[AsteriskBrasil] Ligação entre ramais muda
Fernando Trilha
ftrilha em gmail.com
Segunda Outubro 28 15:41:09 BRST 2013
Fernando, vamos la...
servidor*CLI> sip show settings
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
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 1.8.11.0
SDP Session Name: Asterisk PBX 1.8.11.0
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: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
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: Accept
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:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
servidor*CLI> sip show peer 9960
* Name : 9960
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : ramais
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : 1
Pickupgroup : 1
MOH Suggest :
Mailbox : 9960
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "" <9960>
MaxCallBR : 384 kbps
Expire : 2311
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 : 200.193.70.93:2683
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 9960
SIP Options : (none)
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (ulaw:20,alaw:20,gsm:20)
Auto-Framing : No
Status : OK (201 ms)
Useragent : X-Lite release 1011s stamp 41150
Reg. Contact : sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
servidor*CLI> sip show peer 9961
* Name : 9961
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : ramais
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : 1
Pickupgroup : 1
MOH Suggest :
Mailbox : 9961
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "" <9961>
MaxCallBR : 384 kbps
Expire : 3065
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 : 200.193.70.93:24477
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 9961
SIP Options : (none)
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (ulaw:20,alaw:20,gsm:20)
Auto-Framing : No
Status : OK (179 ms)
Useragent : X-Lite release 1011s stamp 41150
Reg. Contact : sip:9961 em 200.193.70.93:24477;rinstance=0f6639093db4a1d7
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
servidor*CLI> sip set debug peer 9960
SIP Debugging Enabled for IP: 200.193.70.93
Reliably Transmitting (NAT) to 200.193.70.93:2683:
OPTIONS sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK6d96589d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 192.168.1.5>;tag=as1db12108
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
Contact: <sip:asterisk em 192.168.1.5:5060>
Call-ID: 49863d0e131380840c553e5d7854a8a2 em 192.168.1.5:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Mon, 28 Oct 2013 17:37:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:200.193.70.93:2683 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060
;branch=z9hG4bK6d96589d;rport=5060;received=189.114.206.85
Contact: <sip:10.0.0.100:14069>
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=b1594d26
From: "asterisk"<sip:asterisk em 192.168.1.5>;tag=as1db12108
Call-ID: 49863d0e131380840c553e5d7854a8a2 em 192.168.1.5:5060
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 '
49863d0e131380840c553e5d7854a8a2 em 192.168.1.5:5060' Method: OPTIONS
<--- SIP read from UDP:200.193.70.93:2683 --->
<------------->
== Using SIP RTP CoS mark 5
-- Executing [9960 em ramais:1] Dial("SIP/9961-00000008", "sip/9960") in
new stack
== Using SIP RTP CoS mark 5
Audio is at 13418
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 200.193.70.93:2683:
INVITE sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4b9378e4;rport
Max-Forwards: 70
From: "9961" <sip:9961 em 192.168.1.5>;tag=as1be09e45
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
Contact: <sip:9961 em 192.168.1.5:5060>
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Mon, 28 Oct 2013 17:37:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 2034619578 2034619578 IN IP4 192.168.1.5
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.1.5
t=0 0
m=audio 13418 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called sip/9960
Retransmitting #1 (NAT) to 200.193.70.93:2683:
INVITE sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK4b9378e4;rport
Max-Forwards: 70
From: "9961" <sip:9961 em 192.168.1.5>;tag=as1be09e45
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
Contact: <sip:9961 em 192.168.1.5:5060>
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Mon, 28 Oct 2013 17:37:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 2034619578 2034619578 IN IP4 192.168.1.5
s=Asterisk PBX 1.8.11.0
c=IN IP4 192.168.1.5
t=0 0
m=audio 13418 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:200.193.70.93:2683 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060
;branch=z9hG4bK4b9378e4;rport=5060;received=189.114.206.85
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
From: "9961" <sip:9961 em 192.168.1.5>;tag=as1be09e45
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:200.193.70.93:2683 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.5:5060
;branch=z9hG4bK4b9378e4;rport=5060;received=189.114.206.85
Contact: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19
From: "9961"<sip:9961 em 192.168.1.5>;tag=as1be09e45
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
-- SIP/9960-00000009 is ringing
<--- SIP read from UDP:200.193.70.93:2683 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060
;branch=z9hG4bK4b9378e4;rport=5060;received=189.114.206.85
Contact: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19
From: "9961"<sip:9961 em 192.168.1.5>;tag=as1be09e45
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 180
v=0
o=- 4 2 IN IP4 10.0.0.100
s=CounterPath X-Lite 3.0
c=IN IP4 10.0.0.100
t=0 0
m=audio 6270 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.0.100:6270
list_route: hop: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
set_destination: Parsing
<sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
for address/port to send to
set_destination: set destination to 200.193.70.93:2683
Transmitting (NAT) to 200.193.70.93:2683:
ACK sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK3dcc2c0e;rport
Max-Forwards: 70
From: "9961" <sip:9961 em 192.168.1.5>;tag=as1be09e45
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19
Contact: <sip:9961 em 192.168.1.5:5060>
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11.0
Content-Length: 0
---
-- SIP/9960-00000009 answered SIP/9961-00000008
-- Locally bridging SIP/9961-00000008 and SIP/9960-00000009
[Oct 28 15:37:57] WARNING[2072]: chan_sip.c:3641 retrans_pkt:
Retransmission timeout reached on transmission
ZmE1NmM4MDM5YjY1MzIxMDVkNTA4NWJkMTA0YTBlM2M. for seqno 2 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 11455ms with no response
[Oct 28 15:37:57] WARNING[2072]: chan_sip.c:3670 retrans_pkt: Hanging up
call ZmE1NmM4MDM5YjY1MzIxMDVkNTA4NWJkMTA0YTBlM2M. - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '
362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060' in 32320 ms (Method:
INVITE)
set_destination: Parsing
<sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
for address/port to send to
set_destination: set destination to 200.193.70.93:2683
Reliably Transmitting (NAT) to 200.193.70.93:2683:
BYE sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK3ecf897d;rport
Max-Forwards: 70
From: "9961" <sip:9961 em 192.168.1.5>;tag=as1be09e45
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.11.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (ramais, 9960, 1) exited non-zero on
'SIP/9961-00000008'
<--- SIP read from UDP:200.193.70.93:2683 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060
;branch=z9hG4bK3ecf897d;rport=5060;received=189.114.206.85
Contact: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>
To: <sip:9960 em 200.193.70.93:2683;rinstance=6a65d6fc3bcc8898>;tag=4e3cab19
From: "9961"<sip:9961 em 192.168.1.5>;tag=as1be09e45
Call-ID: 362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060
CSeq: 103 BYE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '
362787273f6e6d56131c13e523f5ea68 em 192.168.1.5:5060' Method: INVITE
servidor*CLI>
Em 28 de outubro de 2013 15:31, Fernando - CIO - NextBilling IP Solutions <
fernando em nextbilling.com.br> escreveu:
> no CLI do asterisk: *sip show settings*
> no CLI do asterisk: *sip show peer 9960*
> no CLI do asterisk: *sip set debug peer 9960*
>
> Faz a ligação e depois posta toda a saida aqui na lista. fica mais facil
> te ajudar.
>
> Em 28-10-2013 15:25, Fernando Trilha escreveu:
>
> Hudson, esta assim:
>
> [9960]
> type=friend
> secret=XXXXX
> host=dynamic
> mailbox=9960
> context=ramais
> callerid=9960
> directmedia=no
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> qualify=yes
> callgroup=1
> pickupgroup=1
> canreinvite=no
> nat=yes
> externrefresh=10
> externhost=189.114.206.85
> localnet=192.168.1.0/255.255.255.0
>
>
>
> Em 28 de outubro de 2013 15:23, Hudson Cardoso <hudsoncardoso em hotmail.com
> > escreveu:
>
>> Coloca
>> nat=yes no teu sip conf.
>>
>>
>> Hudson
>> (048) 8413-7000
>> Para quem nao cre, nenhuma prova converte,Para aquele que cre, nenhuma prova precisa.
>>
>>
>>
>> ------------------------------
>> Date: Mon, 28 Oct 2013 12:38:55 -0200
>>
>> From: ftrilha em gmail.com
>> To: asteriskbrasil em listas.asteriskbrasil.org
>> Subject: Re: [AsteriskBrasil] Ligação entre ramais muda
>>
>> Agora deu este erro:
>>
>> == Using SIP RTP CoS mark 5
>> -- Called SIP/9961
>> -- SIP/9961-00000004 is ringing
>> -- SIP/9961-00000004 answered SIP/9960-00000003
>> -- Locally bridging SIP/9960-00000003 and SIP/9961-00000004
>> == Spawn extension (ramais, 9961, 1) exited non-zero on
>> 'SIP/9960-00000003'
>> [Oct 28 12:35:30] WARNING[1629]: chan_sip.c:3641 retrans_pkt:
>> Retransmission timeout reached on transmission
>> ZWRlMDgyYmUzMWUyNzg1M2I5NzJjNWM4ZWJhOTRhNTk. for seqno 2 (Critical
>> Response) -- See
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>> Packet timed out after 12928ms with no response
>>
>>
>>
>> Em 28 de outubro de 2013 12:05, Hudson Cardoso <hudsoncardoso em hotmail.com
>> > escreveu:
>>
>> Testa com iax, se funcionar, é rtp com problemas.
>>
>>
>> Hudson
>>
>>
>> (048) 8413-7000
>> Para quem nao cre, nenhuma prova converte,Para aquele que cre, nenhuma prova precisa.
>>
>>
>>
>> ------------------------------
>> Date: Mon, 28 Oct 2013 12:03:13 -0200
>> From: ftrilha em gmail.com
>> To: asteriskbrasil em listas.asteriskbrasil.org
>> Subject: Re: [AsteriskBrasil] Ligação entre ramais muda
>>
>>
>> Marcelo, fiz a alteração mas continua a mesma coisa.
>>
>>
>> Em 26 de outubro de 2013 09:57, Marcelo Terres <mhterres em gmail.com>escreveu:
>>
>> Seta o directmedia=no para os dois ramais no sip.conf, registra eles
>> novamente e testa para ver se muda algo.
>>
>> []s
>> Marcelo H. Terres
>> mhterres em gmail.com
>> http://mundoopensource.blogspot.com
>> http://biertasters.blogspot.com
>> http://twitter.com/mhterres
>>
>>
>> Em 26 de outubro de 2013 09:35, Fernando Trilha <ftrilha em gmail.com>
>> escreveu:
>> > Sim, os mesmo, gsm, allaw e ullaw.
>> >
>> >
>> > Em 26 de outubro de 2013 09:28, Marcelo Terres <mhterres em gmail.com>
>> > escreveu:
>> >
>> >> E os codecs, são os mesmos ?
>> >>
>> >> Quais codecs tu configurou no sip.conf e no zoiper?
>> >>
>> >> []s
>> >> Marcelo H. Terres
>> >> mhterres em gmail.com
>> >> http://mundoopensource.blogspot.com
>> >> http://biertasters.blogspot.com
>> >> http://twitter.com/mhterres
>> >>
>> >>
>> >> Em 25 de outubro de 2013 20:18, Fernando Trilha <ftrilha em gmail.com>
>> >> escreveu:
>> >> > Pessoal tenho um asterisk instalado, apenas para ligações entre
>> ramais,
>> >> > toca, atende mas mudo os dois lados.
>> >> > Uso o zoiper configurado em dois smartphones com protocolo SIP, nao
>> da
>> >> > erro
>> >> > no CLI.
>> >> >
>> >> > --
>> >> > Atte.
>> >> > Fernando Trilha
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > 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
>> >>
>> >> _______________________________________________
>> >> 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
>>
>> _______________________________________________
>> 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.
>> Conhe�a 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
>>
>> _______________________________________________
>> 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.
>> Conhe�a 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
>>
>> _______________________________________________
>> 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
>
>
>
> _______________________________________________
> 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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