Elastix + Cisco Callmanager 4

agenovez

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#1
Hi Folks, I am tired to install, but I learned a lot, here is my problem:

I followed the steps from this page:

http://www.hurdman.net/mirror/voip-info ... ion-2.html

Everything I put on sip_custom.conf and extensions_custom.conf

One trick I learned, this is very important


[callman01]
type=friend
host=192.168.10.10
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
context=from-internal (this is veryyyy important!!!)

I managed to make the link active:

elastix*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
1001/1001 172.40.22.246 D N 61973 OK (3 ms)
callman01 192.168.10.10 5060 OK (11 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
elastix*CLI>


But when I try to make calls it says

-- Executing [100@from-internal:1] Macro("SIP/1001-091ad010", "dialout-callmanager|100") in new stack
-- Executing [s@macro-dialout-callmanager:1] ChanIsAvail("SIP/1001-091ad010", "SIP/callman01") in new stack
== Spawn extension (macro-dialout-callmanager, s, 2) exited non-zero on 'SIP/1001-091ad010' in macro 'dialout-callmanager'
== Spawn extension (from-internal, 100, 1) exited non-zero on 'SIP/1001-091ad010'
-- Executing [h@from-internal:1] Macro("SIP/1001-091ad010", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1001-091ad010", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1001-091ad010", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1001-091ad010", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1001-091ad010", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1001-091ad010", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1001-091ad010", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1001-091ad010' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-091ad010'

My extension_custom.conf:

exten => _3XXX,1,Macro(dialout-callmanager,${EXTEN})

[macro-dialout-callmanager]
exten => s,1,ChanIsAvail(SIP/callman01)
exten => s,2,Cut(AVAILCHAN=AVAILCHAN,,1)
exten => s,3,Dial(${AVAILCHAN}/${ARG1})
exten => s,4,Hangup
exten => s,102,Congestion



[from-internal-custom]
exten => _XXX,1,Macro(dialout-callmanager,${EXTEN})
exten => _5NXXX,1,Macro(dialout-callmanager,${EXTEN})
exten => _51NXXNXXXXXX,1,Macro(dialout-callmanager,${EXTEN})
exten => i,1,Congestion



exten => 1234,1,Playback(demo-congrats) ; extensions can dial 1234
exten => 1234,2,Hangup()
exten => h,1,Hangup()
include => agentlogin
include => conferences
include => calendar-event
include => weather-wakeup

[agentlogin]
exten => _*8888.,1,Set(AGENTNUMBER=${EXTEN:5})
exten => _*8888.,n,NoOp(AgentNumber is ${AGENTNUMBER})
exten => _*8888.,n,AgentLogin(${AGENTNUMBER})
exten => _*8888.,n,Hangup()

[mm-announce]
exten => 9999,1,Set(CALLERID(name)="MMGETOUT")
exten => 9999,n,Answer
exten => 9999,n,Playback(conf-will-end-in)
exten => 9999,n,Playback(digits/5)
exten => 9999,n,Playback(minutes)
exten => 9999,n,Hangup

[conferences]
;Used by cbEnd script to play end of conference warning
exten => 5555,1,Answer
exten => 5555,n,Wait(3)
exten => 5555,n,CBMysql()
exten => 5555,n,Hangup

[calendar-event]
exten => _*7899,1,Answer
exten => _*7899,2,Playback(${FILE_CALL})
exten => _*7899,3,Wait(2)
exten => _*7899,4,Hangup()

[weather-wakeup]
exten => *61,1,Answer
exten => *61,2,AGI(nv-weather.php)
exten => *61,3,Hangup
exten => *62,1,Answer
exten => *62,2,AGI(wakeup.php)
exten => *62,3,Hangup


Thanks for any help :)
 

danardf

Joined
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#2
Hmmm.
I did make a trunk sip between an Elastix server and a Callmanager, I haven't problem with.

Only careful, the codec (ulaw) and the SIP (UDP) and that's all.
 

agenovez

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#3
Finally I make it work!!, but.... this is a pain in the a**

I now I don´t have audio :( the extension rings I pick it UP but it don´t sound anything, what can be wrong???
 

agenovez

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#4
danardf said:
Hmmm.
I did make a trunk sip between an Elastix server and a Callmanager, I haven't problem with.

Only careful, the codec (ulaw) and the SIP (UDP) and that's all.
Thanks PAL
 

agenovez

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#5
It Works!!! I managed to disable my computer firewall hehehe

Now I have diferent Device Pools in my office, but it only works with One, what can I do?
 

agenovez

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#6
Hi, finally I made it possible to make calls, but I have a problem, I don´t know if it is a codec related problem



Please, help, I can make calls to my cisco extensions, but when I try to make a call from the cisco extensions to Elastix the call is dropped.

Greetings
 

dicko

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#7
If you use g729 (and you chose that codec on the Cisco) you need a license(s) from Digium (10 dollars each) for each connection to elastix (pass-though will work, but MOH and conferences and connections will not, ) or find some other g729 module.

I suggest you use g711 as you do TO the manager.
 

agenovez

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#8
Hi Thanks, that is weird, I can make calls TO the Cisco, but not from the CISCO to Elastix, it says this:

[Jun 22 15:30:36] WARNING[3475] chan_sip.c: Maximum retries exceeded on transmission 42818d80-1df17d70-12d5-a0aa8c0@192.168.10.10 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt.
[Jun 22 15:30:36] WARNING[3475] chan_sip.c: Hanging up call 42818d80-1df17d70-12d5-a0aa8c0@192.168.10.10 - no reply to our critical packet (see doc/sip-retransmit.txt).

Is this the problem with the codec???
 

agenovez

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#9
Finally I managed to make the Phorne(Phone ;)) work, It was a codec Issue, hmmm I am going to install g729 on elastix.

Thanks for everything, I made a change here:

/etc/asterisk/sip_general_custom.conf

t1min=500

Then that problem of timeout went away :)
 

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