search for: promiscredir

I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like "promiscredir=yes" in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo! Mail, o ?nico sem limite de espa?o para armazenamento! http://br.mail.yahoo.com/

I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue.

...pgroup : Mailbox : VM Extension : asterisk LastMsgsSent : -1 Inc. limit : 0 Outg. limit : 0 Dynamic : Yes Callerid : "" <> Expire : 386 Expiry : 900 Insecure : no Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.10.32 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 2608 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : OK (16 ms) Use...

...eened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : "" <> Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 10.4.5.1 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport : TCP Def. Username: 971 SIP Options : (none) Codecs...

...jhZN8lArDgzLI7z8V2fxV type=peer ;context=transit-ivr context=incoming dtmfmode=inband The new section, with many failed experiments commented out, is after the [sipivr] section: [sessionmgr1] type=peer ;type=friend port=5060 host=10.90.0.17 dtmfmode=inband allowguest=yes qualify=yes realm=mcts.org promiscredir=yes ;Some have suggested using canreinvite=no with Avaya- didn't try that yet ;canreinvite=no canreinvite=yes transport=tcp ;context=incoming context=from-internal ;username=10.90.0.103 fromdomain=mcts.org disallow=all allow=ulaw allow=alaw tcpenable=yes tcpbindaddr=0.0.0.0:5060 Nothing I trie...

...25@ASTERISK-IP>' May 24 20:11:25 NOTICE[1112841136]: app_dial.c:232 wait_for_answer: Unable to create local channel for call forward to 'SIP/<PSTN-NUMNER>@<ASTERISK-IP>, sip:'PSTN-NUMBER@SIP-SERVER-DOMAIN' == Everyone is busy/congested at this time I did define promiscredir=yes and also created a section with Asterisk's OWN IP Address in sip.conf [Asterisk-IP] type=friend insecure=very promiscredir=yes 302 redirect back to Asterisk. Before that it rewrites the URI hostname with the IP of Asterisk. 4) Asterisk get the 302 redirect and then tries to dial th...

My asterisk should follow 302 redirect which it receives from other sip server(10.10.10.10). By running network sniffer I see, that asterisk receives 302 answer, but doesn't follow it. My config is: sip.conf: ....... [out4] type=peer host=10.10.10.10 canreinvite=no promiscredir=yes insecure=very disallow=all allow=g729 allow=g723 ....... extensions.conf: [to-sip] exten => _0011X., 1, Dial(SIP/${EXTEN:4}@out4) exten => _0011X., 2, Hangup() Any ideas? Vitaly ____________________________________________________________________________________ Be a bette...

...nore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport (work around more UNIDEN bugs) ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; ; Note that promiscredir when redirects are made to the ; ; local; ; This will pass incoming calls to the 's' extension ; ; ;register => 2345:password@sip_proxy/1234 ; ;...

...: 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 124 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-...

...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 : 201.217.31.10 Addr->IP : 201.217.31.1...

...ays ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport (work around more UNIDEN bugs) ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; ; Note that promiscredir when redirects are made to the ; ; local system will cause loops since SIP is incapable ; ; of performing a "hairpin" call. ; ; If...

...ps >> Expire : 124 >> Insecure : no >> Force rport : Yes >> ACL : No >> DirectMedACL : No >> T.38 support : No >> T.38 EC mode : Unknown >> T.38 MaxDtgrm: 4294967295 >> DirectMedia : No >> PromiscRedir : No >> User=Phone : No >> Video Support: Yes >> Text Support : No >> Ign SDP ver : No >> Trust RPID : No >> Send RPID : Yes >> TrustIDOutbnd: Legacy >> Subscriptions: Yes >> Overlap dial : No >> DTMFm...

...ype=peer ;context=transit-ivr context=incoming dtmfmode=inband The new section, with many failed experiments commented out, is after the [sipivr] section: [sessionmgr1] type=peer ;type=friend port=5060 host=10.90.0.17 dtmfmode=inband allowguest=yes qualify=yes realm=mcts.org<http://mcts.org> promiscredir=yes ;Some have suggested using canreinvite=no with Avaya- didn't try that yet ;canreinvite=no canreinvite=yes transport=tcp ;context=incoming context=from-internal ;username=10.90.0.103 fromdomain=mcts.org<http://mcts.org> disallow=all allow=ulaw allow=alaw tcpenable=yes tcpbindaddr=0.0.0...

...: Yes Callerid : "JDOE" <100> MaxCallBR : 384 kbps Expire : 2680 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID : No TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 1...

...ou to change the user agent string ;nat=no ; NAT settings ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581 support ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => user[:secret[:authuser]]@host[:port][/extension] ; ; If no extension is given, the 's' extens...

...forwards : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 3326 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : x.x.x.x:5060 Defaddr-&...

...tion Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 245 at device VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : "device" <245> MaxCallBR : 384 kbps Expire : 67 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.0.239 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 245 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ul...

...ed: sip.conf (for the DID) [general] context=default recordhistory=yes port=5060 bindaddr=0.0.0.0 srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=120 allow=ulaw allow=alaw musicclass=default language=en relaxdtmf=yes rtptimeout=60 trustrpid = no progressinband=yes useragent=Asterisk PBX promiscredir = no [incoming] ; For incoming calls only. type=user username=xxxxxx secret=xxxxxxxx host=sipgate.co.uk fromuser=xxxxxx fromdomain=sipgate.co.uk authuser=xxxxxxx dtmfmode=info context=from-sip insecure=very disallow=all allow=ulaw allow=alaw iax.conf (for the peers/terminating services) Can past...

Hello, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??

...: Yes Callerid : "0049177xxxxxxx" <> MaxCallBR : 384 kbps Expire : -1 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 3200...