[AsteriskBrasil] RES: RES: Ligação cai
Fernando Trilha
ftrilha em gmail.com
Quarta Agosto 14 10:08:50 BRT 2013
Opa, foi mal, achei que tinha colocado, vamos lá
Trunk de saida é pela vono.
Asterisk*CLI> sip show peer vono
* Name : vono
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : entrada
Subscr.Cont. : <Not set>
Language : pt_BR
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : ftrilha
FromDomain : vono.net.br Port 5060
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : vono.net.br
Addr->IP : 177.159.181.50:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: ftrilha
SIP Options : (none)
Codecs : 0x40c (ulaw|alaw|ilbc)
Codec Order : (alaw:20,ilbc:30,ulaw:20)
Auto-Framing : No
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Originate
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
Asterisk*CLI> exit
O debug nao consegui pegar todo.
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:177.159.181.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 200.193.70.69:5070
;received=200.193.70.69;branch=z9hG4bK7734503c;rport=5070
From: "9962" <sip:ftrilha em vono.net.br>;tag=as3e619f94
To: <sip:04832422500 em vono.net.br:5060>;tag=as0ea32ab4
Call-ID: 3d17f51128684d3e6577ffc041e6485c em vono.net.br
CSeq: 103 INVITE
User-Agent: Plataforma Vono
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 177.159.181.50:5060:
ACK sip:04832422500 em 177.159.181.50:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 200.193.70.69:5070;branch=z9hG4bK7734503c;rport
Max-Forwards: 70
From: "9962" <sip:ftrilha em vono.net.br>;tag=as3e619f94
To: <sip:04832422500 em vono.net.br:5060>;tag=as0ea32ab4
Contact: <sip:ftrilha em 200.193.70.69:5070>
Call-ID: 3d17f51128684d3e6577ffc041e6485c em vono.net.br
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.23.0
Content-Length: 0
---
Scheduling destruction of SIP dialog '
3d17f51128684d3e6577ffc041e6485c em vono.net.br' in 32000 ms (Method: INVITE)
Retransmitting #1 (NAT) to 179.252.225.2:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-;received=179.252.225.2;rport=5060
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP>;tag=as24f38f74
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:179.252.225.2:5060 --->
ACK sip:04832422500 em 200.193.70.69:5070;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-
Max-Forwards: 70
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Retransmitting #2 (NAT) to 179.252.225.2:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-;received=179.252.225.2;rport=5060
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP>;tag=as24f38f74
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #3 (NAT) to 179.252.225.2:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-;received=179.252.225.2;rport=5060
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP>;tag=as24f38f74
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:179.252.225.2:5060 --->
ACK sip:04832422500 em 200.193.70.69:5070;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-
Max-Forwards: 70
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:179.252.225.2:5060 --->
ACK sip:04832422500 em 200.193.70.69:5070;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-
Max-Forwards: 70
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Retransmitting #4 (NAT) to 179.252.225.2:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-;received=179.252.225.2;rport=5060
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP>;tag=as24f38f74
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:179.252.225.2:5060 --->
<------------->
<--- SIP read from UDP:179.252.225.2:5060 --->
ACK sip:04832422500 em 200.193.70.69:5070;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-
Max-Forwards: 70
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Retransmitting #5 (NAT) to 179.252.225.2:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-;received=179.252.225.2;rport=5060
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP>;tag=as24f38f74
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:179.252.225.2:5060 --->
ACK sip:04832422500 em 200.193.70.69:5070;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-
Max-Forwards: 70
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Reliably Transmitting (NAT) to 172.16.100.38:47554:
OPTIONS sip:9960 em 172.16.100.38:47554;rinstance=cc7c4c76d033eabf SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5070;branch=z9hG4bK38ecf3df;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 192.168.0.101:5070>;tag=as0fc82013
To: <sip:9960 em 172.16.100.38:47554;rinstance=cc7c4c76d033eabf>
Contact: <sip:asterisk em 192.168.0.101:5070>
Call-ID: 160a2b12139b41f47b8d73234568c230 em 192.168.0.101:5070
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0
Date: Wed, 14 Aug 2013 13:06:47 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:172.16.100.38:47554 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:5070;branch=z9hG4bK38ecf3df;rport=5070
Contact: <sip:172.16.100.38:47554>
To: <sip:9960 em 172.16.100.38:47554;rinstance=cc7c4c76d033eabf>;tag=64039574
From: "asterisk"<sip:asterisk em 192.168.0.101:5070>;tag=as0fc82013
Call-ID: 160a2b12139b41f47b8d73234568c230 em 192.168.0.101:5070
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '
160a2b12139b41f47b8d73234568c230 em 192.168.0.101:5070' Method: OPTIONS
Retransmitting #6 (NAT) to 179.252.225.2:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-;received=179.252.225.2;rport=5060
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP>;tag=as24f38f74
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Aug 14 10:06:49] WARNING[16819]: chan_sip.c:3982 retrans_pkt:
Retransmission timeout reached on transmission
OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU. for seqno 2 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
Really destroying SIP dialog 'OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.'
Method: CANCEL
<--- SIP read from UDP:179.252.225.2:5060 --->
ACK sip:04832422500 em 200.193.70.69:5070;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 179.252.225.2:5060
;branch=z9hG4bK-d8754z-c338cc0ec9d805e7-1---d8754z-
Max-Forwards: 70
To: <sip:04832422500 em 200.193.70.69:5070>;transport=UDP
From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=cac7ba28
Call-ID: OTVhY2U0MmJlNzhkZmU1ZTc4NDNlY2MyNDI3N2FiNGU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
2013/8/14 Fernando - NextBilling IP Solutions <fernando em nextbilling.com.br>
> Opa chará.****
>
> ** **
>
> Falta o *sip show peer XXXX* do trunk que ta completando a chamada.****
>
> ** **
>
> E também falta a saída completa do *sip set debug on*****
>
> ** **
>
> Com essas saídas, da pra gente ver a configuração completa do peer e do
> ramal, e o siptrace vai complementar as informações e vai ser possível
> saber se o diálogo SIP ta sendo formado corretamente obedecendo as regras
> de NAT, que no caso é o seu problema.****
>
> ** **
>
> Atenciosamente,****
>
> ** **
>
> *Fernando da Silva Santos*
>
> *CEO* – Chief Executive Officer****
>
> *NextBilling IP Solutions*
>
> * *
>
> *SP: *+55 (11) 3522-9200****
>
> *RJ: *+55 (21) 4063-8854****
>
> *Tollfree:* 0800 580-9200****
>
> http://www.nextbilling.com.br****
>
> ** **
>
> *De:* asteriskbrasil-bounces em listas.asteriskbrasil.org [mailto:
> asteriskbrasil-bounces em listas.asteriskbrasil.org] *Em nome de *Fernando
> Trilha
> *Enviada em:* quarta-feira, 14 de agosto de 2013 09:23
> *Para:* asteriskbrasil em listas.asteriskbrasil.org
> *Assunto:* Re: [AsteriskBrasil] RES: Ligação cai****
>
> ** **
>
> Fernando irei postar os comando abaixo....****
>
> ** **
>
> ** **
>
> root em Asterisk:~# curl -dump http://ipecho.net/plain;echo****
>
> 200.193.70.69****
>
> root em Asterisk:~# asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
> Asterisk 1.8.23.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
> ****
>
> Created by Mark Spencer <markster em digium.com>****
>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
> details.****
>
> This is free software, with components licensed under the GNU General
> Public****
>
> License version 2 and other licenses; you are welcome to redistribute it
> under****
>
> certain conditions. Type 'core show license' for details.****
>
> =========================================================================*
> ***
>
> Connected to Asterisk 1.8.23.0 currently running on Asterisk (pid = 16799)
> ****
>
> Verbosity is at least 33****
>
> Asterisk*CLI> sip show settings****
>
> ** **
>
> ** **
>
> Global Settings:****
>
> ----------------****
>
> UDP Bindaddress: 0.0.0.0:5070****
>
> TCP SIP Bindaddress: Disabled****
>
> TLS SIP Bindaddress: Disabled****
>
> Videosupport: No****
>
> Textsupport: No****
>
> Ignore SDP sess. ver.: No****
>
> AutoCreate Peer: No****
>
> Match Auth Username: No****
>
> Allow unknown access: Yes****
>
> Allow subscriptions: Yes****
>
> Allow overlap dialing: Yes****
>
> Allow promisc. redir: No****
>
> Enable call counters: No****
>
> SIP domain support: No****
>
> Realm. auth: No****
>
> Our auth realm asterisk****
>
> Use domains as realms: No****
>
> Call to non-local dom.: Yes****
>
> URI user is phone no: No****
>
> Always auth rejects: Yes****
>
> Direct RTP setup: No****
>
> User Agent: Asterisk PBX 1.8.23.0****
>
> SDP Session Name: Asterisk PBX 1.8.23.0****
>
> SDP Owner Name: root****
>
> Reg. context: (not set)****
>
> Regexten on Qualify: No****
>
> Legacy userfield parse: No****
>
> Caller ID: asterisk****
>
> From: Domain:****
>
> Record SIP history: Off****
>
> Call Events: Off****
>
> Auth. Failure Events: Off****
>
> T.38 support: No****
>
> T.38 EC mode: Unknown****
>
> T.38 MaxDtgrm: -1****
>
> SIP realtime: Disabled****
>
> Qualify Freq : 60000 ms****
>
> Q.850 Reason header: No****
>
> Store SIP_CAUSE: No****
>
> ** **
>
> Network QoS Settings:****
>
> ---------------------------****
>
> IP ToS SIP: CS0****
>
> IP ToS RTP audio: CS0****
>
> IP ToS RTP video: CS0****
>
> IP ToS RTP text: CS0****
>
> 802.1p CoS SIP: 4****
>
> 802.1p CoS RTP audio: 5****
>
> 802.1p CoS RTP video: 6****
>
> 802.1p CoS RTP text: 5****
>
> Jitterbuffer enabled: No****
>
> ** **
>
> Network Settings:****
>
> ---------------------------****
>
> SIP address remapping: Enabled using externhost****
>
> Externhost: 200.193.70.69****
>
> Externaddr: 200.193.70.69:0****
>
> Externrefresh: 15****
>
> Localnet: 192.168.0.0/255.255.255.0****
>
> 172.16.100.0/255.255.254.0****
>
> ** **
>
> Global Signalling Settings:****
>
> ---------------------------****
>
> Codecs: 0x4 (ulaw)****
>
> Codec Order: ulaw:20****
>
> Relax DTMF: No****
>
> RFC2833 Compensation: No****
>
> Symmetric RTP: No****
>
> Compact SIP headers: No****
>
> RTP Keepalive: 0 (Disabled)****
>
> RTP Timeout: 0 (Disabled)****
>
> RTP Hold Timeout: 0 (Disabled)****
>
> MWI NOTIFY mime type: application/simple-message-summary****
>
> DNS SRV lookup: Yes****
>
> Pedantic SIP support: Yes****
>
> Reg. min duration 60 secs****
>
> Reg. max duration: 3600 secs****
>
> Reg. default duration: 120 secs****
>
> Outbound reg. timeout: 20 secs****
>
> Outbound reg. attempts: 0****
>
> Notify ringing state: Yes****
>
> Include CID: No****
>
> Notify hold state: No****
>
> SIP Transfer mode: open****
>
> Max Call Bitrate: 384 kbps****
>
> Auto-Framing: No****
>
> Outb. proxy: <not set>****
>
> Session Timers: Originate****
>
> Session Refresher: uas****
>
> Session Expires: 1800 secs****
>
> Session Min-SE: 90 secs****
>
> Timer T1: 500****
>
> Timer T1 minimum: 100****
>
> Timer B: 32000****
>
> No premature media: Yes****
>
> Max forwards: 70****
>
> ** **
>
> Default Settings:****
>
> -----------------****
>
> Allowed transports: UDP****
>
> Outbound transport: UDP****
>
> Context: entrada****
>
> Force rport: Yes****
>
> DTMF: rfc2833****
>
> Qualify: 0****
>
> Use ClientCode: No****
>
> Progress inband: Never****
>
> Language: pt_BR****
>
> MOH Interpret: default****
>
> MOH Suggest:****
>
> Voice Mail Extension: asterisk****
>
> ** **
>
> ----****
>
> ** **
>
> Asterisk*CLI> sip show peer 9962****
>
> ** **
>
> ** **
>
> * Name : 9962****
>
> Secret : <Set>****
>
> MD5Secret : <Not set>****
>
> Remote Secret: <Not set>****
>
> Context : entrada****
>
> Subscr.Cont. : <Not set>****
>
> Language : pt_BR****
>
> AMA flags : Unknown****
>
> Transfer mode: open****
>
> CallingPres : Presentation Allowed, Not Screened****
>
> Callgroup : 1****
>
> Pickupgroup : 1****
>
> MOH Suggest :****
>
> Mailbox : 9962****
>
> VM Extension : asterisk****
>
> LastMsgsSent : 0/0****
>
> Call limit : 0****
>
> Max forwards : 0****
>
> Dynamic : Yes****
>
> Callerid : "" <9962>****
>
> MaxCallBR : 384 kbps****
>
> Expire : 54****
>
> Insecure : no****
>
> Force rport : Yes****
>
> ACL : No****
>
> DirectMedACL : No****
>
> T.38 support : No****
>
> T.38 EC mode : Unknown****
>
> T.38 MaxDtgrm: -1****
>
> DirectMedia : No****
>
> PromiscRedir : No****
>
> User=Phone : No****
>
> Video Support: No****
>
> Text Support : No****
>
> Ign SDP ver : No****
>
> Trust RPID : No****
>
> Send RPID : No****
>
> Subscriptions: Yes****
>
> Overlap dial : Yes****
>
> DTMFmode : rfc2833****
>
> Timer T1 : 500****
>
> Timer B : 32000****
>
> ToHost :****
>
> Addr->IP : 179.252.225.2:5060****
>
> Defaddr->IP : (null)****
>
> Prim.Transp. : UDP****
>
> Allowed.Trsp : UDP****
>
> Def. Username: 9962****
>
> SIP Options : (none)****
>
> Codecs : 0xe (gsm|ulaw|alaw)****
>
> Codec Order : (ulaw:20,alaw:20,gsm:20)****
>
> Auto-Framing : No****
>
> Status : OK (287 ms)****
>
> Useragent : Zoiper r18976****
>
> Reg. Contact :
> sip:9962 em 179.252.225.2:5060;rinstance=13ef272eed0550ea;transport=UDP****
>
> Qualify Freq : 60000 ms****
>
> Sess-Timers : Originate****
>
> Sess-Refresh : uas****
>
> Sess-Expires : 1800 secs****
>
> Min-Sess : 90 secs****
>
> RTP Engine : asterisk****
>
> Parkinglot :****
>
> Use Reason : No****
>
> Encryption : No****
>
> ** **
>
> Asterisk*CLI>****
>
> ** **
>
> ** **
>
> ** **
>
> Em 13 de agosto de 2013 21:46, Fernando - NextBilling IP Solutions <
> fernando em nextbilling.com.br> escreveu:****
>
> No bash do servidor digite:****
>
> ****
>
> [root em serv ~]# curl -dump http://ipecho.net/plain;echo****
>
> ****
>
> Isso vai te mostrar qual o seu IP externo.****
>
> ****
>
> Poste aqui pra o retorno do comando no CLI do Asterisk: *sip show settings
> *****
>
> ****
>
> Pare todas as chamadas do servidor, e digite: *sip set debug on*****
>
> ****
>
> No CLI do Asterisk, digite *sip show peer XXXXX*****
>
> Onde o XXXX você substitui pelo nome do seu ramal.****
>
> ****
>
> Faz o mesmo comando para o nome do trunk que ta terminando a chamada.****
>
> ****
>
> Pegue os dados retornados pelos comandos, mais o siptrace gerado e envia
> pra gente.****
>
> ****
>
> Fica mais fácil ajudar. Senão vc irá postar pedaço por pedaço e ficará
> difícil te dar uma Luz.****
>
> ****
>
> Atenciosamente,****
>
> ****
>
> *Fernando da Silva Santos*****
>
> *CEO* – Chief Executive Officer****
>
> *NextBilling IP Solutions*****
>
> * *****
>
> *SP: *+55 (11) 3522-9200****
>
> *RJ: *+55 (21) 4063-8854****
>
> *Tollfree:* 0800 580-9200****
>
> http://www.nextbilling.com.br****
>
> ****
>
> *De:* asteriskbrasil-bounces em listas.asteriskbrasil.org [mailto:
> asteriskbrasil-bounces em listas.asteriskbrasil.org] *Em nome de *Fernando
> Trilha
> *Enviada em:* terça-feira, 13 de agosto de 2013 21:21
> *Para:* asteriskbrasil em listas.asteriskbrasil.org
> *Assunto:* Re: [AsteriskBrasil] Ligação cai****
>
> ****
>
> Testei de outra rede, agora esta assim, a primeira ligacao caiu com 10
> segundos, a segunda ja caiu com 45 segundos, porem sem erros na console do
> asterisk.****
>
> ****
>
> Em 13 de agosto de 2013 13:40, Fernando Trilha <ftrilha em gmail.com>
> escreveu:****
>
> Entao fiz os testes agora pelo 3g e funcionou, mas de manha pela wirelles
> de um cliente nao.****
>
> ****
>
> Em 13 de agosto de 2013 12:20, Fernando Trilha <ftrilha em gmail.com>
> escreveu:****
>
> ****
>
> no server asterisk não, to esperando o pessoal do firewall me retornar.***
> *
>
> ****
>
> Em 13 de agosto de 2013 11:40, Jefferson B. Limeira <
> jbl em internexxus.com.br> escreveu:****
>
> ****
>
> Esses parecem estar certos, mas são todos da interface interna, ou seja,
> estariam certos mesmo... o problema é na outra interface.****
>
> E o lsmod, retornou alguma coisa?****
>
> Em 2013-08-13 11:15, Fernando Trilha escreveu:****
>
> Este ai é o tcpdump, o do firewall terei de ver com a pessoa que cuida
> dele, ele é baseado em linux.****
>
> ****
>
> ****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 630****
>
> ACK sip:04833421234 em 200.193.70.69:5070 SIP/2.0****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-97b114f393f60c33-1---d8754z-****
>
> Max-Forwards: 70****
>
> Contact: <sip:9962 em 179.252.225.2:5060;transport=UDP>****
>
> To: <sip:04833421234 em 200.193.70.69:5070>;transport=UDP****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 2 ACK****
>
> User-Agent: Zoiper r18976****
>
> Authorization: Digest
> username="9962",realm="asterisk",nonce="42d49556",uri="
> sip:04833421234 em 200.193.70.69:5070;transport=UDP
> ",response="92a9dcb0400142f6745fea5d142bd419",algorithm=MD5****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> 11:13:33.051106 IP (tos 0x0, ttl 64, id 6743, offset 0, flags [none],
> proto UDP (17), length 897)****
>
> 192.168.0.101.5070 > 179.252.225.2.5060: SIP, length: 869****
>
> SIP/2.0 200 OK****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-054d56ba92c5b7a5-1---d8754z-;received=179.252.225.2;rport=5060
> ****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> To: <sip:04833421234 em 200.193.70.69:5070
> >;transport=UDP>;tag=as45a78870****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 2 INVITE****
>
> Server: Asterisk PBX 1.8.23.0****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO, PUBLISH****
>
> Supported: replaces, timer****
>
> Session-Expires: 1800;refresher=uas****
>
> Contact: <sip:04833421234 em 200.193.70.69:5070>****
>
> Content-Type: application/sdp****
>
> Require: timer****
>
> Content-Length: 228****
>
> ****
>
> v=0****
>
> o=root 398579285 398579286 IN IP4 200.193.70.69****
>
> s=Asterisk PBX 1.8.23.0****
>
> c=IN IP4 200.193.70.69****
>
> t=0 0****
>
> m=audio 12514 RTP/AVP 0 8 3****
>
> a=rtpmap:0 PCMU/8000****
>
> a=rtpmap:8 PCMA/8000****
>
> a=rtpmap:3 GSM/8000****
>
> a=ptime:20****
>
> a=sendrecv****
>
> ****
>
> 11:13:33.257404 IP (tos 0x0, ttl 58, id 0, offset 0, flags [DF], proto UDP
> (17), length 658)****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 630****
>
> ACK sip:04833421234 em 200.193.70.69:5070 SIP/2.0****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-97b114f393f60c33-1---d8754z-****
>
> Max-Forwards: 70****
>
> Contact: <sip:9962 em 179.252.225.2:5060;transport=UDP>****
>
> To: <sip:04833421234 em 200.193.70.69:5070>;transport=UDP****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 2 ACK****
>
> User-Agent: Zoiper r18976****
>
> Authorization: Digest
> username="9962",realm="asterisk",nonce="42d49556",uri="
> sip:04833421234 em 200.193.70.69:5070;transport=UDP
> ",response="92a9dcb0400142f6745fea5d142bd419",algorithm=MD5****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> 11:13:36.376701 IP (tos 0x0, ttl 64, id 6744, offset 0, flags [none],
> proto UDP (17), length 897)****
>
> 192.168.0.101.5070 > 179.252.225.2.5060: SIP, length: 869****
>
> SIP/2.0 200 OK****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-054d56ba92c5b7a5-1---d8754z-;received=179.252.225.2;rport=5060
> ****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> To: <sip:04833421234 em 200.193.70.69:5070
> >;transport=UDP>;tag=as45a78870****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 2 INVITE****
>
> Server: Asterisk PBX 1.8.23.0****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO, PUBLISH****
>
> Supported: replaces, timer****
>
> Session-Expires: 1800;refresher=uas****
>
> Contact: <sip:04833421234 em 200.193.70.69:5070>****
>
> Content-Type: application/sdp****
>
> Require: timer****
>
> Content-Length: 228****
>
> ****
>
> v=0****
>
> o=root 398579285 398579286 IN IP4 200.193.70.69****
>
> s=Asterisk PBX 1.8.23.0****
>
> c=IN IP4 200.193.70.69****
>
> t=0 0****
>
> m=audio 12514 RTP/AVP 0 8 3****
>
> a=rtpmap:0 PCMU/8000****
>
> a=rtpmap:8 PCMA/8000****
>
> a=rtpmap:3 GSM/8000****
>
> a=ptime:20****
>
> a=sendrecv****
>
> ****
>
> 11:13:36.432176 IP (tos 0x0, ttl 58, id 0, offset 0, flags [DF], proto UDP
> (17), length 658)****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 630****
>
> ACK sip:04833421234 em 200.193.70.69:5070 SIP/2.0****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-97b114f393f60c33-1---d8754z-****
>
> Max-Forwards: 70****
>
> Contact: <sip:9962 em 179.252.225.2:5060;transport=UDP>****
>
> To: <sip:04833421234 em 200.193.70.69:5070>;transport=UDP****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 2 ACK****
>
> User-Agent: Zoiper r18976****
>
> Authorization: Digest
> username="9962",realm="asterisk",nonce="42d49556",uri="
> sip:04833421234 em 200.193.70.69:5070;transport=UDP
> ",response="92a9dcb0400142f6745fea5d142bd419",algorithm=MD5****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> 11:13:40.377363 IP (tos 0x0, ttl 64, id 6745, offset 0, flags [none],
> proto UDP (17), length 897)****
>
> 192.168.0.101.5070 > 179.252.225.2.5060: SIP, length: 869****
>
> SIP/2.0 200 OK****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-054d56ba92c5b7a5-1---d8754z-;received=179.252.225.2;rport=5060
> ****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> To: <sip:04833421234 em 200.193.70.69:5070
> >;transport=UDP>;tag=as45a78870****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 2 INVITE****
>
> Server: Asterisk PBX 1.8.23.0****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO, PUBLISH****
>
> Supported: replaces, timer****
>
> Session-Expires: 1800;refresher=uas****
>
> Contact: <sip:04833421234 em 200.193.70.69:5070>****
>
> Content-Type: application/sdp****
>
> Require: timer****
>
> Content-Length: 228****
>
> ****
>
> v=0****
>
> o=root 398579285 398579286 IN IP4 200.193.70.69****
>
> s=Asterisk PBX 1.8.23.0****
>
> c=IN IP4 200.193.70.69****
>
> t=0 0****
>
> m=audio 12514 RTP/AVP 0 8 3****
>
> a=rtpmap:0 PCMU/8000****
>
> a=rtpmap:8 PCMA/8000****
>
> a=rtpmap:3 GSM/8000****
>
> a=ptime:20****
>
> a=sendrecv****
>
> ****
>
> 11:13:40.422249 IP (tos 0x0, ttl 58, id 0, offset 0, flags [DF], proto UDP
> (17), length 658)****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 630****
>
> ACK sip:04833421234 em 200.193.70.69:5070 SIP/2.0****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-97b114f393f60c33-1---d8754z-****
>
> Max-Forwards: 70****
>
> Contact: <sip:9962 em 179.252.225.2:5060;transport=UDP>****
>
> To: <sip:04833421234 em 200.193.70.69:5070>;transport=UDP****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 2 ACK****
>
> User-Agent: Zoiper r18976****
>
> Authorization: Digest
> username="9962",realm="asterisk",nonce="42d49556",uri="
> sip:04833421234 em 200.193.70.69:5070;transport=UDP
> ",response="92a9dcb0400142f6745fea5d142bd419",algorithm=MD5****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> 11:13:40.574233 IP (tos 0x0, ttl 58, id 0, offset 0, flags [DF], proto UDP
> (17), length 644)****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 616****
>
> BYE sip:04833421234 em 200.193.70.69:5070 SIP/2.0****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-14cb719dd333c35c-1---d8754z-****
>
> Max-Forwards: 70****
>
> Contact: <sip:9962 em 179.252.225.2:5060;transport=UDP>****
>
> To: <sip:04833421234 em 200.193.70.69:5070>;transport=UDP****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 3 BYE****
>
> User-Agent: Zoiper r18976****
>
> Authorization: Digest
> username="9962",realm="asterisk",nonce="42d49556",uri="
> sip:04833421234 em 200.193.70.69:5070
> ",response="4911bfc1a62265954226f72e9b36d4b1",algorithm=MD5****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> 11:13:41.069084 IP (tos 0x0, ttl 58, id 0, offset 0, flags [DF], proto UDP
> (17), length 32)****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 4****
>
> ****
>
> ****
>
> ****
>
> 11:13:41.081575 IP (tos 0x0, ttl 58, id 0, offset 0, flags [DF], proto UDP
> (17), length 644)****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 616****
>
> BYE sip:04833421234 em 200.193.70.69:5070 SIP/2.0****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-14cb719dd333c35c-1---d8754z-****
>
> Max-Forwards: 70****
>
> Contact: <sip:9962 em 179.252.225.2:5060;transport=UDP>****
>
> To: <sip:04833421234 em 200.193.70.69:5070>;transport=UDP****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 3 BYE****
>
> User-Agent: Zoiper r18976****
>
> Authorization: Digest
> username="9962",realm="asterisk",nonce="42d49556",uri="
> sip:04833421234 em 200.193.70.69:5070
> ",response="4911bfc1a62265954226f72e9b36d4b1",algorithm=MD5****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> 11:13:42.074742 IP (tos 0x0, ttl 58, id 0, offset 0, flags [DF], proto UDP
> (17), length 644)****
>
> 179.252.225.2.5060 > 192.168.0.101.5070: SIP, length: 616****
>
> BYE sip:04833421234 em 200.193.70.69:5070 SIP/2.0****
>
> Via: SIP/2.0/UDP 179.252.225.2:5060
> ;branch=z9hG4bK-d8754z-14cb719dd333c35c-1---d8754z-****
>
> Max-Forwards: 70****
>
> Contact: <sip:9962 em 179.252.225.2:5060;transport=UDP>****
>
> To: <sip:04833421234 em 200.193.70.69:5070>;transport=UDP****
>
> From: <sip:9962 em 200.193.70.69:5070;transport=UDP>;tag=6f423601****
>
> Call-ID: ZTVhMDgwZTNjOTVhODk1ZjkwNGMzZTkyODgyZjk0N2E.****
>
> CSeq: 3 BYE****
>
> User-Agent: Zoiper r18976****
>
> Authorization: Digest
> username="9962",realm="asterisk",nonce="42d49556",uri="
> sip:04833421234 em 200.193.70.69:5070
> ",response="4911bfc1a62265954226f72e9b36d4b1",algorithm=MD5****
>
> Content-Length: 0****
>
> ****
>
> ****
>
> Em 13 de agosto de 2013 11:10, Jefferson B. Limeira <
> jbl em internexxus.com.br> escreveu:****
>
> Firewall/linux ou baseado?****
>
> Mostre o resultado de :****
>
> # lsmod |grep sip****
>
> Para debuggar tcpdump -i INTERFACE_LOCAL -s 0 -nn -v port 5060****
>
> Aqui minha martelada, aquela pode ser única e certeira ou não ajudar em
> absolutamente nada... se fizer na interface interna e na externa, verá que
> os pacotes são diferentes, teu firewall deve estar reescrevendo o pacote
> SIP. Manda o resultado do lsmod e veja se tem os módulos de conntrack e nat
> rodando...****
>
> Em 2013-08-13 11:03, Fernando Trilha escreveu:****
>
> Sim tenho um firewall no meio do caminho, mas liberei as portas pra ele. *
> ***
>
> Como faço para debugar o sip?****
>
> --****
>
> --
> []'s Jefferson B. Limeira****
>
> ** **
>
> jbl em internexxus.com.br****
>
> (41) 9928-8628****
>
>
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
> Intercomunicadores para acesso remoto via rede IP. Conheça em
> www.Khomp.com.
> _______________________________________________
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para
> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org****
>
>
>
> ****
>
> ****
>
> -- ****
>
> Atte.
> Fernando Trilha
> Analista de Suporte ****
>
> 8414 - 6008
> ftrilha em gmail.com****
>
> ::Soluções em informatica e redes corporativas::****
>
> ****
>
> ** **
>
> _______________________________________________****
>
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;****
>
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;****
>
> Intercomunicadores para acesso remoto via rede IP. Conheça em www.Khomp.com.****
>
> _______________________________________________****
>
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.****
>
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.****
>
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.****
>
> _______________________________________________****
>
> Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org****
>
> ****
>
> --
> []'s Jefferson B. Limeira****
>
> jbl em internexxus.com.br****
>
> (41) 9928-8628****
>
>
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
> Intercomunicadores para acesso remoto via rede IP. Conheça em
> www.Khomp.com.
> _______________________________________________
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para
> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org****
>
>
>
> ****
>
> ****
>
> -- ****
>
> Atte.
> Fernando Trilha
> Analista de Suporte ****
>
> 8414 - 6008
> ftrilha em gmail.com****
>
> ::Soluções em informatica e redes corporativas::****
>
>
>
> ****
>
> ****
>
> -- ****
>
> Atte.
> Fernando Trilha
> Analista de Suporte ****
>
> 8414 - 6008
> ftrilha em gmail.com****
>
> ::Soluções em informatica e redes corporativas::****
>
>
>
> ****
>
> ****
>
> -- ****
>
> Atte.
> Fernando Trilha
> Analista de Suporte ****
>
> 8414 - 6008
> ftrilha em gmail.com****
>
> ::Soluções em informatica e redes corporativas::****
>
>
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
> Intercomunicadores para acesso remoto via rede IP. Conheça em
> www.Khomp.com.
> _______________________________________________
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para
> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org****
>
>
>
> ****
>
> ** **
>
> -- ****
>
> Atte.
> Fernando Trilha
> Analista de Suporte ****
>
> 8414 - 6008
> ftrilha em gmail.com****
>
> ::Soluções em informatica e redes corporativas::****
>
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
> Intercomunicadores para acesso remoto via rede IP. Conheça em
> www.Khomp.com.
> _______________________________________________
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para
> asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>
--
Atte.
Fernando Trilha
Analista de Suporte
8414 - 6008
ftrilha em gmail.com
::Soluções em informatica e redes corporativas::
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