Re: [asterisk-users] sip error logging
On Sat, Apr 16, 2011 at 6:05 PM, Jeremy Kister asterisk...@jeremykister.com wrote: bumping once before sending it to the tracker. Original Message Subject: [asterisk-users] sip error logging Date: Fri, 15 Apr 2011 03:39:23 -0400 I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? * i can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ This may sound like a stupid question, but what are your verbosity and debug levels set at currently? Sherwood McGowan -- _ -- 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] sip error logging
On 4/17/2011 3:16 AM, Sherwood McGowan wrote: This may sound like a stupid question, but what are your verbosity and debug levels set at currently? nope, thats exactly the type of thing i'm wondering if i'm missing :) but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried with verbose 10/debug 10 before posting. no dice. -- Jeremy Kister http://jeremy.kister.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
Re: [asterisk-users] sip error logging
On Sun, Apr 17, 2011 at 2:24 AM, Jeremy Kister asterisk...@jeremykister.com wrote: On 4/17/2011 3:16 AM, Sherwood McGowan wrote: This may sound like a stupid question, but what are your verbosity and debug levels set at currently? nope, thats exactly the type of thing i'm wondering if i'm missing :) but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried with verbose 10/debug 10 before posting. no dice. Ah right on mate! Glad to see that you checked it *and* didn't mind being asked (after all, we're all IT/VOIP professionals, and we all know the first thing to ask is the simplest possible solution ;-] ) Cheers! -- _ -- 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] sip error logging
bumping once before sending it to the tracker. Original Message Subject: [asterisk-users] sip error logging Date: Fri, 15 Apr 2011 03:39:23 -0400 I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? * i can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.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
Re: [asterisk-users] sip error logging
I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com -- Sent from my BlackBerry® Senior Support Engineer US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -- _ -- 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] sip error logging
I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? -- Jeremy Kister http://jeremy.kister.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
Re: [asterisk-users] sip error logging
On 4/15/2011 3:39 AM, Jeremy Kister wrote: I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. forgot to mention that I can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.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
[asterisk-users] SIP Error
*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b7910cc0, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b7910cc0, SIP/Sama203/119545090201||tTor) in new stack -- Called Sama203/119545090201 Sep 8 14:19:09 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as09c56cf2' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b7910cc0, ) in new stack == Spawn extension (default, 800119545090201, 3) exited non-zero on 'SIP/cc101-b7910cc0' -- Executing DeadAGI(SIP/cc101-b7910cc0, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b79017c8, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b79017c8, SIP/Sama203/19545090201||tTor) in new stack -- Called Sama203/19545090201 Sep 8 14:19:53 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as168401db' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b79017c8, ) in new stack == Spawn extension (default, 80019545090201, 3) exited non-zero on 'SIP/cc101-b79017c8' -- Executing DeadAGI(SIP/cc101-b79017c8, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 My sip settings are : [Sama203] type=peer username= fromuser= authuser= secret=x host=203.xxx.xxx.56 fromdomain=203.xxx.xxx.56 nat=no canreinvite=yes insecure=very disallow=all allow=g729 context=default dtmfmode=rfc2833 It happens when I add 2 SIP in single asterisk server. 1.2.30.2 If I remove one, I dont get this error. Anyway to find out , what password asterisk recieves when I use Sama203 ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Error
I have 2 sips configured : 1) register =sama:xx...@209.51.191.xxx:5060 2) register =sama:xx...@209.51.192.xxx:5060 Both are active. 5060 port will be same or different ? On Wed, Sep 9, 2009 at 12:29 AM, David @ULC ucoms2...@gmail.com wrote: *I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b7910cc0, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b7910cc0, SIP/Sama203/119545090201||tTor) in new stack -- Called Sama203/119545090201 Sep 8 14:19:09 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as09c56cf2' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b7910cc0, ) in new stack == Spawn extension (default, 800119545090201, 3) exited non-zero on 'SIP/cc101-b7910cc0' -- Executing DeadAGI(SIP/cc101-b7910cc0, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b79017c8, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b79017c8, SIP/Sama203/19545090201||tTor) in new stack -- Called Sama203/19545090201 Sep 8 14:19:53 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as168401db' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b79017c8, ) in new stack == Spawn extension (default, 80019545090201, 3) exited non-zero on 'SIP/cc101-b79017c8' -- Executing DeadAGI(SIP/cc101-b79017c8, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 My sip settings are : [Sama203] type=peer username= fromuser= authuser= secret=x host=203.xxx.xxx.56 fromdomain=203.xxx.xxx.56 nat=no canreinvite=yes insecure=very disallow=all allow=g729 context=default dtmfmode=rfc2833 It happens when I add 2 SIP in single asterisk server. 1.2.30.2 If I remove one, I dont get this error. Anyway to find out , what password asterisk recieves when I use Sama203 ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP error message
As of today, during startup I get lots of the following: ERROR[2704] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data Does anyone know what it means? This is with Asterisk 1.6.0.9. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip Error
Someone could help me with this error :: Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:47 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Jan 11 14:43:55 ERROR[3371] chan_sip.c: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 10.85.1.8, but there is no hint for that extension Thanks in advance Hernany -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.9/622 - Release Date: 10/1/2007 14:52 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Error message
Does anyone know what causes the following error message means. Aug 29 10:11:08 WARNING[30913]: chan_sip.c:2561 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 256/256) I've not yet tracked down what is causing this, but I get a lot of them at the same time. It may be related to a nokia E60 trying to pick up the call. (It's hard to tell atm.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip error messages
Please can anyone advise what these messages mean? Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/213.xxx.5.xxx-0816e1b8! Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: ACK Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Asterisk 1.2.9.1 and most importantly whether I should worry about them Cheers, Neil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Error 401 Problem
Dear All, I am having this strange problem on my Asterisk 1.2.1. We have a web dialer that can register to the Asterisk box in Hong Kong, but another user using the same account can't register to the Asterisk box using the same web dialer. Below is an output of the sip debug logs. It seems that the digest is missing the username and password, but why? I have also have this call flow for the an IP Phone, but after a while, it will register to the Asterisk. One thing I don't understand is that I have registered successfully in Hong Kong and when the user tries in South Africa, it doesn't work. Please Help! SIP Logs: From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERContact: *User-Agent: VaxSIP UserAgent/1.0Expires: 0Max-Forwards: 70Content-Length: 0 --- (11 headers 0 lines)---Using latest REGISTER request as basis requestSending to 192.168.0.3 : 2232 (non-NAT)Transmitting (NAT) to 196.38.228.123:5060:SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.0.3:2232;received=196.38.228.123From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Transmitting (NAT) to 196.38.228.123:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.0.3:2232;received= 196.38.228.123From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED];tag=as63889026 Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=4929aec7Content-Length: 0 Regards, Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Error Message, pbx.c: 1938
I get these warnings when I reload my config through the console:
Nov 5 04:31:10 WARNING[1301281712]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-5364' sent into invalid extension '321235689' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1309670320]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-cf25' sent into invalid extension '3213084999' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1266076592]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-6b54' sent into invalid extension '13215435249' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1274465200]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-ec56' sent into invalid extension '13215435249' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1318058928]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-2357' sent into invalid extension '3213084999' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1232481200]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-1a6c' sent into invalid extension '13215435249' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1282853808]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-9a92' sent into invalid extension '13215435249' in context
'default', but no invalid handler
Nov 5 04:31:10 WARNING[1249299376]: pbx.c:1938 ast_pbx_run: Channel
'SIP/6822170331-b411' sent into invalid extension '13215435249' in context
'default', but no invalid handler
Destroying call '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'
in my extension.conf, I have one extension that passes all the digits to my
softswitch:
[default]
exten = _.,1,Dial(Zap/15/${EXTEN})
We are testing so I am forcing all calls down channel 15 on my PRI.
sip.conf
[general]
context=default
disallow=all; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=alaw
allow=gsm
allow=g723.1
allow=g729
[2000]
type=friend
username=scott
host=dynamic
context=default
nat=yes
[2001]
type=friend
username=steve
host=dynamic
context=default
nat=yes
[6822170331]
type=friend
username=brian
host=dynamic
context=default
nat=yes
dtmfmode=rfc2833
callerid=3213084999
Should I be concerned? Thanks -
--
--
Heritage Communications Corporation
Melbourne, FL USA 32935
http://www.hcc.net
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[Asterisk-Users] SIP Error Handling :: Pre-Recorded Voice Message?
I have my Asterisk PBX connected to both Free World Dialup and IAXtel. What I am hoping is possible is a way of having the PBX say something other than I am sorry, that is an invalid extension, please try again. when a call is not successfully completed. For example if I dial a number and it returns a All-circuts busy error I would love the PBX to come on and say something like All circuts are busy, please try again. and the same for any other error codes.. I was wondering if this was possible.. I am using the current cvs of Asterisk and zaptel for the ztdummy timing. No hardware cards exist in my machine. Thanks. -- Stephen Rosebush, [EMAIL PROTECTED] http://www.desynched.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP error 407 - can't make outgoing calls
I am using a IPDialog siptone II. I can take incoming calls, but when I try and make an outgoing call I get a SIP 407 error. Can some kind soul explain to me what I am doing wrong? Here's what I found in the wiki: If a proxy does not accept the credentials sent with a request, it SHOULD return a 407 (Proxy Authentication Required). The response MUST include a Proxy-Authenticate header field containing a (possibly new) challenge applicable to the proxy for the requested resource. Here's what I have in sip.conf [514] type=friend ; This device takes and makes calls username=514 secret=password context=inside callerid=Paul Mahler 4154424024 qualify=1000 host=dynamic ; This host is not on the same IP addr every time canreinvite=no [EMAIL PROTECTED] ; Activate the message waiting light for waiting messages ;defaultip=192.168.0.102 Here's the sip debug showing the error: to 209.234.100.68:5060 Retransmitting #5 (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.234.100.68:5060 From: 514 sip:[EMAIL PROTECTED];tag=3397-f0f0c367 To: 503 sip:[EMAIL PROTECTED];tag=as0528d61b Call-ID: [EMAIL PROTECTED] CSeq: 14057 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=230958ab Content-Length: 0 The password at the phone is the same as the password in sip.conf. Thanks! Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Error: Network Unreachable
Hi, I have my asterisk running directly plugged on an static public IP accessible by Internet. I try to contact an Cisco VoIP system on another static public IP address but asterisk return me 403 - Forbidden. When I debug sip flow, it tells me : Network Unreachable. I'ts certainly an error in my sip.conf but I can't find where it is... I'm sure about IP addresses. SIP.conf [general] port = 5060 bindaddr = 213.177.xxx. ;My Pubic Address. context = incoming_SIP disallow=all allow=alaw allow=gsm allow=ulaw #include sip_additional.conf Extensions.conf exten = 4700,1,Dial(SIP/[EMAIL PROTECTED]|30|m) Adress to Contact... Please help Regards, Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP error
Means RFC3389 support is incomplete. Neither Mark or developers @ digiumwill not accept it when it gets completed by anyone. The best way is to turn off client if possible. :-) Please change the settings on your client todisable VAD settings. That will remove that Notice. Kannaiyan - Original Message - From: Deepakumar JV To: [EMAIL PROTECTED] Sent: Thursday, January 29, 2004 2:10 AM Subject: [Asterisk-Users] SIP error Hello When ever i make calls via a SIP provider I keep getting this error message Jan 29 02:09:20 NOTICE[1228887360]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible any idea what is it? Regards Deepak
[Asterisk-Users] SIP error
Hello When ever i make calls via a SIP provider I keep getting this error message Jan 29 02:09:20 NOTICE[1228887360]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible any idea what is it? Regards Deepak
[Asterisk-Users] SIP error: Asked to transmit frame type 64
Hi there,
I'll need some help with this: Trying to establish an IAX2 link between
two servers works in one direction (SIP client with ulaw), but not in the
other (SIP client with GSM). The client used for this is X-Lite behind
NAT while both servers have a public IP (I playback an anouncement before
trying to connect to the second *).
Error on the originating * server:
WARNING[27670]: File chan_sip.c, Line 1148 (sip_write): Asked to transmit
frame type 64, while native formats is 2 (read/write = 2/2)
I really _really_ have no clue why codec 16 bit Signed Linear PCM is n
the game here, to my knowledge that is not supported by X-Lite, and it is
certainly not enabled anyware in the conf files either.
Should I file a bug report, or is this a setup problem on my side?
Philipp
In both sip.conf and iax.conf on both servers I have (with slight
variations):
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
We dial 98616 here:
exten = _9,1,Playback(transfer)
exten = _9,2,Ringing
exten = _9,3,Wait(1)
exten = _9,4,Dial(IAX2/myserv:[EMAIL PROTECTED]/${EXTEN:1})
exten = _9,5,Congestion
exten = _9,105,Playback(tt-monkeysintro)
exten = _9,106,Hangup
my chan_sip.c:
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast-pvt-pvt;
int res = 0;
if (frame-frametype == AST_FRAME_VOICE) {
if (!(frame-subclass ast-nativeformats)) {
-- -- ast_log(LOG_WARNING, Asked to transmit frame type %d, while
native formats is %d (read/write = %d/%d)\n,
frame-subclass, ast-nativeformats, ast-readformat,
ast-
writeformat);
return -1;
}
Related error reports I found:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg12648.html
http://www.mail-archive.com/[EMAIL PROTECTED]/msg05602.html
http://www.mail-archive.com/[EMAIL PROTECTED]/msg03242.html
http://www.mail-archive.com/[EMAIL PROTECTED]/msg01139.html
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Re: [Asterisk-Users] SIP error messages
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 18 September 2003 18:12, marrandy wrote: I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' What's this all about ? Pretty straight forward. A SIP phone at '192.168.1.70' failed registration at your Asterisk box at '192.168.1.1'. Try sip debug at your CLI, and you'll see similar messages as the ones I described in my SIP registration thread. (I still can't make the damned thing work) - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/asxv2TEAILET3McRAovYAJ9GAFOo1ANJekQwhUgIYEhZMaJKtwCgk3os vCeIOKqfjV9XmPzjWL4gfFY= =7Y3l -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP error messages
Hello. I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70' What's this all about ? Regards...Martin -- Osborn's Law: Variables won't; constants aren't. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
