Re: [asterisk-users] Peer doesn't answer
Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- _ -- 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 -- Arlen Nascimento -- _ -- 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 -- _ -- 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] ssh to a Cisco 7961 is not working
Ken, Does your phone is realy able to accept ssh connection? I mean , it is set up for it ? As we can see in the log, it is sending reset to the ssh client. 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 look like it is not accepting ssh connections. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 01:06:34 -0800 From: k...@impulse.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] ssh to a Cisco 7961 is not working I am trying to ssh to my Cisco 7961 VoIP phone (computer and phone on the same LAN and switch) but I always get a connection refused. I have tried from my desktop and a laptop running different OS's. I have tried ssh 10.0.0.155 and ssh cisco@10.0.0.155 from a command prompt. Here are the results from sniffing via Wireshark: 11038 2272.240571 10.0.0.172 10.0.0.155 TCP 78 57665 ssh [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=8 TSval=963558895 TSecr=0 SACK_PERM=1 11039 2272.240681 10.0.0.172 10.0.0.155 TCP 78 57665 ssh [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=8 TSval=963558895 TSecr=0 SACK_PERM=1 11046 2272.241550 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 Ack=1 Win=8192 Len=0 11047 2272.241554 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 Ack=1 Win=8192 Len=0 I don't know why everything is duplicated, but I'm not very proficient at Wireshark. Here is a snippet from my SEP*cnf.xml file: sshUserIdcisco/sshUserId sshPasswordcisco/sshPassword Can anyone offer any help/suggestions? Thank you! Ken Alker Impulse Internet Services http://www.impulse.net -- _ -- 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 -- _ -- 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] ssh to a Cisco 7961 is not working
Ken, According with cisco docs, ssh is disable by default: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/firmware/9_2_1/english/release/notes/7900_921.html SSH Access The SSH Access settings option allows the administrator to enable or disable the SSH port on the phone using Cisco Unified CM Administration. When enabled, it allows the phone to accept the SSH connections. Disabling the SSH server functionality of the phone blocks the SSH access to the phone. This setting is disabled by default. This feature is supported on the following Cisco Unified IP Phones (SCCP and SIP): •Cisco Unified IP Phone 7906G •Cisco Unified IP Phone 7911G •Cisco Unified IP Phone 7931G •Cisco Unified IP Phone 7941G •Cisco Unified IP Phone 7941G-GE •Cisco Unified IP Phone 7942G •Cisco Unified IP Phone 7945G •Cisco Unified IP Phone 7961G •Cisco Unified IP Phone 7961G-GE •Cisco Unified IP Phone 7962G •Cisco Unified IP Phone 7965G •Cisco Unified IP Phone 7970G •Cisco Unified IP Phone 7971G •Cisco Unified IP Phone 7975G Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 13:32:36 -0800 From: k...@impulse.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ssh to a Cisco 7961 is not working Flavio, Thank you for your response. According to various wiki's (voip-info.org included), the 7961 is supposed to accept SSH connections (and in fact, many people recommend this for debugging, but what I often see is just connect via SSH as if it should simply work; I haven't run across any data indicating people have had problems connecting via ssh as I am). I must assume that either the wiki's are wrong (doubtful, but possible), or Cisco deactivated ssh in this firmware build, or I need to alter a setting in my SEP*.cnf.xml file or on the phone itself; but I don't know what that would be. As per below, I've defined an ssh userid and password via the xml file. --On January 15, 2012 10:20:06 AM -0200 Flavio Miranda flaviormira...@hotmail.com wrote: Ken, Does your phone is realy able to accept ssh connection? I mean , it is set up for it ? As we can see in the log, it is sending reset to the ssh client. 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 look like it is not accepting ssh connections. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 01:06:34 -0800 From: k...@impulse.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] ssh to a Cisco 7961 is not working I am trying to ssh to my Cisco 7961 VoIP phone (computer and phone on the same LAN and switch) but I always get a connection refused. I have tried from my desktop and a laptop running different OS's. I have tried ssh 10.0.0.155 and ssh cisco@10.0.0.155 from a command prompt. Here are the results from sniffing via Wireshark: 11038 2272.240571 10.0.0.172 10.0.0.155 TCP 78 57665 ssh [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=8 TSval=963558895 TSecr=0 SACK_PERM=1 11039 2272.240681 10.0.0.172 10.0.0.155 TCP 78 57665 ssh [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=8 TSval=963558895 TSecr=0 SACK_PERM=1 11046 2272.241550 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 Ack=1 Win=8192 Len=0 11047 2272.241554 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 Ack=1 Win=8192 Len=0 I don't know why everything is duplicated, but I'm not very proficient at Wireshark. Here is a snippet from my SEP*cnf.xml file: sshUserIdcisco/sshUserId sshPasswordcisco/sshPassword Can anyone offer any help/suggestions? Thank you! Ken Alker Impulse Internet Services http://www.impulse.net -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Peer doesn't answer
Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- _ -- 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] NAT yes
Hello averybody, In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] NAT yes
Thanks Alex Balashov, I am experiencing some one-way audio, that's the reason of the questions! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Tue, 26 Jul 2011 09:23:42 -0400 From: abalas...@evaristesys.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] NAT yes On 07/26/2011 09:19 AM, Flavio Miranda wrote: In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? There is no harm unless the endpoint you are dealing with does not do symmetric RTP. The nat=yes option assumes that it is okay to send RTP back to the source port from which it originated, irrespectively of what's in the SDP. This will cause one-way audio if the endpoint happens to want to receive RTP on a different port than the one it is sending it from. Almost all endpoints these days do symmetric RTP, though, so it's not a huge concern. That said, from a methodological and aesthetic perspective, it is better not to break standard RFC-compliant behaviour unnecessarily. Thus, I would not enable nat=yes unless there really is no direct network and transport-layer reachability to the endpoint. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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 -- _ -- 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] Scheduling destruction of SIP dialog
Hello, I am receiving the following message all the time, all sip peers, and always finishing with destructing dialog... : --- (13 headers 0 lines) --- Sending to 192.168.0.106 : 5060 (no NAT) Reliably Transmitting (no NAT) to 192.168.0.106:5060: OPTIONS sip:2036@192.168.0.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK58b8c6b7;rport Max-Forwards: 70 From: asterisk sip:asterisk@192.168.0.254;tag=as34ab67bd To: sip:2036@192.168.0.106:5060 Contact: sip:asterisk@192.168.0.254 Call-ID: 21adef7521218c116309d7784527451c@192.168.0.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.18 Date: Tue, 26 Jul 2011 18:09:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- Transmitting (no NAT) to 192.168.0.106:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.106:5060;branch=z9hG4bK1228024af6;received=192.168.0.106;rport=5060 From: Central2 sip:2036@192.168.0.254;tag=40e337db To: Central2 sip:2036@192.168.0.254;tag=as11725d36 Call-ID: 393c15291791541a4628830c0db3acd0@192.168.0.106 CSeq: 802 REGISTER Server: Asterisk PBX 1.6.2.18 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: sip:2036@192.168.0.106:5060;expires=60 Date: Tue, 26 Jul 2011 18:09:32 GMT Content-Length: 0 Scheduling destruction of SIP dialog '393c15291791541a4628830c0db3acd0@192.168.0.106' in 32000 ms (Method: REGISTER) --- SIP read from UDP:192.168.0.106:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.254:5060;rport=5060;received=192.168.0.254;branch=z9hG4bK58b8c6b7 From: asterisk sip:asterisk@192.168.0.254;tag=as34ab67bd To: sip:2036@192.168.0.106:5060;tag=0c6ccbbd Call-ID: 21adef7521218c116309d7784527451c@192.168.0.254 Contact: sip:2036@192.168.0.106:5060 CSeq: 102 OPTIONS Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS Content-Length: 0 Nay body know what's wrong here ? Thanks! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Security questions
Hello everybody!
I'd like to heard from those with more experience in Security if the
following configuration is a good attempt to prevent hack:
exten = CALLER,2,Set(header=${SIP_HEADER(User-Agent)})
exten = CALLER,3,NoOp(Cabecalho ${header})
exten = CALLER,4,GotoIf($[${header}= My User Agent]?6:7)
Considering I have only one type of IP phone in my scenario.
I know, somebody with another IP phone will succeed in dial on my asterisk but
I think it will limit at one only kind of IP phone.
My question is , if there are some way to break it and use any kind of User
Agent despite this configuratio.
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda --
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[asterisk-users] Add # at the end of dialled number
Hi all,
Anybody know if is it possible to add # at the end of dialled number ?
kinda : exten = _00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4},25,T)
In this line I am switching the C.O code but , how could I put # automatic at
the end ?
Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda --
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Re: [asterisk-users] Add # at the end of dialled number
works !
Thanks!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 28 Jun 2011 13:31:41 -0500
Subject: Re: [asterisk-users] Add # at the end of dialled number
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Tuesday, June 28, 2011 1:22 PM
To: Asterisk Asterisk
Subject: [asterisk-users] Add # at the end of dialled number
Hi all, Anybody know if is it possible to add # at the end of dialled number
? kinda : exten = _00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4},25,T) In
this line I am switching the C.O code but , how could I put # automatic at the
end ? Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda Have you tried this?kinda : exten =
_00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4}#,25,T)
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Re: [asterisk-users] Conference feature
Very simple.. Just edit the meetme.conf in /etc/asterisk like this :[rooms] conf = 888 And then, in /etc/asterisk/ extensions.conf , put something like that: [conference] exten = 888,1,Set(CHANNEL(language)=pt_BR)if you have pt_BR audioexten = 888,n,MeetMe(888,pdM)exten = 888,n,Playback(vm-goodbye)exten = 888,n,Hangup When an user call 888 he will be in a conference room. I hope it help! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 26 Jun 2011 22:25:00 -0300 From: rafaels...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Conference feature Hi How to create the conference feature in Asterisk? Thank'sAtt,Rafael Saraiva -- _ -- 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 -- _ -- 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] PADLOCK asterisk 1.6
Hi all,
There is a feature very common in PBX called PADLOCK , and I'd like to set
up it on Asterisk 1.6. I have seen it in the internet but such scripts never
work to me. I am trying to do something like that:Create a password and
associate it with the callerid:
exten = _*11*,1,Set(DB(CADASTRA/${CALLERID(num)})=${EXTEN:4})
Create a flag in order to verify later if PADLOCK is ON or OFF
exten = _*14*,1,set(DB(${CALLERID(num)}/${EXTEN:4})=1)
Now the problem:
When dialing out, I need to verify if such {CALLERID(num) have its value ON of
OFF in order to permit or denny, and I can´t perform this.
??exten = _X.,1,Dial(SIP/${EXTEN}@10.201.201.254)exten = _X.,n,Hangup()
Below, my AstDB:
/3003/1234: 1
/CADASTRA/3003 : 1234
Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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[asterisk-users] Asterisk FOP
Hi ,
I am truing to set up FOP but I getting the following log:
Use of uninitialized value $hash_temporal{SrcUniqueID} in hash element at
./op_server.pl line 3367
What can I change in order to get something like initialized value.
Thanks!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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Re: [asterisk-users] Asterisk FOP
Hi, FOP 1 OS Debian Lenny Asterisk 1.6 Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sat, 9 Apr 2011 14:11:39 -0400 From: supp...@drdos.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk FOP Flavio Miranda wrote: I am truing to set up FOP but I getting the following log: What version of FOP? 1 or 2, what OS? What version of Asterisk? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 -- _ -- 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 FOP
So... something has changed now.When I run ./op_server.pl , I get the following verbose: all my asterisk configuration therefore, my buttons dont work as it should...I am wondering if my extentions.conf must have something different, like a hint function,or something else in order to the FOP show the extensions status, Thanks for any help!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sat, 9 Apr 2011 14:11:39 -0400 From: supp...@drdos.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk FOP Flavio Miranda wrote: I am truing to set up FOP but I getting the following log: What version of FOP? 1 or 2, what OS? What version of Asterisk? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 ** SIP/9001 in next position 2 ** SIP/2010 in next position 3 ** IAX2/9003 in next position 4 ** IAX2/9004 in next position 5 ** PARK/701 in next position 13 ** PARK/702 in next position 14 ** 2099 in next position 9 ** 902 in next position 10 ** 0^_DGV/.*=1 in position 23 ** 0^_DGV/.*=2 in position 24 ** 0^_DGV/.*=3 in position 25 ** 0^_DGV/.*=4 in position 26 ** 0^_DGV/.*=5 in position 27 ** Ignored button DAHDI/4, position? ** 0^QUEUE/SUPORTE=1 in position 16 ** 0^QUEUE/SUPORTE=2 in position 17 ** 0^QUEUE/SUPORTE=3 in position 18 ** 0^QUEUE/SUPORTE=4 in position 19 ** 0^QUEUE/SUPORTE=5 in position 20 ** 0^QUEUE/SUPORTE=6 in position 21 ** MANAGER CONNECTION Connecting to 127.0.0.1:5038 (Server 0) ** MANAGER CONNECTION Connected to 127.0.0.1:5038 (Server 0) 127.0.0.1 - Action: Challenge 127.0.0.1 - AuthType: MD5 127.0.0.1 - Asterisk Call Manager/1.1 127.0.0.1 - Response: Success 127.0.0.1 - Challenge: 162195937 127.0.0.1 - Server: 0 127.0.0.1 - Action: Login 127.0.0.1 - Username: myuser 127.0.0.1 - AuthType: MD5 127.0.0.1 - Key: 95b3af5164aea0e7ac5db9f8a5364eee 127.0.0.1 - Response: Success 127.0.0.1 - Message: Authentication accepted 127.0.0.1 - Server: 0 127.0.0.1 - Action: Events 127.0.0.1 - EventMask: call 127.0.0.1 - Action: Agents 127.0.0.1 - Action: QueueStatus 127.0.0.1 - Member: 127.0.0.1 - Action: Status 127.0.0.1 - Action: ZapShowChannels 127.0.0.1 - Action: SIPPeers 127.0.0.1 - ActionID: autosip 127.0.0.1 - Action: Command 127.0.0.1 - ActionID: parkedcalls 127.0.0.1 - Command: show parkedcalls 127.0.0.1 - Action: MailboxStatus 127.0.0.1 - Mailbox: 2010@default 127.0.0.1 - Action: MailboxStatus 127.0.0.1 - Mailbox: 9003@default 127.0.0.1 - Action: MailboxStatus 127.0.0.1 - Mailbox: 9004@default 127.0.0.1 - Action: MailboxStatus 127.0.0.1 - Mailbox: 9001@default 127.0.0.1 - Action: Command 127.0.0.1 - ActionID: meetme_902 127.0.0.1 - Command: meetme list 902 127.0.0.1 - Action: Command 127.0.0.1 - ActionID: meetme_2099 127.0.0.1 - Command: meetme list 2099 127.0.0.1 - Response: Success 127.0.0.1 - Events: On 127.0.0.1 - Server: 0 Response: Success Events: On Server: 0 127.0.0.1 - Response: Success 127.0.0.1 - Message: Agents will follow 127.0.0.1 - Server: 0 Response: Success Message: Agents will follow Server: 0 127.0.0.1 - Event: AgentsComplete 127.0.0.1 - Server: 0 127.0.0.1 - Response: Success 127.0.0.1 - Message: Queue status will follow 127.0.0.1 - Server: 0 Response: Success Message: Queue status will follow Server: 0 127.0.0.1 - Event: QueueParams 127.0.0.1 - Queue: vendas 127.0.0.1 - Max: 0 127.0.0.1 - Strategy: ringall 127.0.0.1 - Calls: 0 127.0.0.1 - Holdtime: 0 127.0.0.1 - TalkTime: 0 127.0.0.1 - Completed: 0 127.0.0.1 - Abandoned: 0 127.0.0.1 - ServiceLevel: 0 127.0.0.1 - ServicelevelPerf: 0.0 127.0.0.1 - Weight: 0 127.0.0.1 - Server: 0 fake- Server: 0 fake- Holdtime: 0 fake- Completed: 0 fake- Event: QueueStatus fake- ServicelevelPerf: 0.0 fake- Queue: vendas fake- Calls: 0 fake- Abandoned: 0 fake- ServiceLevel: 0 fake- Weight: 0 fake- Strategy: ringall fake- Max: 0 fake- TalkTime: 0 127.0.0.1
[asterisk-users] Number Conversion
Hi all,
Please, could somebody point me out what is going wrong in this line below?
exten = _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT)
As I know, such line must convert any number dialed to 021, therefore, as we
can see, it's kept the number dialed!
-- Executing [00151236445600@a2billing:1] Dial(SIP/2000-,
DAHDI/G0/0151236445600,45,rT}) in new stack-- Called G0/0151236445600
Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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[asterisk-users] Number Conversion
I did
That's weird, doesn't it!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Tue, 5 Apr 2011 13:02:06 +1200
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Number Conversion
On 5/04/11 1:00 PM, Flavio Miranda wrote:
Hi all,
Please, could somebody point me out what is going wrong in this line below?
exten = _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT)
As I know, such line must convert any number dialed to 021, therefore,
as we can see, it's kept the number dialed!
It must not be running that line - have you done a dialplan reload?
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[asterisk-users] AstMail
Hello everybody, Anybody here knows about Astmail ? I have set up in a server but something is going wrong! i can open its web interface but when I put the extension number and its password I receive:Invalid mailbox or password My asterisk is 1.6 and my S.O is debian lenny. I know this is not asterisk but as the project has not a forum or mail list, I am trying help here! Thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] ReceiveFax
Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] ReceiveFax
Hi, I set up ReceiveFax to answer a specific number (2134-4805) , so , the first caller get the fax signal and transmit the fax normal, but, if another caller to call the same number almost at the same time, it gets the signal as well but the fax is not sent! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 20 Jan 2011 09:13:44 -0600 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ReceiveFax On 01/20/2011 09:00 AM, Flavio Miranda wrote: Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Of course ReceiveFAX can be run on multiple channels at once. What makes you think it cannot? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] Sendind e-mail with Hylafax
Hi all, I know Hylafax is an application and not Asterisk but I'd like to post a problem found in configuring such application and Asterisk. I am able to reveive fax,but , I can't receive it in e-mail. Although I put my e-mail in /etc/hylifax/Dispatch I can't receive. Anybody know where I must to add something else in order to make it works! Thanks in advanced!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] WARNING T.30 ECM carrier not found
Hi list, I have search for a clear explanation about this mensage WARNING T.30 ECM carrier not found, but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] WARNING T.30 ECM carrier not found
CORRECTING: I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXS dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 13 Jan 2011 09:51:24 -0200 Subject: [asterisk-users] WARNING T.30 ECM carrier not found Hi list, I have search for a clear explanation about this mensage WARNING T.30 ECM carrier not found, but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- _ -- 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] Do not disturbe
Hi all,
I am trying to set up DND in my asterisk, I am using the following context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})exten =
*11,2,GotoIf($[${DND} = YES]?*11,3:*11,101)exten =
*11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)exten = *11,4,Playback(beep)exten =
*11,5,Hangup()exten = *11,101,Set(DB(ddisturbe/${CALLERIDNUM})=YES)exten =
*11,102,Playback(beep)exten = *11,103,Hangup() I am testing with a softphone
and when I dial *11, I receive the following log from cli:
Executing [...@a2billing:1] Set(SIP/2015-0187, DND=YES) in new stack
-- Executing [...@a2billing:2] GotoIf(SIP/2015-0187, 1?*11,3:*11,101)
in new stack-- Goto (a2billing,*11,3)-- Executing [...@a2billing:3]
Set(SIP/2015-0187, DB(ddisturbe/)=NO) in new stack-- Executing
[...@a2billing:4] Playback(SIP/2015-0187, beep) in new stack--
SIP/2015-0187 Playing 'beep.gsm' (language 'en')-- Executing
[...@a2billing:5] Hangup(SIP/2015-0187, ) in new stack == Spawn
extension (a2billing, *11, 5) exited non-zero on 'SIP/2015-0187'
Therefore, the facilite is not working!!What I am doing wrong, could somebody
point me out please?!!
Thanks in advanced!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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Re: [asterisk-users] Do not disturbe
I really would like to understand why dont works!
should I to set up any other function? maybe on features?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Tue, 4 Jan 2011 20:08:39 -0500
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Do not disturbe
Flavio Miranda wrote:
Hi all,
I am trying to set up DND in my asterisk, I am using the following
context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})
This is mine:
[dnd]
;***
;* Do not disturb can be set via Asterisk
;* instead of the phones by dialing this
;* number.
;***
exten = 79*,1,Set(DND=${DB(DND/${CALLERID(num)})})
exten = 79*,n,GotoIf($[${DND} = YES]?3:100)
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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[asterisk-users] Dialplan not found
Hi there! Anybody knows why I am receiving this output from CLI: No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) Look like asterisk dont see dialplan? Is it possible to restart it ? Thansk Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] pickup problem
Hi all, I can´t pickup calls on my asterisk. When I try to load app_pickupchan.so I receive following message: Module 'app_pickupchan.so' was not compiled with the same compile-time options as this version of Asterisk It was working fine until few time ago. What is going on? Thanks! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Problem in receiving calls from E1
Hi there!
I am having some difficult in receiving calls from my E1 link using mfcr2. I
can make calls normally , but when I receive an incoming calls, the phone ring
I answer it ,so, I listen busy tone and then the phone ring again and again.
look the log:
-- Executing [4...@from-pstn-te1:1] NoOp(DAHDI/2-1, 1233220567
1233220567) in new stack-- Executing [4...@from-pstn-te1:2]
Dial(DAHDI/2-1, SIP/4801,25) in new stack == Using SIP RTP CoS mark 5
-- Called 4801-- SIP/4801-00f9 is ringing-- SIP/4801-00f9
answered DAHDI/2-1-- Started music on hold, class 'default', on
DAHDI/1-1New MFC/R2 call detected on chan 3.MFC/R2 call offered on chan 3. ANI
= 1233220567, DNIS = 4801, Category = National SubscriberMFC/R2 call has been
accepted on backward channel 3-- Executing [4...@from-pstn-te1:1]
NoOp(DAHDI/3-1, 1233220567 1233220567) in new stack-- Executing
[4...@from-pstn-te1:2] Dial(DAHDI/3-1, SIP/4801,25) in new stack == Using
SIP RTP CoS mark 5-- Called 4801-- SIP/4801-00fa is ringing--
SIP/4801-00fa answered DAHDI/3-1-- Started music on hold, class
'default', on DAHDI/2-1 == Spawn extension (from-pstn-TE1, 4801, 2) exited
non-zero on 'DAHDI/3-1'-- Hungup 'DAHDI/3-1'MFC/R2 call end on channel
3Chan 1 - Far end disconnected. Reason: Normal ClearingMFC/R2 call disconnected
on channel 1-- Stopped music on hold on DAHDI/1-1 == Spawn extension
(from-pstn-TE1, 4801, 2) exited non-zero on 'DAHDI/1-1'MFC/R2 call end on
channel 1-- Hungup 'DAHDI/1-1'Chan 2 - Far end disconnected. Reason: Normal
ClearingMFC/R2 call disconnected on channel 2-- Stopped music on hold on
DAHDI/2-1 == Spawn extension (from-pstn-TE1, 4801, 2) exited non-zero on
'DAHDI/2-1'MFC/R2 call end on channel 2-- Hungup 'DAHDI/2-1'
Thats my dial plan:
[from-pstn-TE1]exten = _X.,1,Noop(${CALLERID(all)})exten =
_X.,n,Dial(SIP/${EXTEN},25)exten = _X.,n,VoiceMail(${EXTEN},u)exten =
_X.,n,Hangup
Thanks for any help.
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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[asterisk-users] d-link dvg-3032 guidance
Hi all,
Anyboby has some experience with D-link dvg-3032s with asterisk and could
give me some support.
I am using this dial plan :
exten = _X. ,1,Dial(SIP/${EXTEN:0...@192.168.0.60,30)But I receive the mesagem
from C.O that the number is incorrect.
And about incoming calls, I put I asterisk extension on FXO Hot-line, the call
is shown on status of the dvg-3032 but nothing is shown on asterisk CLI.
Thanks for any help.
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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[asterisk-users] Incoming calls
Hi all, I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore, when I get call, it always is ringing on the first extensions, dont matter trunk . Anybody could teach me how can I organize that ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Incoming calls
Hi Steve, thanks for the tips Better bait = better fish ! As you said, I am in the right track. Looking to dahdi show channles , I realized that all the trunks was in the same context. So, I have changed this and everything works! thanks you !! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 18 Nov 2010 11:53:26 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming calls On Thu, 18 Nov 2010, Flavio Miranda wrote: I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore, when I get call, it always is ringing on the first extensions, dont matter trunk. Anybody could teach me how can I organize that ? 0) Use a subject that gives a clue what you're looking for. Almost everybody has had a question about an incomig call at some point in time. Better bait = better fish. 1) It sounds like you have a clue about how to do it and are on the right track. 2) Including some details like the console output from: zap show channel 1 (I'm a 1.2 Luddite.) zap show channel 2 zap show channel 3 zap show channel 4 as well as the console log from a call coming in on each channel will help in assisting you in resolving this issue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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] SMS Gateway
Hi list, Anyone has some guidance in how can I project a SMS gateway with Asterisk. I mean, some good web link,pdf or something like that? Thanks in advanced!!Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormirandaru -- _ -- 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] Call using password
Thanks! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Fri, 5 Nov 2010 17:25:25 -0700 From: cwall...@lodgingcompany.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call using password On Fri, 5 Nov 2010 22:05:28 -0200 Flavio Miranda flaviormira...@hotmail.com wrote: What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Use the Authenticate app: Authenticate(1234); For more info, run 'core show application Authenticate' on your asterisk CLI. -- _ -- 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 -- _ -- 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] Call using password
Hi, What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 + Mediatrix
Hi all, I have configured a Mediatrix 8 FXS with Asterisk . The extensions on Mediatrix are able to do external calls and receive calls from softphone quite normal . However, when it originate internal calls, the call hung up as soon as we pick up the phone, don´t matter if other end is an mediatrix extension or even a softphone. Best regards!! Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] E1 configuration
hi, So, I think it depend of what environment are you setting up your link . In my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change dahdi-channels parameter,chan_dahdi.conf , system.conf as well. If you need I can send you such configuration. good look! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Tue, 26 Oct 2010 14:24:13 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration hi my friend would ou say what did you do for solving the problem? because i use a digium te121p and have many problems. thanks in advance On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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 -- _ -- 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 -- free is to know that you have a different option -- _ -- 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 -- _ -- 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
Re: [asterisk-users] E1 configuration
Hi, /etc/dahdi/system.conf Att,# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) HDB3/ span=1,1,0,cas,hdb3cas=1-15:1101dchan=16cas=17-31:1101#echocanceller=mg2,1-15,17-31 /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] usecallerid=yescallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yessignalling=mfcr2mfcr2_variant=brmfcr2_get_ani_first=nomfcr2_max_ani=20mfcr2_max_dnis=4mfcr2_category=national_subscribermfcr2_logdir=span1mfcr2_call_files=yesmfcr2_logging=allmfcr2_mfback_timeout=-1mfcr2_metering_pulse_timeout=-1mfcr2_allow_collect_calls=yesmfcr2_double_answer=nomfcr2_immediate_accept=yesmfcr2_forced_release=nomfcr2_charge_calls=yes;language=pt_BRcontext=Saida-de-ligacoesgroup=0callgroup=0pickupgroup=0channel = 1-15,17-31immediate=no#include dahdi-channels.conf /etc/asterisk/dahdi-channels.conf ; Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) HDB3/group=0,11context=Saida-de-ligacoesswitchtype = nationalsignalling = pri_cpechannel = 1-15,17-31context = defaultgroup = 63 Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Wed, 27 Oct 2010 00:15:01 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration dear please send these configurations. thanks On Tue, Oct 26, 2010 at 3:04 PM, Flavio Miranda flaviormira...@hotmail.com wrote: hi, So, I think it depend of what environment are you setting up your link . In my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change dahdi-channels parameter,chan_dahdi.conf , system.conf as well. If you need I can send you such configuration. good look! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Tue, 26 Oct 2010 14:24:13 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration hi my friend would ou say what did you do for solving the problem? because i use a digium te121p and have many problems. thanks in advance On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- _ -- 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 -- Best Regards Shakeel Abbas
Re: [asterisk-users] E1 configuration
Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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 -- _ -- 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] E1 configuration
Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5-- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] E1 configuration
Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5-- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- _ -- 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] Incoming calls
Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5-- Executing [33220...@local:1] Dial(SIP/4804-001a, DAHDI/g11/33220567,,T) in new stack == Everyone is busy/congested at this time (1:0/1/0)-- Auto fallthrough, channel 'SIP/4804-001a' status is 'CONGESTION'This is my dahdi show status:Digium Wildcard TE110P T1/E1 Card 0 REC 0 0 0 CAS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)thi´s my dahdi show channels: asterisk*CLI dahdi show channels Chan Extension Context Language MOH InterpretBlockedState pseudodefault default In Service 1 4800 default default In Service 2 4800 defaultdefault In Service 3 4805 defaultdefault In Service 4defaultdefault In Service 5defaultdefault In Service 6defaultdefault In Service 7default default In Service 8default default In Service 9default default In Service 10 defaultdefault In Service 11 defaultdefault In Service 12defaultdefault In Service 13defaultdefault In Service 14defaultdefault In Service 15defaultdefault In Service 17default default In Service 18default default In Service 19default default In Service 20 defaultdefault In Service 21 defaultdefault In Service 22defaultdefault In Service 23defaultdefault In Service 24defaultdefault In Service 25defaultdefault In Service 26default default In Service 27default default In Service 28default default In Service 29 defaultdefault In Service 30 defaultdefault In Service 31defaultdefault In Service *In my incoming call , the log is: MFC/R2 call offered on chan 1. ANI = 1221341400, DNIS = 4804, Category = National SubscriberNew MFC/R2 call detected on chan 2. and don't ring nowhere! Thanks for help!Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] FW: Incoming calls
Hi, After some changes, the status now is: == Using SIP RTP CoS mark 5-- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,T) in new stack == Everyone is busy/congested at this time (1:0/0/1)-- Auto fallthrough, channel 'SIP/4804-' status is 'CHANUNAVAIL'MFC/R2 call disconnected on channel 1 ThanksAtt, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Subject: Incoming calls Date: Thu, 21 Oct 2010 17:59:35 -0200 Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5-- Executing [33220...@local:1] Dial(SIP/4804-001a, DAHDI/g11/33220567,,T) in new stack == Everyone is busy/congested at this time (1:0/1/0)-- Auto fallthrough, channel 'SIP/4804-001a' status is 'CONGESTION'This is my dahdi show status:Digium Wildcard TE110P T1/E1 Card 0 REC 0 0 0 CAS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)thi´s my dahdi show channels: asterisk*CLI dahdi show channels Chan Extension Context Language MOH InterpretBlockedState pseudodefault default In Service 1 4800 default default In Service 2 4800 defaultdefault In Service 3 4805 defaultdefault In Service 4defaultdefault In Service 5defaultdefault In Service 6defaultdefault In Service 7default default In Service 8default default In Service 9default default In Service 10 defaultdefault In Service 11 defaultdefault In Service 12defaultdefault In Service 13defaultdefault In Service 14defaultdefault In Service 15defaultdefault In Service 17default default In Service 18default default In Service 19default default In Service 20 defaultdefault In Service 21 defaultdefault In Service 22defaultdefault In Service 23defaultdefault In Service 24defaultdefault In Service 25defaultdefault In Service 26default default In Service 27default default In Service 28default default In Service 29 defaultdefault In Service 30 defaultdefault In Service 31defaultdefault In Service *In my incoming call , the log is: MFC/R2 call offered on chan 1. ANI = 1221341400, DNIS = 4804, Category = National SubscriberNew MFC/R2 call detected on chan 2. and don't ring nowhere! Thanks for help!Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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
[asterisk-users] dahdi_genconf
Hi , Please, I am trying to understand the hardware installation on asterisk and I have some doubt. If I uncomment the hardware type in /etc/dahdi/modules and then I run the dahdi_genconf , It create the dahdi_channels and system.conf. Therefore, it is created with a kind of signalling that is not used in my country. Can I edit it? regards! Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] dahdi_genconf
Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Just one more question, what it means the RED under alarms when I type dahdi show status. It should be OK? Thanks for your guidance! Date: Tue, 19 Oct 2010 22:38:25 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi_genconf On 10/19/10 10:03 PM, Flavio Miranda wrote: Please, I am trying to understand the hardware installation on asterisk and I have some doubt. If I uncomment the hardware type in /etc/dahdi/modules and then I run the dahdi_genconf , It create the dahdi_channels and system.conf. Therefore, it is created with a kind of signalling that is not used in my country. Can I edit it? Yes..dahdi_genconf just makes the best guess it can based on the hardware that is loaded. You should feel free to edit those files. Keep in mind that if you rerun dahdi_genconf your changes could be overwritten. dahdi_genconf can be thought of as a tool to get you started with your configuration. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] Meetme
Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Meetme
hi Bakko, thanks! Acctualy, I had tried this but still don´t works! [conference]exten = 1001,3,MeetMe(1001,ipdM)exten = 1001,4,Set(CHANNEL(language)=pt_BR)exten = 1001,5,Playback(pt_BR/vm-goodbye)exten = 1001,6,Hangup this is my config! What´s wrong? thanks again! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: asannu...@gmail.com To: asterisk-users@lists.digium.com Date: Sun, 17 Oct 2010 16:36:34 -0500 Subject: Re: [asterisk-users] Meetme Hi Flavio, try with this funtion before the line with the english meetme application Set(CHANNEL(language)=en) and Set(CHANNEL(language)=pr) before the line with the portugues meetme application Regards - Bakko -- _ -- 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 -- _ -- 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] Meetme
It works!!! Thanks a lot! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: asannu...@gmail.com To: asterisk-users@lists.digium.com Date: Sun, 17 Oct 2010 17:56:37 -0500 Subject: Re: [asterisk-users] Meetme Hi Flavio is: [conference] exten = 1001,3,Set(CHANNEL(language)=pt_BR) exten = 1001,4,MeetMe(1001,ipdM) exten = 1001,5,Playback(vm-goodbye) exten = 1001,6,Hangup Regards - Bakko -- _ -- 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 -- _ -- 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] Unable to specify channel 5: No such device or address
Hi, I am trying to set up two bords on my server: TDM410p(This on is ok) and TE110p. This is my system.conf # Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER) fxsks=1,2,3,4 # Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,1,0,cas,hdb3cas=1-15:1101cas=17-31:1101dchan=16loadzone=brdefaultzone=br And this is my chan_dahdi.conf: [channels] #include dahdi-channels.conf ;General optionsusecallerid = yeshidecallerid = nocallwaiting = yesthreewaycalling = yestransfer = yesechocancel = yesechocancelwhenbridged = yesrxgain = 3.0txgain = 3.0 ;FXO Modules group = 1;echocancel = yessignalling = fxs_kscontext = Troncos-Analogicoschannel = 1,2,3,4 ;E1 Modules signalling=mfcr2mfcr2_variant=brmfcr2_get_ani_first=nomfcr2_max_ani=20mfcr2_max_dnis=20mfcr2_category=national_subscribermfcr2_logdir=span1mfcr2_logging=all And my dahdi-chennel.conf ; Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER);;; line=1 WCTDM/0/0 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 1callerid=group=context=default ;;; line=2 WCTDM/0/1 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 2callerid=group=context=default ;;; line=3 WCTDM/0/2 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 3callerid=group=context=default ;;; line=4 WCTDM/0/3 FXSKS (In use)signalling=fxs_kscallerid=asreceivedgroup=0context=from-pstnchannel = 4callerid=group=context=default ; Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 signalling = mfcr2channel = 1-15,17-31group=1context=Troncos-Digitais In dahdi show status, is shown only the TDM410p board.In dahdi show channels is shown only four analogic trunk If I type: dahdi restart, I see the following messages: [Oct 13 14:36:50] WARNING[930]: chan_dahdi.c:2124 dahdi_open: Unable to specify channel 5: No such device or address This is repeated a lot help!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] OpenR2
Thanks for while!! I will do that! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 11 Oct 2010 12:06:04 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OpenR2 On Mon, Oct 11, 2010 at 01:08:50AM -0300, Flavio Miranda wrote: Hi all, Is it Openr2 supported by asterisk 1.6.2 without pach instalation ? Yes. Provided you have libopenr2. I am a little bit confuse about that. My asterisk 1.6.2 show me the following warning: Unknown signalling method 'mfcr2' at line 29. Most likely you didn't have libopenr2 (or its development headers, depending on the installaation type) I had downloaded and instaled openr2-1.3.0 but the messages is still shown. Which files I must to change in order to have everything working properly. A. You need to re-run the configure script of asterisk and rebuild it. B. How have you installaed openr2? From source or from a binary package? If from a binary package, you probably also need the -dev / -devel package. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 -- _ -- 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] Dahdi missing
Thanks! I am from Brasil and intend to use mfc r2 and now I have following conditions: Digium Wildcard TE110P T1/E1 Card 0 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)Wildcard TDM410P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) look like good! The only thing strange is that dahdi-channels.conf don´t appear after dahdi_genconf command. Is it indispensable? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 10 Oct 2010 00:58:48 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi missing On 10/10/10 12:11 AM, Flavio Miranda wrote: Actully, my system.conf in /etc/dahdi was wrong or at least , imcomplete. I am trying for a long time to configure one TDM 410p board and one TE110p om the server . so far without success! In my opinion, the easiest way to get jump started is with dahdi_genconf that is installed as part of the dahdi-tools package. 1) Assuming you are not in North America you most likely want lines like options wctdm24xxp companding=alaw and options wcte11xp t1e1override=1 in /etc/modprobe.d/dahdi.conf. 2) modprobe wcte12xp modprobe wctdm24xxp (make sure the order here matches the order in /etc/dahdi/modules) 3) run dahdi_genconf. This will create /etc/dahdi/system.conf and an /etc/asterisk/dahdi-channels.conf that you can use in your chan_dahdi.conf (either copy / or include). 4) OPTIONAL but recommended: Comment out all the other drivers in /etc/dahdi/modules (leaving only wcte11xp and wctdm24xxp) 5) '/etc/init.d/dahdi restart' and run dahdi_scan to make sure all the expected channels came up. You may have to tweak the settings some and most definitely will if you're using the TE110P as an interface to a channel bank (but I doubt you are if you also have a TDM410 installed). But at least you'll have a good start at this point. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] Dahdi missing
hi, My tdm410p is working Thank ou very much! Now I am looking for how to set up my TE110p. If I have some error, I will ask to the List for help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 10 Oct 2010 16:02:12 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi missing On Sun, Oct 10, 2010 at 12:04:27AM -0500, Shaun Ruffell wrote: On 10/9/10 11:54 PM, Flavio Miranda wrote: Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. [snip] Anybody know what file control the presence or not od dahdi in asterisk cli? I don't know what version of Asterisk you're running, but I'll take a guess and say the output of module load chan_dahdi.so on the Asterisk CLI will result in an error that will give you the clue you need. Also: chan_dahdi is likely to have automatically attempted to load. If it failed, it had already registered some commands. Thus re-attempting to load it will automatically fail. Try: module unload chan_dahdi.so module load chan_dahdi.so -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 -- _ -- 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] OpenR2
Hi all, Is it Openr2 supported by asterisk 1.6.2 without pach instalation ? I am a little bit confuse about that. My asterisk 1.6.2 show me the following warning: Unknown signalling method 'mfcr2' at line 29. I had downloaded and instaled openr2-1.3.0 but the messages is still shown. Which files I must to change in order to have everything working properly. Best regards! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Dahdi missing
Hi, Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. !ael agentagi cdr channel cli config console core database devstate dialplan dnsmgr dundifeatures file group hangup help http iax2 indication keys locallogger manager meetme mgcp minivm mixmonitor module moh no originateparkedcalls phoneprovpri queuerealtime reload rtcp rtp say sip skinny sla stun timing transcoder udptlulimit unistim voicemail Anybody know what file control the presence or not od dahdi in asterisk cli? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Dahdi missing
Hi Shaun, Thanks! Actully, my system.conf in /etc/dahdi was wrong or at least , imcomplete. I am trying for a long time to configure one TDM 410p board and one TE110p om the server . so far without success! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 10 Oct 2010 00:04:27 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi missing On 10/9/10 11:54 PM, Flavio Miranda wrote: Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. [snip] Anybody know what file control the presence or not od dahdi in asterisk cli? I don't know what version of Asterisk you're running, but I'll take a guess and say the output of module load chan_dahdi.so on the Asterisk CLI will result in an error that will give you the clue you need. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] Dahdi error
You´re right!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Fri, 8 Oct 2010 00:16:58 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi error On 10/7/10 2:07 PM, Flavio Miranda wrote: asterisk:/etc/asterisk# /etc/init.d/dahdi start Loading DAHDI hardware modules: FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Device or resource busy wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: error wctc4xxp: done xpp_usb: done Error: missing /dev/dahdi! Best guess based on the information you provided: zaptel was installed on this machine and is already loaded and registered major number 196. That would explain both the Device or resource busy error, and the fact that dahdi failed to load, yet most of the board drivers appear to have loaded (since the zaptel ones probably loaded up and the wcb4xxp driver did not load). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] Dahdi error
Hi all, What hell hapen here? asterisk:/etc/asterisk# /etc/init.d/dahdi startLoading DAHDI hardware modules:FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Device or resource busy wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: error wctc4xxp: done xpp_usb: doneError: missing /dev/dahdi! When I installed the board, everything was going ok,but, suddenly, everything is going wrong Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Web-meetme
Hi there! I am trying to configure Web-meetme on Asterisk 1.6. I have followed the README and everything looks ok,therefore, when I try to open the webpage appear the folowing messages:DB Error: connect failed Testing with a php script, the message Connected successfully is shown. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Module reload
Hi all, Every time I reload my asterisk it fall down and the following message appear on log: parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf If I comment that line, it change to other line. There are some thing wrong with my dahdi? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] Module reload
Asterisk:/var/log/asterisk# pico /etc/asterisk/chan_dahdi.conf ; DAHDI telephony ;language=en ;echocancel=yes echocancelwhenbridged=yes ss7type = itu ss7_called_nai=dynamic ss7_calling_nai=dynamic ;General options usecallerid = yes hidecallerid = no callwaiting = yes threewaycalling = yes transfer = yes echocancel = yes rxgain = 0.0 txgain = 0.0 ;FXO Modules group = 1 echocancel = yes signalling = fxs_ks context = local channel = 1 That´s it!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Mon, 4 Oct 2010 15:10:19 +0100 Subject: Re: [asterisk-users] Module reload On Monday 04 Oct 2010, Flavio Miranda wrote: Hi all, Every time I reload my asterisk it fall down and the following message appear on log: parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf If I comment that line, it change to other line. There are some thing wrong with my dahdi? It's more than likely a problem with your chan_dahdi.conf file -- which, by the way, you appear to have forgotten to attach. -- AJS -- _ -- 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 -- _ -- 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] Module reload
You are right! Thanks! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 4 Oct 2010 16:39:24 +0100 From: paulo.r.san...@sapo.pt To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Module reload Hello, Flavio Miranda wrote: Asterisk:/var/log/asterisk# pico /etc/asterisk/chan_dahdi.conf [...] You're missing the context [channels] at the start. Best regards, Paulo Santos -- _ -- 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 -- _ -- 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] a2billing
Hi all, I am trying to integrate a2b with asterisk 1.6, but ,when i try to do external call, I receive this mensage: -- Executing [01221341...@ramais:1] AGI(SIP/3000-b5ba6e80, a2billing.php,2) in new stack-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/3000-b5ba6e80AGI Script a2billing.php completed, returning 0-- Auto fallthrough, channel 'SIP/3000-b5ba6e80' status is 'UNKNOWN Anybody know what could be happen? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] a2billing
[ramais]
include = internalinclude = externalinclude = conference
[internal]
exten = 3000,1,DIAL(SIP/3000,10)exten = 3000,2,VoiceMail(3000,u)
exten = 3003,1,DIAL(SIP/3003,30)exten = 3003,2,VoiceMail(3003,u)
exten = 3004,1,DIAL(SIP/3004,10)exten = 3004,2,VoiceMail(3004,u)
exten = 3005,1,DIAL(SIP/3005,10)exten = 3005,2,VoiceMail(3005u)
exten = 1001,1,DIAL(SIP/1001,10)exten = 1001,2,VoiceMail(1001u)
exten = 3999,1,VoiceMailMain($(CALLERID(num))
[external]
;exten = _0NXXX,1,DIAL(SIP/10.201.201.254/${EXTEN},20,rt);exten =
_0NXXX,1,AGI,a2billing.phpexten = _X.,1,AGI(a2billing.php,2)
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Thu, 30 Sep 2010 14:59:38 -0500
Subject: Re: [asterisk-users] a2billing
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursday, September 30, 2010
2:44 PM
To: Asterisk Asterisk
Subject: [asterisk-users]
a2billing
Hi all,
I am trying to integrate a2b with asterisk 1.6, but
,when i try to do external call, I receive this mensage:
-- Executing [01221341...@ramais:1]
AGI(SIP/3000-b5ba6e80, a2billing.php,2) in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/a2billing.php
-- SIP/3000-b5ba6e80AGI Script
a2billing.php completed, returning 0
-- Auto fallthrough, channel
'SIP/3000-b5ba6e80' status is 'UNKNOWN
Anybody know what could be happen?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
It
appears that the AGI is completing successfully and you have a dialplan issue
after that. Please post the dialplan snippet.
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[asterisk-users] A2billing
Hi, I am trying to configure a2billing 1.8 in my asterisk 1.6 but no value to DIALPREFIX and DESTINATION PREFIX is accepted when I try to create a RATE. thanks! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] one way audio for xlite clients behind NAT
If you are using linux firewall, try this, it was very usefull to me: iptables -t nat -A PREROUTING -i ethx -p tcp --dport 5060 -j DNAT --to ip_phoneiptables -t nat -A PREROUTING -i ethx -p udp --dport 5060 -j DNAT --to iip_phoneiptables -A FORWARD -p TCP --dport 5060 -j ACCEPTiptables -A FORWARD -p UDP --dport 5060 -j ACCEPT Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 16 Sep 2010 18:45:38 -0400 From: paul.belan...@polybeacon.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] one way audio for xlite clients behind NAT On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson tomfma...@gmail.com wrote: The server is not behind NAT only the client above is Sounds like a phone (not asterisk) issue then, make sure you have setup your NAT and port forwarding properly on the client side. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- _ -- 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] incoming call FXO
Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler I have not this 's' extension. Anybody knows what happen? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] incoming call FXO
Ok. Problem solved . Thank you very much!!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Wed, 15 Sep 2010 09:56:36 -0400 From: zisha...@gmail.com To: kpflem...@digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] incoming call FXO As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten = s,1,Verbose( - - - Call received - - - ) exten = s,n,Playback(hello-world) extent = s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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 -- _ -- 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
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Hi all, I am trying to set up Web-meetme in Asterisk 1.6. After some attemps, I am receiving the message:DB Error: connect failed What could be ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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
