ilbc codec so this is your problem
Just in case anyone else is reading this thread.

It is Asterisk that has removed the inclusion of the ILBC Codec as of a particular version 1.4.19. More info at:

http://www.asterisk.org/node/48469

Naturally if you were running it with Trixbox, then they may have an earlier version of Asterisk e.g. pre 1.4.19, or you may have updated that kept it included from 1.4.18 to 1.4.19...who knows...

But unless you have a pressing need for ILBC, then best to ignore it entirely....

Regards
Bob
 

danardf

Joined
Dec 3, 2007
Messages
8,069
Likes
10
Points
88
#20
yes... now it's ok.

The problem was to SPA configuration.
into PSTN line, dialplan:
befor - (S0<:pstn@193.107.20.38>)
after - (S0<:193.107.20.38>)

Incoming Settings

allow=alaw
context=from-trunk
disallow=all
host=193.107.20.38
insecure=invite
nat=no
qualify=yes
type=user
Allow Anonymous Inbound SIP Calls = yes

There are a little difference between TB26 and Elastix.!!!

Thanks for your help
 

Members online

No members online now.

Latest posts

Forum statistics

Threads
30,902
Messages
130,887
Members
17,565
Latest member
omarmenichetti
Top