Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous
Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com -- _ -- 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] Failed to authenticate on INVITE to Anonymous
Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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] Failed to authenticate on INVITE to Anonymous
Hi, I am using asterisk ver 1.8.8.1. My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register = test02:test02@192.168.1.55 [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;realm=test02@192.168.1.55 context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote: Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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 -- _ -- 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] Failed to authenticate on INVITE to Anonymous
Hi checked your debug like. Did you check that your SIP device ir registered with server ? if yes then dial below command from CLI *originate sip/test02 application dial* On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hi, I am using asterisk ver 1.8.8.1. My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register = test02:test02@192.168.1.55 [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;realm=test02@192.168.1.55 context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote: Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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 -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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] Failed to authenticate on INVITE to Anonymous
Hi virendra, Dialed same command.. I got below output ast18*CLI originate sip/test02 application dial == Using SIP RTP CoS mark 5 [Jan 4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid:192;tag=as417a5527' Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 4:35 PM, virendra bhati virbh...@gmail.com wrote: Hi checked your debug like. Did you check that your SIP device ir registered with server ? if yes then dial below command from CLI *originate sip/test02 application dial* On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Hi, I am using asterisk ver 1.8.8.1. My SIP trunk conf details are below.. [general] context=default ; Default context for incoming calls realm=192.168.1.55 allowguest=yes realmauth=yes send_rpid=pai register = test02:test02@192.168.1.55 [test02] type=peer nat=no canreinvite=no host=192.168.1.55 ;realm=test02@192.168.1.55 context=incoming secret=test02 permit=192.168.1.0/255.255.255.0 username=test02 fromuser=test02 fromdomain=192.168.1.55 defaultuser=test02 insecure=invite,port outboundproxy=192.168.1.55 promiscredir=yes userphone=yes For more details you can find my paste in pastebin.. Links given below. While Dialing call fro Xlite send following Sip header F= sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.comwrote: Hi, Give the complete details about the asterisk version, and SIP trunk conf details On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com wrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid ;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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 -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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 -- _ -- 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] Failed to authenticate on INVITE to Anonymous
On 1/4/2012 4:37 AM, Jayesh Labade wrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com wrote: Hello Experts, I have pasted my issue in http://pastebin.com/zBGVmdcY I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid;tag=as57d3a806' i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. please find sip.conf file in http://pastebin.com/zBGVmdcY I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com -- _ -- 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 Try: register = test02:test02@192.168.1.55/s sean -- _ -- 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] Failed to authenticate on INVITE
I use realtime. Both information and extensions are stored in DB. It is just a simple setting of the user with dial plan Dial([EMAIL PROTECTED]). exten = 9003,1,Dial([EMAIL PROTECTED]) What I found is the following. 9002 --- S1 --- S2 9002 can make request to S1 and S1 forward the request to S2. 9002 --- S1 --- S2 S2 returns the mentioned error message to S1. (What I guess is 9002 only registers in S1 not in S2, so mentioned error message issued by S2). It is what I got from the above case. Do you have such configuration? I have no idea to solve the problem On 4/20/07, dave cantera [EMAIL PROTECTED] wrote: ango, can you provide some sip.conf and extens.conf info? daveC Rilawich Ango wrote: hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa. Both processes failed to make call with the following error. Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite: Failed to authenticate on INVITE to '9002 sip:[EMAIL PROTECTED];tag=as2ff0c493' Any solution to let them call each others? ango ___ --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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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 ___ --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
Re: [asterisk-users] Failed to authenticate on INVITE
ango,
I have been playing with connecting two * servers... I had to stop but
I do think I had it working... even with this link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers
it wasn't as straight forward as I would have liked... I used a
register on one box and a conf entry on the other. then I reversed the
config for the other * box
pbx82 = 10.10.15.82
pbx15 = 10.10.15.15
on pbx15
sip.conf
register = sip_pbx15:[EMAIL PROTECTED]
[sip_to_pbx82]
type=user
username=sip_pbx15
accountcode=sip_from_pbx15
secret=1234
context=sip_from_pbx15
host=10.10.15.82
disallow=all
allow=ulaw
allow=alaw
allow=gsm
extensions.conf
[sip_pbx15_to_pbx82]
; dial a pbx82 extension via SIP with 982XXX where XXX is the extension
exten =
_982XXX,1,Dial(SIP/sip_pbx15:[EMAIL PROTECTED]/${EXTEN:3},20,r)
;exten = _982XXX,1,Dial(SIP/${EXTEN:3},20,r)
exten = _982XXX,n,Playback(connection-failed)
exten = _982XXX,n,Playback(vm-goodbye)
exten = _982XXX,n,Congestion
exten = _982XXX,n,Hangup
on pbx82
extensions.conf
[sip_from_pbx15]
exten = _XXX,1,Wait(1)
exten = _XXX,n,Answer()
exten = _XXX,n,Dial(SIP/${EXTEN},20,,r)
exten = _XXX,n,VoiceMailMain
exten = _XXX,n,Hangup()
[sip_from_pbx15] must be accessible in your inbound or default
context...
I don't think I made any general section changes...
it has been a few weeks since I played with it and I went only one
way... but if it worked one way it should work the other way too by
reverse duplicating the above config on pbx82 and pbx15 respectively.
let me know how you make out...
daveC
Rilawich Ango wrote:
I use realtime. Both information and extensions are
stored in DB. It
is just a simple setting of the user with dial plan "Dial([EMAIL PROTECTED])".
exten = 9003,1,Dial([EMAIL PROTECTED])
What I found is the following.
9002 --- S1 --- S2
9002 can make request to S1 and S1 forward the request to S2.
9002 --- S1 --- S2
S2 returns the mentioned error message to S1. (What I guess is 9002
only registers in S1 not in S2, so mentioned error message issued by
S2).
It is what I got from the above case. Do you have such configuration?
I have no idea to solve the problem
On 4/20/07, dave cantera [EMAIL PROTECTED] wrote:
ango,
can you provide some sip.conf and extens.conf info?
daveC
Rilawich Ango wrote:
hi,
I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both users to both servers, i.e.
asterisk server 1 (S1) has users 9002,9003
asterisk server 2 (S2) has users 9002,9003
When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa. Both
processes
failed to make call with the following error.
Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802
handle_response_invite:
Failed to authenticate on INVITE to '"9002"
sip:[EMAIL PROTECTED];tag=as2ff0c493'
Any solution to let them call each others?
ango
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[asterisk-users] Failed to authenticate on INVITE
hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa. Both processes failed to make call with the following error. Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite: Failed to authenticate on INVITE to '9002 sip:[EMAIL PROTECTED];tag=as2ff0c493' Any solution to let them call each others? ango ___ --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
Re: [asterisk-users] Failed to authenticate on INVITE
ango, can you provide some sip.conf and extens.conf info? daveC Rilawich Ango wrote: hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa. Both processes failed to make call with the following error. Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite: Failed to authenticate on INVITE to '9002 sip:[EMAIL PROTECTED];tag=as2ff0c493' Any solution to let them call each others? ango ___ --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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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] failed to authenticate on invite
I have 2 asterisk boxes connected via SIP box 1 sip peer connected to box 2 (ip addresses intentionally removed) [ast20] type=friend host=x.x.x.20 insecure=very context=subscriber dtmfmode=inband qualify=no canreinvite=no disallow=all allow=ulaw box 2 sip peer connected to box 1 [sbb19] type=friend host=64.1.8.19 insecure=very context=inbound dtmfmode=inband qualify=yes canreinvite=no disallow=all allow=ulaw I then have 2 UAs registed on box 1, both have identical configs with the exception of username, but one is a Polycom IP501 and the other is a Linksys PAP2 The IP 501 can call to box 2 with no issues, also calls originated on a PRI connected to box 1 connect to box 2 with no issues. The Linksys UA can not call box 2, here is the error (numbers intentionally removed); -- Executing dial(SIP/##0850-b6669f58, SIP/[EMAIL PROTECTED]) -- Called [EMAIL PROTECTED] Nov 7 07:20:45 NOTICE[21059]: chan_sip.c:9709 handle_response_invite: Failed to authenticate on INVITE to 'name removed sip:[EMAIL PROTECTED];tag=as38826922' -- SIP/ast20-09c8b110 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) I have looked at sip debugs from both scenarios, and the invites from box 1 to box 2 look nearly identical, box 2 never shows the call when it fails. I am assuming that there is something that needs to be changed on the ATA or peer config to get it to be able to call via box1 to box2 without requiring authentication, but can not figure out what. Any ideas? ___ --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] Failed to authenticate on INVITE to '601 ...
I have installed the first time Asterisk, (forgive me simple questions)
I have also installed the demo.
After testing demo (call 1000, call 600, ...) I changed in the
extensions.conf:
; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
exten = 5552220,1,Dial(SIP/601,20,r)
exten = 4782156,1,Dial(SIP/602,20,r)
[sipgate.de]
exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten = _0049X.,2,Playback(invalid)
exten = _0049X.,3,Hangup
[sipgate.co.uk]
exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _0044X.,2,Playback(invalid)
exten = _0044X.,3,Hangup
in sip.conf I have:
register = 5552220:[EMAIL PROTECTED]/5552220
register = 4782156:[EMAIL PROTECTED]/4782156
[601]
type=friend
username=601
secret=pwd-601
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=601
nat=yes
caller-id=Ronald 1 601
[602]
type=friend
username=602
secret=pwd-602
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=602
nat=yes
caller-id=Ronald 2 602
[sipgate.de]
type=friend
username=5552220
secret=pwd-de
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=pwd-uk
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.co.uk
nat=yes
context=incomingsipgate
canreinvite=no
The console shows when I want to dial at sipgate.de the number 1
(test) or 5 (Voicemail): 00491
-- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new stack
-- Called [EMAIL PROTECTED]
Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to
authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254'
-- Nobody picked up in 3 ms
-- Executing Playback(SIP/601-ea8b, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, 00491, 3) exited non-zero on 'SIP/601-ea8b'
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b'
What do I miss?
bye
Ronald
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
version:2.1
end:vcard
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Re: [Asterisk-Users] Failed to authenticate on INVITE to '601 ... solved
Ronald Wiplinger wrote:
I have installed the first time Asterisk, (forgive me simple
questions)
I have also installed the demo.
I solved it with the newest cvs version !!!
bye
Ronald
After testing demo (call 1000, call 600, ...) I changed in the
extensions.conf:
; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
exten = 5552220,1,Dial(SIP/601,20,r)
exten = 4782156,1,Dial(SIP/602,20,r)
[sipgate.de]
exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten = _0049X.,2,Playback(invalid)
exten = _0049X.,3,Hangup
[sipgate.co.uk]
exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _0044X.,2,Playback(invalid)
exten = _0044X.,3,Hangup
in sip.conf I have:
register = 5552220:[EMAIL PROTECTED]/5552220
register = 4782156:[EMAIL PROTECTED]/4782156
[601]
type=friend
username=601
secret=pwd-601
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=601
nat=yes
caller-id=Ronald 1 601
[602]
type=friend
username=602
secret=pwd-602
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=602
nat=yes
caller-id=Ronald 2 602
[sipgate.de]
type=friend
username=5552220
secret=pwd-de
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=pwd-uk
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.co.uk
nat=yes
context=incomingsipgate
canreinvite=no
The console shows when I want to dial at sipgate.de the number 1
(test) or 5 (Voicemail): 00491
-- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new
stack
-- Called [EMAIL PROTECTED]
Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to
authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254'
-- Nobody picked up in 3 ms
-- Executing Playback(SIP/601-ea8b, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, 00491, 3) exited non-zero on
'SIP/601-ea8b'
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b'
What do I miss?
bye
Ronald
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--
Ronald Wiplinger
Senior Software Engineer
AGP Telecom Co. Ltd.
Tel. (O) +886-2-2741-7890 # 7303, (M) +886-939-77-55-16
(from USA dial (408)253-3153 # 7303)
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unlawful. The views expressed in this document are not necessarily those of
AGP Telecom Co., Ltd. Notice is hereby given that no representation, contract
or other binding obligation shall be created by this e-mail, which must be
interpreted accordingly. Any representations, contractual rights or
obligations shall be separately communicated in writing and signed in the
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--
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RE: [Asterisk-Users] Failed to authenticate on INVITE
For info The new chan_sip2.c and recent CVS (yesterday) fix this and I can now use Asterisk to make calls on the sip.btcommunicator.bt.net service. If anyone wants help withthe settings, e-mail me off list. :) Peter -Original Message-From: Whisker, Peter [mailto:[EMAIL PROTECTED]Sent: 17 September 2004 14:40To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Failed to authenticate on INVITE I am getting this also. I am trying to get Asterisk to talk similarly to BT Communicator to the BT server. I can register but then the INVITE fails. BT are mixed up with theirdomains (in fact in the INVITE their software has a To: header withnumber@domain1 and an auth URI referencing number@domain2. The realm is domain1.) This can't be done in Asterisk where it is consistent about the URI. I had been blaming this, but if you are having problems too... I get the standard 407 header requesting Proxy Auth for the call. Asterisk submits the INVITE with auth and after the usual "Trying" I just get another 407. I have traces of Asterisk and the client which works and they seem so similar in what they do. I have made all the port ranges the same too. BT Communicator fails if you use port 5060 for the SIP client- they use 5052. Peter -Original Message-From: Stig Thune [mailto:[EMAIL PROTECTED]Sent: 17 September 2004 12:55To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Failed to authenticate on INVITE NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' sip.conf register = 1234:[EMAIL PROTECTED] extension.conf -- ;; Own extensions;exten = 0852509516,1,Goto(resepsjon-own,s,1) ;[resepsjon-own];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/choose) ; Meny, 1 for support, 2 for support, 3 for wx3exten = s,6,Wait(1)exten = s,7,Background(own/choosenumber) ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangupexten = 1,1,Goto(privatanslutningar,s,1)exten = 2,1,Goto(foretagsanslutningar,s,1) ; #=hangupexten = #,1,Playback(custom/no-key-registered)exten = #,2,Hangup exten = t,1,Goto(#,1) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[privatanslutningar];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/privatanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. exten = 1,1,Answerexten = 1,2,Queue(help-privatanslutningar-queue)exten = 2,1,Answerexten = 2,2,Queue(order-privatanslutningar-queue)exten = 3,1,Answerexten = 3,2,Queue(info-privatanslutningar-queue) ; #=hangup;exten = #,1,Playback(custom/no-key-registered);exten = #,2,Hangup exten = t,1,Queue(general-privatanslutningar-queue) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[foretagsanslutningar];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/foretagsanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangupexten = 1,1,Answerexten = 1,2,Queue(info-bedriftsanslutningar-queue)exten = 2,1,Answerexten = 2,2,Queue(help-bedriftsanslutningar-queue)exten = 3,1,Answerexten = 3,2,Queue(error-bedriftsanslutningar-queue) ; #=hangup;exten = #,1,Playback(custom/no-key-registered);exten = #,2,Hangup exten = t,1,Queue(general-bedriftsanslutningar-queue) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] -- The call gets into queue, then... the other phone rings.. and when I pick up - I get this message: NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' I know that the register = works.. I have checked with my SIP-provider, and they say that it is logged in. What else can be wrong ? / Stig HenningThis e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privi
[Asterisk-Users] Failed to authenticate on INVITE
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' sip.conf register = 1234:[EMAIL PROTECTED] extension.conf -- ;; Own extensions;exten = 0852509516,1,Goto(resepsjon-own,s,1) ;[resepsjon-own];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/choose) ; Meny, 1 for support, 2 for support, 3 for wx3exten = s,6,Wait(1)exten = s,7,Background(own/choosenumber) ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangupexten = 1,1,Goto(privatanslutningar,s,1)exten = 2,1,Goto(foretagsanslutningar,s,1) ; #=hangupexten = #,1,Playback(custom/no-key-registered)exten = #,2,Hangup exten = t,1,Goto(#,1) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[privatanslutningar];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/privatanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. exten = 1,1,Answerexten = 1,2,Queue(help-privatanslutningar-queue)exten = 2,1,Answerexten = 2,2,Queue(order-privatanslutningar-queue)exten = 3,1,Answerexten = 3,2,Queue(info-privatanslutningar-queue) ; #=hangup;exten = #,1,Playback(custom/no-key-registered);exten = #,2,Hangup exten = t,1,Queue(general-privatanslutningar-queue) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[foretagsanslutningar];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/foretagsanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangupexten = 1,1,Answerexten = 1,2,Queue(info-bedriftsanslutningar-queue)exten = 2,1,Answerexten = 2,2,Queue(help-bedriftsanslutningar-queue)exten = 3,1,Answerexten = 3,2,Queue(error-bedriftsanslutningar-queue) ; #=hangup;exten = #,1,Playback(custom/no-key-registered);exten = #,2,Hangup exten = t,1,Queue(general-bedriftsanslutningar-queue) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] -- The call gets into queue, then... the other phone rings.. and when I pick up - I get this message: NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' I know that the register = works.. I have checked with my SIP-provider, and they say that it is logged in. What else can be wrong ? / Stig Henning ___ 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] Failed to authenticate on INVITE
I am getting this also. I am trying to get Asterisk to talk similarly to BT Communicator to the BT server. I can register but then the INVITE fails. BT are mixed up with theirdomains (in fact in the INVITE their software has a To: header withnumber@domain1 and an auth URI referencing number@domain2. The realm is domain1.) This can't be done in Asterisk where it is consistent about the URI. I had been blaming this, but if you are having problems too... I get the standard 407 header requesting Proxy Auth for the call. Asterisk submits the INVITE with auth and after the usual "Trying" I just get another 407. I have traces of Asterisk and the client which works and they seem so similar in what they do. I have made all the port ranges the same too. BT Communicator fails if you use port 5060 for the SIP client- they use 5052. Peter -Original Message-From: Stig Thune [mailto:[EMAIL PROTECTED]Sent: 17 September 2004 12:55To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Failed to authenticate on INVITE NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' sip.conf register = 1234:[EMAIL PROTECTED] extension.conf -- ;; Own extensions;exten = 0852509516,1,Goto(resepsjon-own,s,1) ;[resepsjon-own];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/choose) ; Meny, 1 for support, 2 for support, 3 for wx3exten = s,6,Wait(1)exten = s,7,Background(own/choosenumber) ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangupexten = 1,1,Goto(privatanslutningar,s,1)exten = 2,1,Goto(foretagsanslutningar,s,1) ; #=hangupexten = #,1,Playback(custom/no-key-registered)exten = #,2,Hangup exten = t,1,Goto(#,1) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[privatanslutningar];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/privatanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. exten = 1,1,Answerexten = 1,2,Queue(help-privatanslutningar-queue)exten = 2,1,Answerexten = 2,2,Queue(order-privatanslutningar-queue)exten = 3,1,Answerexten = 3,2,Queue(info-privatanslutningar-queue) ; #=hangup;exten = #,1,Playback(custom/no-key-registered);exten = #,2,Hangup exten = t,1,Queue(general-privatanslutningar-queue) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[foretagsanslutningar];exten = s,1,Answerexten = s,2,SetMusicOnHold(default)exten = s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten = s,5,Background(own/foretagsanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangupexten = 1,1,Answerexten = 1,2,Queue(info-bedriftsanslutningar-queue)exten = 2,1,Answerexten = 2,2,Queue(help-bedriftsanslutningar-queue)exten = 3,1,Answerexten = 3,2,Queue(error-bedriftsanslutningar-queue) ; #=hangup;exten = #,1,Playback(custom/no-key-registered);exten = #,2,Hangup exten = t,1,Queue(general-bedriftsanslutningar-queue) ; If they take too long, give upexten = i,1,Playback(invalid) ; "That's not valid, try again" inmenu] -- The call gets into queue, then... the other phone rings.. and when I pick up - I get this message: NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' I know that the register = works.. I have checked with my SIP-provider, and they say that it is logged in. What else can be wrong ? / Stig Henning This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ 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] Failed to authenticate on INVITE
At 16:49 16/06/2004 -0400, Eric wrote: I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error Failed to authenticate on INVITE trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. include the line in sip.conf for each user the call insecure=yes ; To match a peer based by IP address only and not peer ___ 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] Failed to authenticate on INVITE
Hi Jason, Thanks for your reply. I didn't really want to use the insecure option, that defeats the purpose of using a password :) I was, however, able to specify user= in my sip.conf entity and that solved the problem I was having. Thanks again. - Eric On Thu, 17 Jun 2004 10:17:54 +0100 Jason Williams [EMAIL PROTECTED] wrote: At 16:49 16/06/2004 -0400, Eric wrote: I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error Failed to authenticate on INVITE trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. include the line in sip.conf for each user the call insecure=yes ; To match a peer based by IP address only and not peer ___ 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 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] Failed to authenticate on INVITE
Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error Failed to authenticate on INVITE trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric ___ 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] Failed to authenticate on INVITE
Hi, I need to inteconnection to VoIP Provider but I have this error message when i try to dial external number : -- Executing SetCallerID(SIP/491-1f64, x x ) in new stack -- Executing Dial(SIP/491-1f64, SIP/[EMAIL PROTECTED]|30|r) in new stack -- Called [EMAIL PROTECTED] May 13 18:30:16 NOTICE[294931]: chan_sip.c:5013 handle_response: Failed to authenticate on INVITE to 'x SIP/[EMAIL PROTECTED];tag=as3f82df08' Any help are welcome. Sorry for the first message; Zouhair Echchelh. OPTION-SERVICE.FR --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.647 / Virus Database: 414 - Release Date: 29/03/2004 ___ 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
