Hello,
I’m new there and I don’t have a lot of knowledge on Asterisk and the PJSIP module. So there are my problems actually. I recently move from a VPS to my home servers the Asterisk PBX and during the re-installation I forgot to add CHAN_SIP module so I get the opportunity to start migrate my configuration from chan_sip to pjsip then I wouldn’t have known it would be such harder as it is. I have 5 Cisco 7940 and 7960 IP phones in my home all configured to the SIP firmware and everything was working fine before I started to move.
Then from now, I can : receive calls from the SIP trunk I configured, call internal but only to the number registered using MicroSIP from my computer on the same network of all of my Cisco phones. When I try to make a call using the MicroSIP (103) client to my Cisco phone (100) the request timeout. I can hear my voice when I do that too, can of course call the standard 8002. When I try to call outside I stuck on “Proceeding in 100” on the Cisco Phones and Request timeout on MicroSIP client. I’m literally lost for the moment and don’t know what I can do furthermore.
Here is my pjsip.conf file.
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.20.0/24
external_media_address=217.x.x.x.
external_signaling_address=217.x.x.x.
[homesip_reg]
type=registration
transport=transport-udp
outbound_auth=homesip_auth
server_uri=sip:sip5.ovh.fr:5060
client_uri=sip:00331@sip5.ovh.fr:5060
retry_interval=60
max_retries=10
expiration=3600
line=yes
endpoint=homesip
[homesip_auth]
type=auth
auth_type=userpass
password=
username=00331
realm=sip5.ovh.fr
[homesip]
type=aor
contact=sip:sip5.ovh.fr:5060
qualify_frequency=5
remove_existing=yes
[homesip]
disallow=all
type=endpoint
transport=transport-udp
context=homesip-in
from_user=00331
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
allow = alaw,ulaw,gsm
direct_media=no
aors=homesip
ice_support=yes
outbound_auth=homesip_auth
[homesip]
type=identify
endpoint=homesip
match=sip5.ovh.fr
;Users
;100-louis
[100]
type=endpoint
transport=transport-udp
context=home
disallow=all
allow=gsm,ulaw,alaw
force_rport=no
aors=100
auth=100-auth
[100-auth]
type=auth
auth_type=userpass
password=
username=100
realm=asterisk
[100]
type=aor
max_contacts=1
qualify_frequency=5
remove_existing=yes
;101-gabriel
[101]
type=endpoint
transport=transport-udp
context=home
disallow=all
allow=gsm,ulaw,alaw
force_rport=no
aors=101
auth=101-auth
[101-auth]
type=auth
auth_type=userpass
password=
username=101
realm=asterisk
[101]
type=aor
max_contacts=1
qualify_frequency=5
remove_existing=yes
;Cuisine
[102]
type=endpoint
transport=transport-udp
context=home
disallow=all
allow=gsm,ulaw,alaw
force_rport=no
aors=102
auth=102-auth
[102-auth]
type=auth
auth_type=userpass
password=
username=102
realm=asterisk
[102]
type=aor
max_contacts=1
qualify_frequency=5
remove_existing=yes
;103-ChMaitre
[103]
type=endpoint
transport=transport-udp
context=home
disallow=all
allow=gsm,ulaw,alaw
force_rport=no
aors=103
auth=103-auth
[103-auth]
type=auth
auth_type=userpass
password=
username=103
realm=asterisk
[103]
type=aor
max_contacts=3
qualify_frequency=5
remove_existing=yes
So, I removed passwords, SIP trunk user, then ip adresses.
Here is my extensions.
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=DAHDI/G2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[guest]
;Ce groupe ne peut pas recevoir d'appels
;Appels internes
exten => _1XX,1,Dial(PJSIP/${EXTEN},35)
;Appels d'urgence
exten => _18,1,Dial(PJSIP/homesip/${EXTEN})
exten => _15,1,Dial(PJSIP/homesip/${EXTEN})
exten => _17,1,Dial(PJSIP/homesip/${EXTEN})
exten => _112,1,Dial(PJSIP/homesip/${EXTEN})
[home]
;Ce groupe peut recevoir des appels
;Serveur vocal params
exten => 8002,1,Goto(ivr-home,s,1)
;Appels internes
exten => _1XX,1,Dial(PJSIP/${EXTEN},35)
;Appels externes
exten => _0[1-9]XXXXXXXX,1,Set(CALLFILENAME=${EXTEN}${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => _0[1-9]XXXXXXXX,2,MixMonitor(/var/spool/asterisk/recording/${CALLFILENAME}.wav,bW(3))
exten => _0[1-9]XXXXXXXX,3,Dial(PJSIP/${EXTEN}@homesip)
exten => _[1-9]XXX,1,Set(CALLFILENAME=${EXTEN}${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => _[1-9]XXX,2,MixMonitor(/var/spool/asterisk/recording/${CALLFILENAME}.wav,bW(3))
exten => _[1-9]XXX,3,Dial(PJSIP/${EXTEN}@homesip)
;Recording IVRs
exten => _5XXX,1,Record(IVR-${EXTEN:1}:ulaw)
exten => _5XXX,2,Playback(IVR-${EXTEN:1})
;exten => 900,1,ConfBridge(Room_1,ConfRoom_1,User_NoAuth)
;exten => 901,1,ConfBridge(Room_1,ConfRoom_1,User_Admin,ConfMenu)
;exten => 903,1,ConfBridge(Room_2,ConfRoom_2,User_Auth)
;exten => 904,1,ConfBridge(Room_2,ConfRoom_2,User_Admin)
;Appels d'urgence
exten => _18,1,Dial(PJSIP/homesip/${EXTEN})
exten => _15,1,Dial(PJSIP/homesip/${EXTEN})
exten => _17,1,Dial(PJSIP/homesip/${EXTEN})
exten => _112,1,Dial(PJSIP/homesip/${EXTEN})
[homesip-in]
exten => s,1,GotoIf($["${CALLERID(num)}" = "04"]?dial1)
exten => s,n,GotoIf($["${CALLERID(num)}" != "04"]?10)
exten => s,n(dial1),Dial(PJSIP/100,25)
exten => s,n,Goto(ivr-busy-louis,s,1)
exten => s,10,Goto(home,8002,1)
exten => s,n,Hangup()
[ivr-home]
exten => s,1,Answer()
exten => s,2,Set(TIMEOUT(response)=10)
exten => s,3,Background(IVR-005)
exten => s,4,WaitExten()
exten => 1,1,Playback(waiting2)
exten => 1,2,Dial(PJSIP/100&PJSIP/101&PJSIP/102&PJSIP/103,30)
exten => 1,3,Background(IVR-007)
exten => 1,4,Hangup
exten => 2,1,Goto(ivr-home-choose,s,1)
exten => 9,1,Playback(IVR-011)
exten => 9,2,Goto(home,8002,1)
[ivr-home-choose]
exten => s,1,Answer()
exten => s,2,Set(CALLFILENAME=${CALLERID(num)}${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => s,3,MixMonitor(/var/spool/asterisk/recording/${CALLFILENAME}.wav,bW(3))
exten => s,4,Background(IVR-006)
exten => s,5,Set(TIMEOUT(response)=10)
exten => s,6,WaitExten()
exten => 1,1,Answer()
exten => 1,2,Playback(waiting2)
exten => 1,3,Dial(PJSIP/100,25)
exten => 1,4,Goto(ivr-busy-louis,s,1)
exten => 2,1,Answer
exten => 2,2,Playback(waiting2)
exten => 2,3,Dial(PJSIP/101,25)
exten => 2,4,Goto(ivr-busy-gab,s,1)
exten => 3,1,Answer
exten => 3,2,Playback(waiting2)
exten => 3,3,Dial(PJSIP/102,25)
exten => 3,4,Background(IVR-008)
exten => 3,5,Hangup
exten => 4,1,Answer
exten => 4,2,Playback(waiting2)
exten => 4,3,Dial(PJSIP/103,25)
exten => 4,4,Background(IVR-008)
exten => 4,5,Hangup
exten => 5,1,Answer
exten => 5,2,Playback(waiting2)
exten => 5,3,Dial(PJSIP/104,25)
exten => 5,4,Background(IVR-008)
exten => 5,5,Hangup
[ivr-busy-louis]
exten => s,1,Background(IVR-009)
exten => s,2,Set(TIMEOUT(response)=10)
exten => s,3,WaitExten()
exten => 1,1,Dial(PJSIP/xxxxx@homesip)
[ivr-busy-gab]
exten => s,1,Background(IVR-009)
exten => s,2,Set(TIMEOUT(response)=10)
exten => s,3,WaitExten()
exten => 1,1,Dial(PJSIP/xxxxx@homesip)
Do you have any recommendations?
Sincerly.
Louis
