[asterisk-users] IAX and Variables
Hi
a new small question ;=)
We have two Asterisk, connected in IAX2.
On the first, in dialplan, we have:
exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)})
we sent into the IAXVAR "ACCOUNTID" the accountcode.
On the second, in dialplan, we have:
exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
That's work, the second server get the variable.
I want now said at the second server that accountcode =
${IAXVAR(ACCOUNTID)},
for use this accoundcode in CDR. On second server, in cdr_mysql.conf i have:
[columns]
alias start => calldate
alias end => callend
alias clid => clid
alias src => src
alias dst => dst
alias dcontext => dcontext
alias channel => channel
alias dstchannel => dstchannel
alias lastapp => lastapp
alias lastdata => lastdata
alias duration => duration
alias billsec => billsec
alias disposition => disposition
alias amaflags => amaflags
alias accountcode => accountcode
alias userfield => userfield
alias uniqueid => uniqueid
But where i can put into the config that for this cdr entry accountcode
= ${IAXVAR(ACCOUNTID)} ?
thanks for your help
jerome
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[asterisk-users] Ring Busy ?
Hi I use asterisk with Realtime/Mysql. I have put a Call-limit at 1, but if the SIP account receive two call in same time, the second call don't ring busy. Ayone know a solution for the second call get a busy ring ? Best Regards Jerome -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Mysql
Hello, I am looking to know if it is possible to modify the SQL query that is on Realtime sip accounts. I want multiple servers use the same sql table, so getting an extra "server" field to indicate that the data is valid on the X server is this possible? thank you in advance jerome -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chage Asterisk 1.6.1 to 1.6.2
Hi A question, i have upgraded a beta serveur from Asterisk 1.6.1 to 1.6.2 and now all SIP Relatime user are rejeted : [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21639 handle_incoming: Received REGISTER (2) - Command in SIP REGISTER [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21658 handle_incoming: Ignoring SIP message because of retransmit (REGISTER Seqno 44199, ours 44199) [Oct 5 05:39:22] DEBUG[15081]: res_config_mysql.c:1602 mysql_reconnect: MySQL RealTime: Connection okay. [Oct 5 05:39:22] DEBUG[15081]: res_config_mysql.c:371 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM VOIP_Comptes_SIP WHERE name = 'SIPUSER001' AND host = 'dynamic' [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:23734 build_peer: -REALTIME- peer built. Name: SIPUSER001. Peer objects: 13 [Oct 5 05:39:22] WARNING[15081]: acl.c:392 ast_get_ip_or_srv: Unable to lookup '' [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:4651 sip_destroy_peer: Destroying SIP peer SIPUSER001 [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:3557 __sip_xmit: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.20.113:5062 [Oct 5 05:39:22] NOTICE[15081]: chan_sip.c:21538 handle_request_register: Registration from '"DEVSIP000105" ' failed for '88.xx.xx.xx' - No matching peer found The same configuration work on 1.6.1 anyone know this error ? 192.168.20.113 it's the local LAN of IP Phone with NAT thanks Jerome -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Realtime and MySQL
Hi thanks for your answer, i put that don't work, but it's a error, that work. But Asterisk crash when i use my second extensions table, i don't know why (limitation of number of line ?) I don't have the answer actually ;=) bye Jerome Le 01/10/2010 11:07, Захаров Антон a écrit : >[ivr_holiday] > switch => Realtime/ivr_holid...@extensions > > where 'ivr_holidays' is context and 'extensions' is table > > On 01.10.2010 12:52, Phibee Network Operation Center wrote: >> Hi >> >> i am not a expert on Asterisk and search a lot of small information : >> >>I use Asterisk 1.6.1.4 with MySQL. >> >> That's work and in my extension.conf, i have: >>[as5300-incoming] >>switch => Realtime >> >> and in extconfig.conf >>extensions => mysql,general,VOIP_Extensions >> A lot of Extension are into the table VOIP_Extensions. >> >> I am search to know if i can add a : >>[beta-incoming] >>switch => Realtime >> >>but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta" >> >> >> Anyone know if it's possible ? (use two table for extension) >> >> Thanks >> Jerome SCHEVINGT >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1 Realtime Extensions => Limited ?
Hi Do you know if the number of extension are limited into Asterisk 1.6.1 whe we use Realtime and a Database MySQL ? Asterisk stop when a context with a big quantity of entry are used. Bye Jerome SCHEVINGT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Realtime and MySQL
After test, that's don't work :=< Le 01/10/2010 11:07, Захаров Антон a écrit : >[ivr_holiday] > switch => Realtime/ivr_holid...@extensions > > where 'ivr_holidays' is context and 'extensions' is table > > On 01.10.2010 12:52, Phibee Network Operation Center wrote: >> Hi >> >> i am not a expert on Asterisk and search a lot of small information : >> >>I use Asterisk 1.6.1.4 with MySQL. >> >> That's work and in my extension.conf, i have: >>[as5300-incoming] >>switch => Realtime >> >> and in extconfig.conf >>extensions => mysql,general,VOIP_Extensions >> A lot of Extension are into the table VOIP_Extensions. >> >> I am search to know if i can add a : >>[beta-incoming] >>switch => Realtime >> >>but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta" >> >> >> Anyone know if it's possible ? (use two table for extension) >> >> Thanks >> Jerome SCHEVINGT >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Realtime and MySQL
Le 01/10/2010 11:10, Steve Howes a écrit : > On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote: >> That's work and in my extension.conf, i have: >> [as5300-incoming] >> switch => Realtime >> >> and in extconfig.conf >> extensions => mysql,general,VOIP_Extensions >> A lot of Extension are into the table VOIP_Extensions. >> >> I am search to know if i can add a : >> [beta-incoming] >> switch => Realtime >> >> but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta" >> >> >> Anyone know if it's possible ? (use two table for extension) > Dont think you can use two tables.. But you're using two contexts there > right? Just have your 'beta' stuff in the same table, but different context. > > S Yes but i want two table, not two context in one table. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Realtime and MySQL
Thanks, it's limited the number of table ? Le 01/10/2010 11:07, Захаров Антон a écrit : >[ivr_holiday] > switch => Realtime/ivr_holid...@extensions > > where 'ivr_holidays' is context and 'extensions' is table > > On 01.10.2010 12:52, Phibee Network Operation Center wrote: >> Hi >> >> i am not a expert on Asterisk and search a lot of small information : >> >>I use Asterisk 1.6.1.4 with MySQL. >> >> That's work and in my extension.conf, i have: >>[as5300-incoming] >>switch => Realtime >> >> and in extconfig.conf >>extensions => mysql,general,VOIP_Extensions >> A lot of Extension are into the table VOIP_Extensions. >> >> I am search to know if i can add a : >>[beta-incoming] >>switch => Realtime >> >>but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta" >> >> >> Anyone know if it's possible ? (use two table for extension) >> >> Thanks >> Jerome SCHEVINGT >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk/Realtime and MySQL
Hi i am not a expert on Asterisk and search a lot of small information : I use Asterisk 1.6.1.4 with MySQL. That's work and in my extension.conf, i have: [as5300-incoming] switch => Realtime and in extconfig.conf extensions => mysql,general,VOIP_Extensions A lot of Extension are into the table VOIP_Extensions. I am search to know if i can add a : [beta-incoming] switch => Realtime but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta" Anyone know if it's possible ? (use two table for extension) Thanks Jerome SCHEVINGT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=<, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.20
[Srv2]
type=peer
host=192.168.0.20
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontext=Incoming
extension.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
[Incoming]
exten => _X.,1,Playback(demo-thanks)
exten => _X.,2,Hangup
[Out]
exten => _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r)
exten => _201X.,2,Congestion
==
Srv1*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
Srv2 192.168.0.20 (S) 255.255.255.255 4569 (E) OK (39 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
On Srv2
iax.conf
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.5
bandwidth=low
[Srv1]
type=peer
host=192.168.0.5
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontect=Incoming
extensions.conf:
[Incoming]
exten => _X.,1,Playback(demo-thanks)
exten => _X.,2,Hangup
[Out]
exten => _202X.,1,Dial(IAX2/Srv1/${EXTEN:3},90,r)
exten => _202X.,2,Congestion
===
trader-voip*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
Srv1 192.168.0.5 (S) 255.255.255.255 4569 (E) OK (28 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
===
All SIP Poste are connected and have in context in: Out
Now, when i call from a post connected on Srv1, i have this error on Srv1:
[Nov 21 08:09:44] WARNING[6407]: chan_iax2.c:9018 socket_process: Call
rejected by 192.168.0.20: No authority found
and on Srv2:
[Nov 21 08:09:44] NOTICE[9089]: chan_iax2.c:9785 socket_process:
Rejected connect attempt from 192.168.0.5, who was trying to reach
'1...@incoming'
125 are the number called (201125)
Dialplan on Srv2
Srv2*CLI> dialplan show Incoming
[ Context 'Incoming' created by 'pbx_config' ]
'_X.' => 1. Playback(demo-thanks)
[pbx_config]
2. Hangup()
[pbx_config]
-= 1 extension (2 priorities) in 1 context. =-
Anyone can help me for know where is my error ?
thanks
Jerome
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Re: [asterisk-users] I don't know how to authenticate
Ok i have change a lot of conf and now when i call, i have:
On server 1:
[Nov 20 20:03:30] WARNING[12047]: chan_iax2.c:9018 socket_process: Call
rejected by 78.SERVER2: No authority found
On Server 2:
[Nov 20 20:03:30] NOTICE[32234]: chan_iax2.c:10229 socket_process: Host
84.SERVER1 failed to authenticate as 04NUMBERCALLED
Phibee Network Operation Center a écrit :
No change, now:
voip*CLI> iax2 show peers
Name/UsernameHost Mask Port
Status
Trader-Classic/ 78.SERVER2 (D) 255.255.255.255 4569 (E) OK
(2 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
voip*CLI>
voip*CLI> iax2 show registry
Host dnsmgr UsernamePerceived
Refresh State
78.SERVER2:4569N VoIP84.SERVER1:4569 60
Registered
1 IAX2 registrations.
voip*CLI>
trader-voip*CLI> iax2 show peers
Name/UsernameHost Mask Port
Status
VoIP/VoIP84.SERVER1 (D) 255.255.255.255 4569 (E) OK
(16 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
trader-voip*CLI> iax2 show registry
Host dnsmgr UsernamePerceived
Refresh State
84.SERVER1:4569N Trader-Cla 78.SERVER2:4569 60
Registered
1 IAX2 registrations.
trader-voip*CLI>
all registration and peers are Ok, the dial on srv1:
exten => _X.,1,Set(CDR(CodeTier)=CLA-UNKNOW)
exten => _X.,2,Dial(IAX2/${ext...@trader-classic,180,rt)
exten => _X.,3,Hangup
but no change when i want call:
Connected to Asterisk 1.6.1.4 currently running on voip (pid = 12026)
[Nov 20 19:59:16] WARNING[12046]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate VoIP to 78.SERVER2
voip*CLI>
iax.conf:
Server1:
register => VoIP:x...@78.server2:4569
[Trader-Classic]
type=friend
host=dynamic
defaultip=78.SERVER2
username=Trader-Classic
auth=md5
port=4569
qualify=yes
secret=
trunk=no
notransfer=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
Server 2:
register => Trader-Classic:x...@84.server1:4569
[VoIP]
type=friend
host=dynamic
defaultip=84.SERVER1
username=VoIP
auth=md5
secret=XXX
qualify=yes
trunk=no
notransfer=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
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Re: [asterisk-users] I don't know how to authenticate
No change, now:
voip*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
Trader-Classic/ 78.SERVER2 (D) 255.255.255.255 4569 (E) OK (2 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
voip*CLI>
voip*CLI> iax2 show registry
Host dnsmgr UsernamePerceived Refresh
State
78.SERVER2:4569N VoIP84.SERVER1:4569 60
Registered
1 IAX2 registrations.
voip*CLI>
trader-voip*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
VoIP/VoIP84.SERVER1 (D) 255.255.255.255 4569 (E) OK (16 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
trader-voip*CLI> iax2 show registry
Host dnsmgr UsernamePerceived Refresh
State
84.SERVER1:4569N Trader-Cla 78.SERVER2:4569 60
Registered
1 IAX2 registrations.
trader-voip*CLI>
all registration and peers are Ok, the dial on srv1:
exten => _X.,1,Set(CDR(CodeTier)=CLA-UNKNOW)
exten => _X.,2,Dial(IAX2/${ext...@trader-classic,180,rt)
exten => _X.,3,Hangup
but no change when i want call:
Connected to Asterisk 1.6.1.4 currently running on voip (pid = 12026)
[Nov 20 19:59:16] WARNING[12046]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate VoIP to 78.SERVER2
voip*CLI>
iax.conf:
Server1:
register => VoIP:x...@78.server2:4569
[Trader-Classic]
type=friend
host=dynamic
defaultip=78.SERVER2
username=Trader-Classic
auth=md5
port=4569
qualify=yes
secret=
trunk=no
notransfer=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
Server 2:
register => Trader-Classic:x...@84.server1:4569
[VoIP]
type=friend
host=dynamic
defaultip=84.SERVER1
username=VoIP
auth=md5
secret=XXX
qualify=yes
trunk=no
notransfer=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
Phibee Network Operation Center a écrit :
anyone know this error message ?
Phibee Network Operation Center a écrit :
Hi
anyone know what is a the solution of this problems ? :
[Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate Voip-Classic to
[Nov 20 13:04:45] WARNING[12043]: chan_iax2.c:9232 socket_process: I
don't know how to authenticate Voip-Classic to 78.XX.XX.XX
we have two Asterisk 1.6.1.4, this error are on the first server, used
for the connection in SIP of
final user (personnal phone).
78.XX are the IP of our second server used for call routing.
We have tested in SIP and in IAX2 without change
thanks for your suggestion
jerome
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Re: [asterisk-users] I don't know how to authenticate
anyone know this error message ? Phibee Network Operation Center a écrit : > Hi > > anyone know what is a the solution of this problems ? : > > [Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I > don't know how to authenticate Voip-Classic to > [Nov 20 13:04:45] WARNING[12043]: chan_iax2.c:9232 socket_process: I > don't know how to authenticate Voip-Classic to 78.XX.XX.XX > > we have two Asterisk 1.6.1.4, this error are on the first server, used > for the connection in SIP of > final user (personnal phone). > > 78.XX are the IP of our second server used for call routing. > > We have tested in SIP and in IAX2 without change > > thanks for your suggestion > jerome > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I don't know how to authenticate
Hi anyone know what is a the solution of this problems ? : [Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I don't know how to authenticate Voip-Classic to [Nov 20 13:04:45] WARNING[12043]: chan_iax2.c:9232 socket_process: I don't know how to authenticate Voip-Classic to 78.XX.XX.XX we have two Asterisk 1.6.1.4, this error are on the first server, used for the connection in SIP of final user (personnal phone). 78.XX are the IP of our second server used for call routing. We have tested in SIP and in IAX2 without change thanks for your suggestion jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call IAX2 => "Call rejected, CallToken Support required"
Hi i have a small problems on two Asterisk Server 1.6.4 : The first sent the call to the second, and in the second, i have a error : [Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address IP_FIRST_ASTERISK in the calltokenignore list or setting user 04TELNUMBER requirecalltoken=no on the second server, i have into iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes calltokenignore=IP_First8asterisk_Server [Voip] type=friend user=VoIP secret=X host=NAME_DNS_FIRST_SERVER requirecalltoken=no qualify=yes disalow=all allow=alaw allow=g729 context=Incoming Thanks for your help Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error Dialplan ?
Hi
I have a problems with a new Asterisk Server,
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP01' to extension
'00420225352184' rejected because extension not found.
but into my extensions.conf:
exten => _00420X.,1,Set(CDR(CodeTier)=CZE)
exten => _00420X.,2,Dial(SIP/${ext...@as5350,180,rt)
exten => _00420X.,3,Hangup
and a dialplan show:
exten => _00420X.,1,Set(CDR(CodeTier)=CZE)
exten => _00420X.,2,Dial(SIP/${ext...@as5300,180,rt)
exten => _00420X.,3,Hangup
I have a error, it's sure because i am new user, but ?
context of the user are good
thanks
Jerome
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[asterisk-users] Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?
Hi I have finished the installation of my VoIP basic configuration ... Actually: - All calls from my E1 are received by a Cisco AS5300 and sent to my Asterisk (in G711 by SIP). - All user are connected by SIP to the Asterisk - All calls from User are sent by asterisk to the Cisco AS5300 Now, i want see if i can supply T38 Fax Gateway I am search to: - Cisco Receive all Fax, two poss: he detect automatiqueley a Fax, or based on phone number. - Sent the fax in T38 to Asterisk for "Fax Routing" - Based on the number and extensions, Asterisk sent the T38 call to the T38 terminaison (other server) .. Anyone know if it's possible ? and if yes, what is the configuration process (on the cisco and on the Asterisk Gateway) Thanks Jérôme ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime Extensions => for all context ?
Hi I Use Asterisk 1.6.1 with Realtime and a mySQL database, Actually, my extensions.conf are: === [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo [as5300-incoming] switch => Realtime [as5300-outgoing] switch => Realtime [user1] switch => Realtime [user2] switch => Realtime [user3] switch => Realtime === and into my table, i put the context ... anyone know if i can use a generic: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo switch => Realtime and use directly the context of my database ? without put userX into my extensions.conf thanks Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error in MeetMe modules ?
Hi when i use MeetMe, i have this errors: app_meetme.c: Unable to open pseudo device Where is the problems ? i have too warning and error into my logs: [Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. [Nov 1 07:26:17] WARNING[18544] config.c: Realtime mapping for 'iaxpeers' found to engine 'mysql', but the engine is not available [Nov 1 07:26:17] ERROR[18544] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory What is the process for resolv this problems ? "Realtime mapping for 'iaxpeers' found to engine 'mysql', but the engine is not available", i don't understand because realtime Sip+Extension works and cdr too Thanks for your help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?
Hi ok i have understand ;=) bye Phibee Network Operation Center a écrit : > Hi > > actually, i test a new Asterisk Server and i want add Mysql Realtime SIP. > > I read on the wiki: > > === > Database Config > put the following in res_mysql.conf > > [general] > dbhost = 127.0.0.1 > dbname = asterisk > dbuser = myuser > dbpass = mypass > dbport = 3306 > > Values in sip.conf or iax.conf like in older versions of * are no longer > used. > > > Database Table > Lets create the table we need: > > NOTE: You can use any table name you wish, just make sure the table name > matches what you have the family name bound to. > > === > > > But i don't see where i put the Table Name ? (if i don't want use > sip_buddies) > > and he have a sample of Table Structure, can i add a new champs for my > personnal > software without problems ? > > Thanks > jerome > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?
Hi actually, i test a new Asterisk Server and i want add Mysql Realtime SIP. I read on the wiki: === Database Config put the following in res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = myuser dbpass = mypass dbport = 3306 Values in sip.conf or iax.conf like in older versions of * are no longer used. Database Table Lets create the table we need: NOTE: You can use any table name you wish, just make sure the table name matches what you have the family name bound to. === But i don't see where i put the Table Name ? (if i don't want use sip_buddies) and he have a sample of Table Structure, can i add a new champs for my personnal software without problems ? Thanks jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)
Phibee Network Operation Center a écrit : > Hi > > Now, my Cisco AS5300 sent call to my asterisk, but two problems: > > When i call the phone number, i have: > > [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: > Call from '' to extension '042600' rejected because extension not found. > [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: > Call from '' to extension '042600' rejected because extension not found. > > (042600 = my phone number) > <..> > I have put a debug: [Kvoip*CLI> <--- SIP read from UDP://192.168.50.125:59124 ---> INVITE sip:0426000...@192.168.50.130:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.125:5060 From: ;tag=6950F0-25C7 To: Date: Wed, 28 Oct 2009 05:16:26 GMT Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3761097657-3266777566-2192416711-2957366127 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: ;party=calling;screen=yes;privacy=off Timestamp: 1256706986 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 8642 2741 IN IP4 192.168.50.125 s=SIP Call c=IN IP4 192.168.50.125 t=0 0 m=audio 18726 RTP/AVP 8 101 c=IN IP4 192.168.50.125 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-> [Kvoip*CLI> --- (20 headers 11 lines) --- [Kvoip*CLI> Sending to 192.168.50.125 : 5060 (no NAT) [Kvoip*CLI> Using INVITE request as basis request - e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 [Kvoip*CLI> No matching peer for '47700' from '192.168.50.125:59124' [Kvoip*CLI> Found RTP audio format 8 [Kvoip*CLI> Found RTP audio format 101 [Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI> Found audio description format PCMA for ID 8 [Kvoip*CLI> Found audio description format telephone-event for ID 101 [Kvoip*CLI> Got unsupported a:fmtp in SDP offer [Kvoip*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Kvoip*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI> Looking for 042600 in default (domain 192.168.50.130) [Kvoip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.50.125:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.50.125:5060;received=192.168.50.125 From: ;tag=6950F0-25C7 To: ;tag=as25696e60 Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Ok, i see that: 1- Cisco sent the phone number of the caller (47700) 2- I have a "To: " 192.168.50.130 = My Asterisk Server 192.168.50.125 = My Cisco AS5300 3- i have a "No matching peer for '47700' from '192.168.50.125:59124'" why he search a peer with "47700" ?? bye Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)
Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. (042600 = my phone number) First problems: Why he don't see the extension ? sip.conf: [AS5300] host=192.168.50.125 context=as5300-incoming type=peer dtmf=rfc2833 nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp [as5300-incoming] exten => 042600,1,Ringing exten => 042600,2,Answer exten => 042600,3,Dial(SIP/Jpc,25,m) exten => 042600,4,Hangup And second problems: "Call from '' to", AS5300 don't sent the number of the caller ? Thanks for your help Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with a Cisco AS5300 gateway
Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 pri-group timeslots 1-31 description E1 Beta-Test interface Serial0 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial1 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial2 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial3 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial0:15 no ip address encapsulation ppp isdn switch-type primary-net5 no cdp enable voice-port 0:D ! ! ! dial-peer voice 10 voip destination-pattern .T session protocol sipv2 session target ipv4:IP_OF_ASTERISK:5060 session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 42 pots destination-pattern .T direct-inward-dial port 0:D ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:IP_OF_ASTERISK ! Actually, a Tcpdump on my Asterisk server don't see any trafic between asterisk and cisco and when i call a phone number that arrives on the E1, it's "busy" anyone have a idea ? bye jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] International Numbering plan ?
Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Cisco 5300/E1/DSP
Hi I search to know if a company or user use that a Cisco AS5300 with DSP/Voice with Asterisk ? I want use the AS5300 only for the E1/PSTN link in/out thanks Jpc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?
Hi thanks for your answer, not curious, i have one 1760 with FXO card and ~100 Cisco 1751 with FXO Card to at connected to my asterisk in SIP but i don't have touch Asterisk since 18 mounth ... and never connected router at asterisk (only Linksys SPA941 voice unit) if you have a idea of the configuration of the Cisco 1760 .. it's help me bye jerome David Gibbons a écrit : > > Anyone use CIsco 1760 with Asterisk > > > No, but I'm using 7941G-GE and 7961G-GE in a deployment of ~80 phones. Did > you have a question about implementation or are you just curious? > > --Dave > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?
Anyone use CIsco 1760 with Asterisk Phibee Network Operation Center a écrit : > Hi > > i am search a sample config (for asterisk and for cisco) for connect > a cisco 1760 with a FXO card to my asterisk. > > Thanks for your help > Jerome > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and CIsco 1760 SIP ?
Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream
I have a problem connecting a Grandstream ipphone to an asterisk. The ipphone is behind a nat router, I redirected UDP 5060 and 5004 to my phone. It connects well to the asterisk server. I can call outside and receive calls from outside without any problems. But if I call from this ipphone to another ipphone connected on the same asterisk server, using internal dialing, I can hear my correspondant, but he cannot. Do you have any idea? Thanks for advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
