chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks
you probably need to add a correct host entry in your /etc/hosts file for your machine it goes ip name alias 192.168.1.1 asterisk.goober.org asterisk so 192.168.1.1 asterisk asterisk.googber.org is not the same thing. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Shawn Sent: Thursday, April 01, 2004 11:07 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sip problems chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
It looks like you haven't defined a "tina" extension. You have the "tina" SIP account set to be extension "1000". If you want to dial extension "tina" change "1000" to "tina". -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Primoz Kragelj Sent: Monday, May 02, 2005 12:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP problems Hi all, I'm newbie to VOIP/SIP/asterisk... and I having problems with SIP on local network. I have Freebsd server 5.3 running asterisk and two x-lite clients. I added following lines to sip.conf [tina] type=friend host=dynamic dtmfmode=inband context=sip [primozz] type=friend host=dynamic dtmfmode=inband context=sip And following to extensions.conf [sip] exten => 1000,1,Dial,SIP/tina exten => 2000,1,Dial,SIP/primozz *CLI> sip show users Username Secret Accountcode Def.Context ACL NAT primozz sip No RFC35 tina sip No RFC35 I have X-Lite clinet on Win XP and while trying to make call to "tina" I got 404 error - not found. Same for vice versa...Both users are local.>From debug below following line:To: <sip:tina@192.168.1.3>;tag=as1283188b is very strange to me. Instead od 192.168.1.3 there should be 192.168.1.1. Do I need to put some ware static IP for each client ? And following is debug from asterisk: Peer audio RTP is at port 192.168.1.3:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for tina in sip list_route: hop: <sip:primozz@192.168.1.3:5060> Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz <sip:primozz@192.168.1.3>;tag=1716760483 To: <sip:tina@192.168.1.3>;tag=as1283188b Call-ID: B5CEF7FC-640F-42FD-B93E-FFDE1F9EE6F8@192.168.1.3 CSeq: 22324 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:tina@192.168.1.12> Content-Length: 0 to 192.168.1.3:5060 Sip read: ACK sip:tina@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz <sip:primozz@192.168.1.3>;tag=1716760483 To: <sip:tina@192.168.1.3>;tag=as1283188b Contact: <sip:primozz@192.168.1.3:5060> Call-ID: B5CEF7FC-640F-42FD-B93E-FFDE1F9EE6F8@192.168.1.3 CSeq: 22324 ACK Max-Forwards: 70 Content-Length: 0 Thanks for help ! Regards, Primoz _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, I have been trying to configure one Asterisk to use a Sip provider. My sip.conf is: register => user:passwd@www.xxx.yyy.zzz [www.xxx.yyy.zzz] type=friend secret=passwd username=user host=www.xxx.yyy.zzz insecure=very disallow=all allow=g729;gsm;ulaw;alaw reinvite=no [sipphone] ;dtmfmode=info host=dynamic language=es nat=yes secret=mysecret type=friend username=sipphone allow=g729;ilbc;gsm;ulaw;alaw regseconds=0 cancallforward=yes The problem is: The outgoing call doesn't works, SIP responses 403 in my sip phone the sip debug say Sip read: SIP/2.0 403 Insufficient Balance Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;branch=z9hG4bK6d2b3580 Record-Route: <sip:www.xxx.yyy.zzz;ftag=as752586f3;lr> From: 1090 <sip:1090@aaa.bbb.ccc.ddd>;tag=as752586f3 To: <sip:5642216156@www.xxx.yyy.zzz>;tag=65a531031f6d1fcdde9ff201087cff4e Call-ID: 31d3fdd97a098fd87fed1b6d4e83d8dc@aaa.bbb.ccc.ddd CSeq: 102 INVITE Server: Sippy chan_sip.c:6864 handle_response: Forbidden - wrong password on authentication for INVITE I have installed a sip phone direct to provider, and outgoing call works. I will be happy about any suggestions. Thanks in advance! Jorge Verastegui redcetus.com -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-keys Size: 1328 bytes Desc: PGP Public Key Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050721/52f86876/attachment.key
Hi all, I have a problem with a sip phone connecting via a wireless
bridge. Diagram below.
snom360 ----- dd-wrt54_AP_MODE_WDS ---- dd-wrt54_AP_MODE_WDS -- *
10.0.0.4 10.0.0.252 10.0.0.253 10.0.0.50
The SNOM360 registers with the * box fine
sip show peers
2004/2004 10.0.0.4 D 255.255.255.255 5060
OK (46 ms
The SNOM360 can make local and outgoing calls fine
all local and incoming calls to the 360 fail and go straight to voicemail.
Executing SetCIDName("SIP/2002-42dc", "bails") in new stack
-- Executing AGI("SIP/2002-42dc", "dialparties.agi") in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 4
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode -- dialparties.agi: uniqueid =
1137171878.17385
-- dialparties.agi: channel = SIP/2002-42dc
-- dialparties.agi: callerid = bails <2002>
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = SIP
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = 2004
dialparties.agi: Caller ID name is 'bails' number is '2002'
-- dialparties.agi: Added extension 2004 to extension map
-- dialparties.agi: Extension 2004 cf is disabled
-- dialparties.agi: Extension 2004 do not disturb is disabled
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 2004 has call waiting disabled
-- dialparties.agi: DbSet CALLTRACE/2004 to 2002
dialparties.agi: Dial string is SIP/2004|15|tr
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/2002-42dc", "SIP/2004|15|tr") in
new stack
-- Called 2004
-- SIP/2004-b5c4 is circuit-busy
== Everyone is busy/congested at this time
-- Executing GotoIf("SIP/2002-42dc",
"0?s-CONGESTION|1") in new stack
-- Executing GotoIf("SIP/2002-42dc",
"0?s-CONGESTION|1") in new stack
-- Executing NoOp("SIP/2002-42dc", "Sending to Voicemail box
2004@default") in new stack
from the * box I can ping and browse the webui of the snom
Any Ideas???
Thanks in advance
Bails