system toolbox » voip

ref #14419

UDP portovi koje treba forwardovati na asterisk.rmlh.ba

    * UDP SIP - 5060
    * UDP RTP - 8000-11000

ip adresa asteriska

192.168.77.1

podešen refresh_ip za osvježavanje ns.sigma-com.net

root@router-rmlh-1:~# vi /etc/config/refresh_ip

config cn rmlh
        option cn "mail adsl" 
        option cn "zimbra mail" 
        option cn "jabber adsl" 
        option cn "asterisk adsl"                 
        option cn "mail-50 mail-50.sigma-com.net." 
        option cn ". adsl" 

testirao ..OK

root@router-rmlh-1:/etc# ./refresh_ip.sh 
ip  rmlh.ba = 89.146.181.117
restarting name server
tmp.zone                                                              100%  415     0.4KB/s   00:00    
feedback=_2_
root@router-rmlh-1:/etc#

ping sa ns-a

[root@ernadh ~]# ping asterisk.rmlh.ba
PING adsl.rmlh.ba (89.146.181.117) 56(84) bytes of data.
64 bytes from 89.146.181.117: icmp_seq=0 ttl=51 time=179 ms

ubacio novi firewall na archive.rmlh.ba:/data/router-rmlh-1/etc

resetovao router ...OK sada treba vidjeti dali to fercera izvana

moze se koristiti

ext: 22
pwd: ****

server: asterisk.rmlh.ba
port: 5060
username: 22
pwd: rmlh

u logu vidim da se hernad pokušava zakačit

Jun 19 10:16:18 router-rmlh-1 user.info kernel: RULE 1 -- ACCEPT IN=ppp0 OUT=br-lan SRC=89.146.156.214 DST=192.168.77.1 LEN=954 TOS=0x00 PREC=0x60 TTL=61 ID=91 PROTO=UDP SPT=5060 DPT=5060 LEN=934 
Jun 19 10:16:18 router-rmlh-1 user.info kernel: RULE 1 -- ACCEPT IN=ppp0 OUT=br-lan SRC=89.146.156.214 DST=192.168.77.1 LEN=591 TOS=0x00 PREC=0x60 TTL=61 ID=93 PROTO=UDP SPT=5060 DPT=5060 LEN=571

evo u cli konzoli se desilo ovo

    -- Registered SIP '22' at 192.168.45.151 port 5060 expires 800
    -- Saved useragent "Aastra 53i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5" for peer 22

niste vi ovo dobro podesili, na ovoj ekstenziji treba podesiti nat

evo kako izgleda moj sip.conf

[general]
srvlookup=yes
context=demo
;allowoverlap=no                        ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
canreinvite=yes
;srvlookup=yes
language=bs
localnet=192.168.45.0/255.255.255.0   <<<<<<<<<<<<<<< ovdje treba *.77.* za rmlh
;externip=89.146.152.88               <<<<<<<<<<<<<<< za početak može da se stavi trenutna ip rmlh,
externhost=internet.sigma-com.net     <<<<<<<<<<<<<<  dugoročno, ovo treba podesiti na isti način kako sam ja to podesio  
;Specify how often (in seconds) a hostname DNS lookup should be performed for the value entered in 'externhost'. Default 10 seconds
externrefresh=10
disallow=all
allow=alaw
;allow=gsm
qualify=yes

a sama ekstenzija je ovakva:

[51]
type=friend
username=51
password=supertajnapwd
host=dynamic
callerid="vranici" 
callgroup=3
pickupgroup=3
nat=yes            <<<<<<<<<<<<<<<<<< nat neophodan
qualify=yes

a saša je obavezan da podešenje internet ekstenzije dokumentuje na wiki :)

i kada je sve uredu ja vidim internet adresu tog peer-a:

ifold*CLI> sip show peer 51
ifold*CLI> 

  * Name       : 51
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Context      : demo
  Subscr.Cont. : <Not set>
  Language     : bs
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 3
  Pickupgroup  : 3
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "vranici" <>
  MaxCallBR    : 384 kbps
  Expire       : 30
  Insecure     : no
  Nat          : Always
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 
  Addr->IP     : 77.239.10.223 Port 5060   <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 51
  SIP Options  : (none)
  Codecs       : 0x8 (alaw)
  Codec Order  : (alaw:20)
  Auto-Framing:  No 
  Status       : OK (159 ms)
  Useragent    : IP Phone V1.54.004 CFG0 
  Reg. Contact : sip:51@77.239.10.223:5060

podesio sam sada kod sebe ovdje

evo moj sip.conf

[general]
srvlookup=yes
context=rmlh
;allowoverlap=no        ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
canreinvite=no
localnet=192.168.77.0/255.255.255.0
externip=89.146.135.1   <<<<<<< ovo je trenutna adresa dobivena sa myip.com
externrefresh=10
language=bs

a ekstenzija:

[22]
type=friend
username=22
password=rmlh
host=dynamic
callerid="njemacka2" 
callgroup=2
nat=yes
canreinvite=no

evo i kod mene komadnde

rmlh-1*CLI> sip show peer 22
rmlh-1*CLI>

  * Name       : 22
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Context      : rmlh
  Subscr.Cont. : <Not set>
  Language     : bs
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 2
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "njemacka2" <>
  MaxCallBR    : 384 kbps
  Expire       : 653
  Insecure     : no
  Nat          : Always
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 89.146.156.214 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 22
  SIP Options  : (none)
  Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Auto-Framing:  No
  Status       : Unmonitored
  Useragent    :
  Reg. Contact : sip:22@192.168.45.151

jasko treba podesiti internet.rmlh.ba prema #14426

sada se nešto čudno dešava moj telefon zaglavljuje ?!? pokušam neki poziv i on smrzne :(

jasko treba podesiti internet.rmlh.ba prema #14426

ne kontam dali treba samo na ns.sigma-com.net napraviti za rmlh zonu cn zapis "internet adsl" da bi internet.rmlh.ba resolvirao wan adresu rmlh-a.

evo ga sada ispravka sip.conf

[general]
srvlookup=yes
context=rmlh
;allowoverlap=no        ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
canreinvite=yes
localnet=192.168.77.0/255.255.255.0
externip=89.146.135.1
externrefresh=10
disallow=all  <<<<<<<<<<<<<<< dodao
allow=alaw    <<<<<<<<<<<<<<< dodao
allow=gsm     <<<<<<<<<<<<<<< dodao
language=bs

a na samoj ekstenziji

[22]
type=friend
username=22
password=rmlh
host=dynamic
callerid="njemacka2" 
callgroup=2
nat=yes
canreinvite=no
qualify=yes
disallow=all    <<<<<<<<<<<<< dodao
allow=gsm       <<<<<<<<<<<<< dodao

kada zovem 22 dobijam

    -- Executing [22@rmlh:1] Set("SIP/21-069c7ed0", "ext_to_call=") in new stack
    -- Executing [22@rmlh:2] Macro("SIP/21-069c7ed0", "Get_route_ext_to||22") in new stack
    -- Executing [s@macro-Get_route_ext_to:1] Set("SIP/21-069c7ed0", "ext_to_call=") in new stack
    -- Executing [s@macro-Get_route_ext_to:2] Set("SIP/21-069c7ed0", "default_ext=22") in new stack
    -- Executing [s@macro-Get_route_ext_to:3] GotoIf("SIP/21-069c7ed0", "1?4:5") in new stack
    -- Goto (macro-Get_route_ext_to,s,4)
    -- Executing [s@macro-Get_route_ext_to:4] Set("SIP/21-069c7ed0", "ext_to_call=22") in new stack
    -- Executing [s@macro-Get_route_ext_to:5] NoOp("SIP/21-069c7ed0", "Finish if-Get_route_ext_to-6") in new stack
    -- Executing [22@rmlh:3] Dial("SIP/21-069c7ed0", "SIP/22| 60| tT") in new stack
[Jun 19 12:58:15] WARNING[6474]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [22@rmlh:4] VoiceMail("SIP/21-069c7ed0", "22@default|u") in new stack
    -- <SIP/21-069c7ed0> Playing 'vm-theperson' (language 'bs')
    -- <SIP/21-069c7ed0> Playing 'digits/2' (language 'bs')
    -- <SIP/21-069c7ed0> Playing 'digits/2' (language 'bs')
    -- <SIP/21-069c7ed0> Playing 'vm-isunavail' (language 'bs')
    -- <SIP/21-069c7ed0> Playing 'vm-intro' (language 'bs')

kaze mi da je ext.nedostupna pa mi se javi voicemail

i ovo se u cli konzoli pojavljuje

[Jun 19 12:55:46] NOTICE[27350]: chan_sip.c:15766 sip_poke_noanswer: Peer '22' is now UNREACHABLE!  Last qualify: 181

e sada stoji ono podesenje qualify=yes u samoj ekstenziji... mozda bez toga

imao sam opet probleme sa telefonom, pa sam ga 3-4 puta restartovao, interesanto je da neće da se logira ni na lan ekstenziju

da nije neki bug kada registrujem 2 sip-account-a ... ali to mi opet djeluje nemoguće to je sigurno testirano na ovom telefonu

sada mi izgleda logiran, ... pokušao sam jednom, pa drugi put ... i opet je zaglavio :?!

a koji telefon treba da ide u njemačku ?

koji model mislim

hm ipak nije zaglavio izgleda, ... ali ni sam ne znam kako sam ga odglavio

koja je echo-demo ekstenzija na rmlh ? 60 ?

60 je eho test

siemens ide u njemačku

evo šta test kvalitete non-stop prijavljuje

[Jun 19 13:13:24] NOTICE[27350]: chan_sip.c:15766 sip_poke_noanswer: Peer '22' is now UNREACHABLE!  Last qualify: 165
rmlh-1*CLI>

kao da je loša veza ?!???

recimo evo sada

[Jun 19 13:15:40] NOTICE[27350]: chan_sip.c:12599 handle_response_peerpoke: Peer '22' is now Reachable. (165ms / 2000ms)

uradi sljedeće:

• otvori echo-test ekstenziju
• otvori novu ekstenziju 90 ekstenziju da probam sa ovim tajvanskim telefonom
  • pwd rmlh
  • nat=yes naravno
  • codes=gsm only

evo kada zovem ext.22

    -- Executing [22@rmlh:1] Set("SIP/21-069b18b0", "ext_to_call=") in new stack
    -- Executing [22@rmlh:2] Macro("SIP/21-069b18b0", "Get_route_ext_to||22") in new stack
    -- Executing [s@macro-Get_route_ext_to:1] Set("SIP/21-069b18b0", "ext_to_call=") in new stack
    -- Executing [s@macro-Get_route_ext_to:2] Set("SIP/21-069b18b0", "default_ext=22") in new stack
    -- Executing [s@macro-Get_route_ext_to:3] GotoIf("SIP/21-069b18b0", "1?4:5") in new stack
    -- Goto (macro-Get_route_ext_to,s,4)
    -- Executing [s@macro-Get_route_ext_to:4] Set("SIP/21-069b18b0", "ext_to_call=22") in new stack
    -- Executing [s@macro-Get_route_ext_to:5] NoOp("SIP/21-069b18b0", "Finish if-Get_route_ext_to-18") in new stack
    -- Executing [22@rmlh:3] Dial("SIP/21-069b18b0", "SIP/22| 60| tT") in new stack
    -- Called 22
    -- Got SIP response 486 "Busy Here" back from 89.146.156.214
    -- SIP/22-069d6af0 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'SIP/21-069b18b0' status is 'BUSY'

dobijam da si zauzet ?!???

echo test postoji - ext. 60

a evo otvoram ext.90

mislim da je problem codec-a, pošto nisam siguran kako se podešava gsm codec-a (da je available) na aastri, uzeću tajvanca na 90 ext

podešenje gsm codec-a na aastri 53i

u Global Sip Settings u polje

• customized codecs list

treba upisati, npr:

payload=8;ptime=10;silsupp=on; payload=0;ptime=10;silsupp=off

gdje je:

• payload (Payload is the codec type to be used. This represents the data format carried
  within the RTP packets to the end user at the destination. You can enter payload
  values for G.711 a-law, G.711 u-law, and G.729a.) * 0 for G.711 u-Law * 8 for G.711 a-Law * 18 for G.729a
• ptime (in milliseconds) (Ptime (packetization time) is a measurement of the duration of PCM data within
  each RTP packet sent to the destination, and hence defines how much network
  bandwidth is used for transfer of the RTP stream. You enter the ptime values for
  the customized Codec list in milliseconds. (See table below).) * 5, 10, 15, 20.......90
• silsupp (Silsupp is used to enable or disable silence suppression. Voice Activity Detection
  (VAD) on the IP phones is used to determine whether each individual packet
  contains useful speech data. Enabling silsupp results in decreased network
  bandwidth, by avoiding sending RTP packets for any frame where no voice
  energy was detected by the VAD.) * on * off

što znači na raspolaganju imamo

• G.711 u-Law
• G.711 a-Law
• G.729a

samo ta 3 kodeka

gsm-a izgleda nema

onda za lowbanwidth trebamo testirati g729

podesio sam ext.90

[90]
type=friend
username=90
password=rmlh
host=dynamic
callerid="njemacka3" 
callgroup=2
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=g729  <<<<<<<<<<<<<<<<<<<<<<<<<<<<

inače evo liste dostupnih codec-a

Available values

    * all
    * g723 - G.723.1
    * gsm - GSM
    * ulaw - u-Law
    * alaw - a-Law
    * g726 - G.726-32
    * adpcm - ADPCM
    * slin - SLIN
    * lpc10 - LPC10
    * g729 - G.729
    * speex - SPEEX
    * ilbc - ILBC 

nakon ovoga kada zovem ext.90 kaže:

    -- Executing [90@rmlh:1] Set("SIP/21-069b18b0", "ext_to_call=") in new stack
    -- Executing [90@rmlh:2] Macro("SIP/21-069b18b0", "Get_route_ext_to||90") in new stack
    -- Executing [s@macro-Get_route_ext_to:1] Set("SIP/21-069b18b0", "ext_to_call=") in new stack
    -- Executing [s@macro-Get_route_ext_to:2] Set("SIP/21-069b18b0", "default_ext=90") in new stack
    -- Executing [s@macro-Get_route_ext_to:3] GotoIf("SIP/21-069b18b0", "1?4:5") in new stack
    -- Goto (macro-Get_route_ext_to,s,4)
    -- Executing [s@macro-Get_route_ext_to:4] Set("SIP/21-069b18b0", "ext_to_call=90") in new stack
    -- Executing [s@macro-Get_route_ext_to:5] NoOp("SIP/21-069b18b0", "Finish if-Get_route_ext_to-30") in new stack
    -- Executing [90@rmlh:3] Dial("SIP/21-069b18b0", "SIP/90| 60| tT") in new stack
[Jun 19 14:50:22] WARNING[2172]: chan_sip.c:3028 sip_call: No audio format found to offer. Cancelling call to 90
    -- Couldn't call 90
  == Everyone is busy/congested at this time (0:0/0/0)
  == Auto fallthrough, channel 'SIP/21-069b18b0' status is 'CHANUNAVAIL'

isto tako izbaci i ovo u milion linija

[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:49] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:51] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:51] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:51] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:52] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
[Jun 19 14:51:52] WARNING[2418]: chan_sip.c:3762 sip_write: Can't send 10 type frames with SIP write
  == Spawn extension (rmlh, 60, 2) exited non-zero on 'SIP/90-069c6df0'

kada ext.90 poziva mene

evo sada sip.conf ispravljen

[general]
srvlookup=yes
context=rmlh
;allowoverlap=no        ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
canreinvite=yes
localnet=192.168.77.0/255.255.255.0
externip=89.146.135.1
externrefresh=10
disallow=all
allow=alaw
allow=gsm
allow=g729
language=bs

[Jun 19 15:05:40] NOTICE[27350]: chan_sip.c:5481 process_sdp: No compatible codecs, not accepting this offer!
[Jun 19 15:05:52] NOTICE[27350]: chan_sip.c:5481 process_sdp: No compatible codecs, not accepting this offer!

sada smo testirali sa g722 i izbacuje ovo gore

• g726 ne radi
• g729 ne radi
• g722 - radi ali loš zvuk, odsjeca krajeve riječi

na siemens telefonu audio codec poredak je sljedeći

G729
G726
G711a law
G722u law
G722

telefon sam na kraju podesio, uzeo jednu stanicu siemens i jedan telefon

testirao sam i od kuće, kako će da šljaka konekcija, i to radi super...

Rihad ide u nedjelju, pa ćemo dobaciti u petak telefon, neko od nas

telefon je odnešen

jutros kada smo došli, dobili smo info da nije odnjeo telefon uz poruku: "Nismo to dogovorili !!!!"

šta ovaj čovjek želi ???