Re: [asterisk-users] sip registration
Got it... Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 954) Asterisk*CLI sip show registry Hostdnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060 N 444222146 105 Registered Sun, 07 Apr 2013 09:42:31 1 SIP registrations. Asterisk*CLI Next hurdle is extensions.conf I must need to establish / correlate my DID number to something. When I dial my DID I get you have reached a non working number On Sat, Apr 6, 2013 at 5:36 PM, Steve Edwards asterisk@sedwards.comwrote: A better subject will yield better replies. On Sat, 6 Apr 2013, Thomas Perron wrote: Shouldnt I be able to at least ping the SIP provider IP? Not if they don't allow it. They don't. sip3.voipvoip.com registers fine for me with your credentials. Did you put the registration statement in the [general] section? I use the 'append' format so I can group all the cruft for a provider together. Like: ; voipvoip.com [general](+) register= nn:xx@sip3.** voipvoip.com/nnhttp://nn:xxx...@sip3.voipvoip.com/nn [outgoing] secret = xx username= nn ... I have not configured anything other then entries in the sip.conf I used your credentials and successfully placed a call to all of my Caribbean premium numbers*. Please change your password. Maybe your issue lies elsewhere. Does enabling SIP debugging on the console yield any clues? *) just kidding. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- 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-**usershttp://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] sip registration
Please don't top post. On Sun, 7 Apr 2013, Thomas Perron wrote: Got it... Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 954) Asterisk*CLI sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060 N 444222146 105 Registered Sun, 07 Apr 2013 09:42:31 1 SIP registrations. Asterisk*CLI Next hurdle is extensions.conf I must need to establish / correlate my DID number to something. When I dial my DID I get you have reached a non working number On Sat, Apr 6, 2013 at 5:36 PM, Steve Edwards asterisk@sedwards.com wrote: A better subject will yield better replies. On Sat, 6 Apr 2013, Thomas Perron wrote: Shouldnt I be able to at least ping the SIP provider IP? Not if they don't allow it. They don't. sip3.voipvoip.com registers fine for me with your credentials. Did you put the registration statement in the [general] section? I use the 'append' format so I can group all the cruft for a provider together. Like: ; voipvoip.com [general](+) register = nn:xxx...@sip3.voipvoip.com/nn [outgoing] secret = xx username = nn ... I have not configured anything other then entries in the sip.conf I used your credentials and successfully placed a call to all of my Caribbean premium numbers*. Please change your password. Maybe your issue lies elsewhere. Does enabling SIP debugging on the console yield any clues? *) just kidding. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 registration
I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command sip show registry and do not see it set up. I run sip show peers and I do see an entry. I have not configured anything other then entries in the sip.conf results are: Name/username HostDyn Forcerport ACL Port Status Description outgoing/5552530146 (your 69.90.209.57 5060 OK (85 ms) 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] Asterisk*CLI sip show registry Hostdnsmgr Username Refresh StateReg.Time 0 SIP registrations. Asterisk*CLI my config is this: [outgoing] username=5552530146 (your VoIP VoIP account assigned while signing up) type=peer qualify=yes secret=iblockedthis (your VoIP VoIP password) nat=auto insecure=invite,port host=sip3.voipvoip.com fromuser=5552530146 (your VoIP VoIP account assigned while signing up) fromdomain=sip3.voipvoip.com dtmfmode=rfc2833 disallow=all allow=g729 allow=ilbc allow=ulaw allow=alaw ; ; ; ; ; ;register = 5552530146:7036361399@69.90.209.57/5552530146 register=5552530146:boston!@#1...@sip3.voipvoip.com/5552530146 ; Please send input or guidance... Thanks Thomas -- _ -- 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 registration
A better subject will yield better replies. On Sat, 6 Apr 2013, Thomas Perron wrote: Shouldnt I be able to at least ping the SIP provider IP? Not if they don't allow it. They don't. sip3.voipvoip.com registers fine for me with your credentials. Did you put the registration statement in the [general] section? I use the 'append' format so I can group all the cruft for a provider together. Like: ; voipvoip.com [general](+) register= nn:xxx...@sip3.voipvoip.com/nn [outgoing] secret = xx username= nn ... I have not configured anything other then entries in the sip.conf I used your credentials and successfully placed a call to all of my Caribbean premium numbers*. Please change your password. Maybe your issue lies elsewhere. Does enabling SIP debugging on the console yield any clues? *) just kidding. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 registration Asterisk 1.8
Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register = 808:passw...@as2.x.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99 callerid=child2 808 disallow=all allow=ulaw allow=alaw username=808 secret=x dtmfmode=rfc2833 host=dynamic mailbox=808 nat=yes qualify=yes canreinvite=no == Extension Changed 800[sip-phones] new state Idle for Notify User 812 [Oct 8 09:48:37] NOTICE[12030]: chan_sip.c:26141 sip_poke_noanswer: Peer '808' is now UNREACHABLE! Last qualify: 1 == Using SIP RTP CoS mark 5 - Executing [808@sip-phones:1] Dial(SIP/815-00d8, SIP/808,20,t) in new stack [Oct 8 09:49:02] WARNING[12277]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/815-00d8' status is 'CHANUNAVA Any ideas? Thanks in Advance! -- _ -- 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 registration Asterisk 1.8
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz Sent: Monday, October 08, 2012 12:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sip registration Asterisk 1.8 Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register = 808:passw...@as2.x.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99 callerid=child2 808 disallow=all allow=ulaw allow=alaw username=808 secret=x dtmfmode=rfc2833 host=dynamic mailbox=808 nat=yes qualify=yes canreinvite=no == Extension Changed 800[sip-phones] new state Idle for Notify User 812 [Oct 8 09:48:37] NOTICE[12030]: chan_sip.c:26141 sip_poke_noanswer: Peer '808' is now UNREACHABLE! Last qualify: 1 == Using SIP RTP CoS mark 5 - Executing [808@sip-phones:1] Dial(SIP/815-00d8, SIP/808,20,t) in new stack [Oct 8 09:49:02] WARNING[12277]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/815-00d8' status is 'CHANUNAVA Any ideas? Thanks in Advance! -- IIRC qualify=yes means you get 60 seconds; try it with qualify=300. -- _ -- 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 Registration Hijacking
On 20/01/12 01:36, eherr wrote: It is also register on an AudioCodes MP-118. Thanks, -E Is the Audiocodes gateway accessible online? Have you set a strong password for it's web interface (and cli if it has one)? It is possible someone is breaking into that and getting the SIP password out of it. cheers, Paul. -- _ -- 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 Registration Hijacking
It is accessible from HTTP. However, the access list only allows access from my home and the password is strong. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hayes Sent: Thursday, January 26, 2012 10:30 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sip Registration Hijacking On 20/01/12 01:36, eherr wrote: It is also register on an AudioCodes MP-118. Thanks, -E Is the Audiocodes gateway accessible online? Have you set a strong password for it's web interface (and cli if it has one)? It is possible someone is breaking into that and getting the SIP password out of it. cheers, Paul. -- _ -- 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] Sip Registration Hijacking
On Thu, 26 Jan 2012, eherr wrote: It is accessible from HTTP. However, the access list only allows access from my home and the password is strong. Can you configure it to 'syslog' accesses where you can monitor it. Maybe your access lists are invalid, misunderstood or not being honored. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 Registration Hijacking
Can you please elaborate on rate limiting. Not how to implement it but rather how implementation is beneficiary. Reading up on it, it appears that it just checks the tcp connections and denys connection if limit is passed. In my thoughts, this is essentially a live fail2ban monitor in respects to attempted authentications. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim DeVito Sent: Saturday, January 21, 2012 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip Registration Hijacking Rate limiting (google) via iptables FTW! Good luck! - Original message - Alejandro Imass wrote 20.01.2012 18:09: I would like to know how to block this MF because he makes calls at 1-2 AM I use this construction on my servers [users] exten = _XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1) [block] exten = _X.,1,HangUp(1) -- With Best Regards Mikhail Lischuk -- _ -- 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 Registration Hijacking
This is actually an interesting concept however I do think I want to restrict dialing during a specific time period. If someone is in the office, I would have to reprogram the route so allow dialing which adds overhead. Again, I do like the concept though. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk Sent: Friday, January 20, 2012 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip Registration Hijacking Alejandro Imass wrote 20.01.2012 18:09: I would like to know how to block this MF because he makes calls at 1-2 AM I use this construction on my servers [users] exten = _XXX,1,GotoIfTime(1:00-2:00,*,*,*?block,1,1) [block] exten = _X.,1,HangUp(1) -- With Best Regards Mikhail Lischuk mailto:mlisc...@itx.com.ua -- _ -- 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 Registration Hijacking
I appreciate your 2-cents worth. However, I do not believe they have access to machine If so, they are clever to create three failures in the logs for my benefit before entering the correct one for hijacking. Additionally, I have a lot of sip extensions to hijack and he keeps going for the same one. I was hoping this was something with the MP-118 and someone experienced the same thing with that device. Either way, I posed two questions which are still unanswered and probably I will never get answered: 1 - is this a vulnerability in the MP-118 2 - what method could they possibly be using to hijack a number-alpha extension which is creative to begin with ie) 203-Joes_Insurance_Service with an openssl generated password of 12 characters. Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Saturday, January 21, 2012 1:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip Registration Hijacking On 20/01/2012 9:36 AM, eherr wrote: I have a honey pot box with extensions that are not just numbers ie ) 100-MySipUserName And the passwords are from an openssl generated password ie) Gq5VNIjDFWIQoUT6 Is the password stored in sip.conf in plain text or as an MD5? If it is stored in plain text then it may suggest the hijacker has greater access to your system than you realise. My 2-cents worth. -- _ -- 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 Registration Hijacking
On Thu, Jan 19, 2012 at 8:36 PM, eherr email.eherr9...@gmail.com wrote: I have a honey pot box with extensions that are not just numbers ie ) 100-MySipUserName I have the same problem and I use contactpermit with specific ip blocks! I know for a fact I'm getting hijacked by sip vicious on extension 100 but I can't understand how because I don't even have an extension 100 declared anywhere. I would like to know how to block this MF because he makes calls at 1-2 AM -- Alejandro Imass -- _ -- 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 Registration Hijacking
I always thought Sip Vicious only does numbers ( 0 - 100 ) not Numberic-Alpha ( 100-MySipUserName ). To make my situation more interesting is that I also have fail2ban installed banning after 5 failed attempts. This hijack is only happening to an extension on the honeypot audiocodes with the sip reg authenticating back to my honey pot asterisk which is why I thought it might be a vulnerability in the audiocodes. However, the hijacker manages to make it past the fail2ban and gets the sip reg. I see sipvicious attempts all the time where they run checks against extensions 0 - . Sometimes I see alpha extension name attempts but I do not know how that's done. --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Sent: Friday, January 20, 2012 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sip Registration Hijacking On Thu, Jan 19, 2012 at 8:36 PM, eherr email.eherr9...@gmail.com wrote: I have a honey pot box with extensions that are not just numbers ie ) 100-MySipUserName I have the same problem and I use contactpermit with specific ip blocks! I know for a fact I'm getting hijacked by sip vicious on extension 100 but I can't understand how because I don't even have an extension 100 declared anywhere. I would like to know how to block this MF because he makes calls at 1-2 AM -- Alejandro Imass -- _ -- 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] Sip Registration Hijacking
On Fri, Jan 20, 2012 at 11:17 AM, eherr email.eherr9...@gmail.com wrote: I always thought Sip Vicious only does numbers ( 0 - 100 ) not Numberic-Alpha ( 100-MySipUserName ). To make my situation more interesting is that I also have fail2ban installed banning after 5 failed attempts. I too have fail2ban and running a relatively updated version of FreeBSD. BTW my install is plain Asterisk -- Alejandro Imass -- _ -- 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 Registration Hijacking
I have a honey pot box with extensions that are not just numbers ie ) 100-MySipUserName And the passwords are from an openssl generated password ie) Gq5VNIjDFWIQoUT6 However, this one extension keeps getting hacked and showing up on a different IP address. It is also register on an AudioCodes MP-118. I wanted to know if anyone else ran into this and if it's a vulnerability on the MP-118 or with Asterisk. Thanks, -E -- _ -- 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 registration issues
Hi, Having problems with a client trying to login to Asterisk 1.6.2 from behind a DSL router. The account can be accessed perfectly from other clients. Would appreciate if you could look at the the attached log and see if you spot any glaring issues. The user is very infrequently available for discussion and testing, so please try to batch questions in one mail itself! Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: sip:ACCOUNT-ID@SERVER-IP Contact: sip:ACCOUNT-ID@CLIENT-IP:49153;q=1, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.667, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 - --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) --- Transmitting (no NAT) to CLIENT-IP:49153 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d To: sip:ACCOUNT-ID@SERVER-IP;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=09c83283 Content-Length: 0 Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) --- SIP read from UDP:CLIENT-IP:49152 --- REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: sip:ACCOUNT-ID@SERVER-IP Contact: sip:ACCOUNT-ID@CLIENT-IP:49153;q=1, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.667, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 - --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) --- Transmitting (no NAT) to CLIENT-IP:49153 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d To: sip:ACCOUNT-ID@SERVER-IP;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=09c83283 Content-Length: 0 Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) --- SIP read from UDP:CLIENT-IP:49152 --- REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: sip:ACCOUNT-ID@SERVER-IP Contact: sip:ACCOUNT-ID@CLIENT-IP:49153;q=1, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.667, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 - --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) --- Transmitting (no NAT) to CLIENT-IP:49153 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d To: sip:ACCOUNT-ID@SERVER-IP;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=09c83283 Content-Length: 0 Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) --- SIP read from UDP:CLIENT-IP:49152 --- REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP
Re: [asterisk-users] SIP registration issues
I have not looked at the log files, but often times DSL routers may use PPPoE which has a little bit of overhead so you need to set the MTU below the default of 1500. Some info about the issue can be found here: http://www.ezlan.net/PPPOE.html and http://www.cisco.com/en/US/tech/tk175/tk15/technologies_tech_note09186a0080093bc7.shtml. Another issue could be that the DSL router is doing a nat and you need to set nat=yes in sip.conf to get things to work. - Original Message - From: Raj Mathur (राज माथुर) r...@linux-delhi.org To: asterisk-users@lists.digium.com Sent: Saturday, November 19, 2011 8:43:22 PM Subject: [asterisk-users] SIP registration issues Hi, Having problems with a client trying to login to Asterisk 1.6.2 from behind a DSL router. The account can be accessed perfectly from other clients. Would appreciate if you could look at the the attached log and see if you spot any glaring issues. The user is very infrequently available for discussion and testing, so please try to batch questions in one mail itself! Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] SIP registration DoS but no logs in messages
On 17/03/11 05:37, Patrick wrote: Dear mailing list, I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian and I've a strange behavior. After some days running normally, my asterisk is under heavy attack, however, there is nothing logged in the console (logging from debug - error) or file (level from notice -error) I can see that there is also a peak on the network traffic. My first guess is that I'm suffering from a SIP registration DoS, but, as there is nothing logged about a not matching peer or incorrect password logged to file, my fail2ban script is not blocking the attacker. I normally restarts Asterisk and logs are restarting to log attacks, but, today, it's not working FYI, I've checked and my loggers are not muted and the logging level is at least notice. I've also reloaded my loggers but no effect. Do you already have experienced such situation ? Is there any known issue with logging module stopping while Asterisk is DoS'ed ? Best regards, Patrick It's possible that fail2ban has already blocked the incoming registration attempts but the attacker is still blindly sending packets to you. Often a sign the attacker is using an old version of sip-vicious, you can often stop such things by using the svcrash.py script they now provide. Check your iptables logs. cheers, Paul. -- _ -- 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 registration DoS but no logs in messages
Dear mailing list, I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian and I've a strange behavior. After some days running normally, my asterisk is under heavy attack, however, there is nothing logged in the console (logging from debug - error) or file (level from notice -error) I can see that there is also a peak on the network traffic. My first guess is that I'm suffering from a SIP registration DoS, but, as there is nothing logged about a not matching peer or incorrect password logged to file, my fail2ban script is not blocking the attacker. I normally restarts Asterisk and logs are restarting to log attacks, but, today, it's not working FYI, I've checked and my loggers are not muted and the logging level is at least notice. I've also reloaded my loggers but no effect. Do you already have experienced such situation ? Is there any known issue with logging module stopping while Asterisk is DoS'ed ? Best regards, Patrick -- _ -- 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 registration
Hi, Are sip.conf's defaultexpiry and maxexpiry global? Or can they be used on a per-extension basis? I'd like to force some extensions to re-register more frequently than others (server-side). Thanks, Vieri -- _ -- 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 registration failure stops all SIP activity
I have a problem that when one of my SIP providers has a problem the rest of my SIP extensions and trunks stop working until either the SIP provider fixes the problem or Asterisk stops trying to register to that provider. Why does this happen? A single provider having problems should not grind everything else to a halt! At this moment I either have to comment the register lines for that provider or wait until the registration times out (I have 10 attempts and 60 second delay in sip.conf). During that time all sip phones have no service and other trunk providers (SIP) are all UNREACHABLE. Is there something I can change in my sip.conf to prevent this problem? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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 Registration Failure Logging
Let's say I have two Asterisk boxes, A and B. I am trying to get A to do SIP registration on B, so an extension for A can dial SIP phones covered by B. If I examine the logs on B, if the registration succeeds, I am seeing a notice to that effect on B. But if the registration *fails*, i'm not seeing any message logged on B. Maybe I'm just not looking in the right place. Is there a way to turn on logging or debugging so registration failures are logged on the target? I'm doing something profoundly stupid, and seeing the notorious chan_sip.c:12009 handle_response_invite: Failed to authenticate on INVITE message, and trying to trace why. -Thanks, Jim -- _ -- 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
Re: [asterisk-users] SIP Registration Failure Logging
Try: core set verbose 4 From the Asterisk CLI -uzzi PS: If you're not seeing any connection information, be sure to double-check the IP address is correct. Learned that lesson the hard way =\ On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg j...@amanue.com wrote: Let's say I have two Asterisk boxes, A and B. I am trying to get A to do SIP registration on B, so an extension for A can dial SIP phones covered by B. If I examine the logs on B, if the registration succeeds, I am seeing a notice to that effect on B. But if the registration *fails*, i'm not seeing any message logged on B. Maybe I'm just not looking in the right place. Is there a way to turn on logging or debugging so registration failures are logged on the target? I'm doing something profoundly stupid, and seeing the notorious chan_sip.c:12009 handle_response_invite: Failed to authenticate on INVITE message, and trying to trace why. -Thanks, Jim -- _ -- 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 -- _ -- 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 registration fails
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register = 092779077:x...@85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret= fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes nat=yes disallow=all allow=gsm allow=alaw [Jun 25 16:54:32] NOTICE[32550]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #54) Really destroying SIP dialog '628e05295c1a2cc560d1c6c073b85...@127.0.0.1' Method: REGISTER Really destroying SIP dialog '4f9b2b7a241f3f2a193ceb0020778...@192.168.2.2' Method: OPTIONS Retransmitting #4 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK15be023b;rport From: sip:092779...@85.119.188.3;tag=as36b44350 To: sip:092779...@85.119.188.3 Call-ID: 628e05295c1a2cc560d1c6c073b85...@127.0.0.1 CSeq: 156 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:s...@192.168.2.2 Event: registration Content-Length: 0 --- Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: asterisk sip:aster...@192.168.2.2;tag=as6cd2d842 To: sip:sip.3starsnet.com Contact: sip:aster...@192.168.2.2 Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: asterisk sip:aster...@192.168.2.2;tag=as6cd2d842 To: sip:sip.3starsnet.com Contact: sip:aster...@192.168.2.2 Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #5 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK15be023b;rport From: sip:092779...@85.119.188.3;tag=as36b44350 To: sip:092779...@85.119.188.3 Call-ID: 628e05295c1a2cc560d1c6c073b85...@127.0.0.1 CSeq: 156 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:s...@192.168.2.2 Event: registration Content-Length: 0 --- Retransmitting #2 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: asterisk sip:aster...@192.168.2.2;tag=as6cd2d842 To: sip:sip.3starsnet.com Contact: sip:aster...@192.168.2.2 Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #3 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: asterisk sip:aster...@192.168.2.2;tag=as6cd2d842 To: sip:sip.3starsnet.com Contact: sip:aster...@192.168.2.2 Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #4 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: asterisk sip:aster...@192.168.2.2;tag=as6cd2d842 To: sip:sip.3starsnet.com Contact: sip:aster...@192.168.2.2 Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ___ -- 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
Re: [asterisk-users] SIP registration fails
SIP-registration errors are solved by restarting the Asterisk-server. But I expect them to return in time... I can make call now, but the other end does not hear me. So problem with RTP-flow... Can someone guide me to the solution ? On the Asterisk-server I have this (iptables): -A RH-Firewall-1-INPUT -p udp --dport 4569 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp --dport 5060 -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 11000:11500 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 25 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited In rtp.conf I have this : rtpstart=11000 rtpend=11500 Asterisk is behind firewall. Endian firewall has following configuration : enable SIP proxy transparant RTP port low : 11000 RTP port high : 11500 Firewall port forwarding : uplink:5060 asterisk_ip:5060 Asterisk himself says : -- Executing [050510...@intern:1] NoOp(SIP/grandstream-09813b58, via 3StarsNet) in new stack -- Executing [050510...@intern:2] Dial(SIP/grandstream-09813b58, SIP/3starsnet/050510484) in new stack -- Called 3starsnet/050510484 -- SIP/3starsnet-0981bf08 is making progress passing it to SIP/grandstream-09813b58 -- SIP/3starsnet-0981bf08 answered SIP/grandstream-09813b58 == Spawn extension (intern, 050510484, 2) exited non-zero on 'SIP/grandstream-09813b58' What do I need in sip.conf to overcome these rtp-problems ?? I have : externip=78.21.62.99 canreinvite=no jbenable = yes [3starsnet] type=peer ... nat=yes ... Thanks for the help ! Jonas. On Thu, 2009-06-25 at 17:25 +0200, jonas kellens wrote: Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? ___ -- 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
Re: [asterisk-users] SIP Registration and INVITE question
6 apr 2009 kl. 18.46 skrev Steve Davies: Thanks for the reply - Perhaps I was not clear. On the register= line, if I set /extension to be /12345, then this just replaces 's' with 12345, and ALL calls, regardless of their destination number will be routed on the INVITE line to 12...@x.x.x.x, and the actual destination is specified in the To: header. Not particularly useful, and I'd prefer not to have to go fumbling through the SIP headers to find what was really dialled :) Looking at the SIP RFC, the idea is that you specify a set of What I will accept details with each registration in the Contact: headers, which is intended to include _multiple_ possible destination addresses. The Registrar will then only ever send calls addressed to that list of destinations. Sadly, the RFC authors did not think to consider private point-to-point links where you can usefully say send me anything, you know best. Asterisk fills by defaulting to a single s...@x.x.x.x, where the 's' can be replaced by any single number. The REGISTER request in the RFC was really written for a device. The way providers use it for trunks with multiple DIDs is outside of the RFC and is discussed in relation to the SIPconnect specification in the SIP forum. Some providers solve this by not using the Contact: in the register request at all for the calls, instead guessing a URI with the DID in the user name part, something that breaks communication even more as the Contact might include other hints on call routing internally, like line button in a SNOM phone. I would say that the only way right now is to parse the To: header. I started working on some code a while ago that would handle this better, but never completed it. We simply registered a random string and then replaced it with whatever was sent in the To: header (which should be the original destination) before hitting the dialplan. That code still exists in a branch somewhere and in Pineapple. This code would also solve the issue with registering multiple accounts with one provider. /O --- * Olle E. Johansson - o...@edvina.net * Asterisk Training http://edvina.net/training/ ___ -- 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
Re: [asterisk-users] SIP Registration and INVITE question
2009/4/7 Olle E. Johansson o...@edvina.net:
[snip]
The REGISTER request in the RFC was really written for a device.
The way providers use it for trunks with multiple DIDs is outside of the
RFC and is discussed in relation to the SIPconnect specification in
the SIP forum.
Some providers solve this by not using the Contact: in the register
request at all for the calls, instead guessing a URI with the DID
in the user name part, something that breaks communication
even more as the Contact might include other hints on call routing
internally, like line button in a SNOM phone.
I would say that the only way right now is to parse the To: header.
I started working on some code a while ago that would handle
this better, but never completed it. We simply registered a random
string and then replaced it with whatever was sent in the To: header
(which should be the original destination) before hitting the dialplan.
That code still exists in a branch somewhere and in Pineapple.
This code would also solve the issue with registering multiple
accounts with one provider.
/O
Thanks Olle, as always, a useful response :)
In the meantime, I suspect that the following is the current dialplan
based workaround for calls that come in to 's' because of a default
Registration Contact?
[default]
exten = s,1,Set(DN=${SIP_HEADER(TO):5})
exten = s,n,Set(DN=${CUT(DN,@,1)})
exten = s,n,GotoIf($[${DN} = s]?:default,${DN},1)
exten = s,n,Hangup()
Comments or improvements anyone?
Thanks again.
Steve
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Re: [asterisk-users] SIP Registration and INVITE question
7 apr 2009 kl. 12.08 skrev Steve Davies: 2009/4/7 Olle E. Johansson o...@edvina.net: [snip] The REGISTER request in the RFC was really written for a device. The way providers use it for trunks with multiple DIDs is outside of the RFC and is discussed in relation to the SIPconnect specification in the SIP forum. Some providers solve this by not using the Contact: in the register request at all for the calls, instead guessing a URI with the DID in the user name part, something that breaks communication even more as the Contact might include other hints on call routing internally, like line button in a SNOM phone. I would say that the only way right now is to parse the To: header. I started working on some code a while ago that would handle this better, but never completed it. We simply registered a random string and then replaced it with whatever was sent in the To: header (which should be the original destination) before hitting the dialplan. That code still exists in a branch somewhere and in Pineapple. This code would also solve the issue with registering multiple accounts with one provider. /O Thanks Olle, as always, a useful response :) In the meantime, I suspect that the following is the current dialplan based workaround for calls that come in to 's' because of a default Registration Contact? Yes, if you don't add an extension at the end of the register= configuration, Asterisk defaults to s which really is used all around Asterisk when we don't have a given extension. /O ___ -- 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 Registration and INVITE question
I have an ITSP we are trying to work with that has an Unusual way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: sip:s...@x.x.x.x This then causes ALL calls from the ITSP inbound to us to be addressed: INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] In fact, whatever we send in the Contact: header is reflected in the INVITE for inbound calls, and the actual number dialled is always placed in the To: header. What 99.9% of our ITSPs would send is: INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] As you can see, the correct destination number is placed into the INVITE header AND the To: header, and Asterisk routes it correctly based on the INVITE. My questions: - Is there a way of telling chan_sip to register with multiple Contact: headers in the registration request, so that all of the acceptable DDI numbers can be presented to the ITSP (This is what the RFC seems to suggest is the correct way to operate.) - Alternatively, has anyone encountered this previously, and perhaps created an s extension that then digs into the To: header, and routes according to that? Examples, workarounds and solutions are all welcome! Help? Thanks, Steve ___ -- 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
Re: [asterisk-users] SIP Registration and INVITE question
Have you looked at the syntax of register = keyword ? register = [transport://]user[:secret[:authuse...@host[:port][/extension] ; If no extension is given, the 's' extension is used. There you have it ... Contact: sip:s set the extension and you should be fine Martin On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies davies...@gmail.com wrote: I have an ITSP we are trying to work with that has an Unusual way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: sip:s...@x.x.x.x This then causes ALL calls from the ITSP inbound to us to be addressed: INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] In fact, whatever we send in the Contact: header is reflected in the INVITE for inbound calls, and the actual number dialled is always placed in the To: header. What 99.9% of our ITSPs would send is: INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] As you can see, the correct destination number is placed into the INVITE header AND the To: header, and Asterisk routes it correctly based on the INVITE. My questions: - Is there a way of telling chan_sip to register with multiple Contact: headers in the registration request, so that all of the acceptable DDI numbers can be presented to the ITSP (This is what the RFC seems to suggest is the correct way to operate.) - Alternatively, has anyone encountered this previously, and perhaps created an s extension that then digs into the To: header, and routes according to that? Examples, workarounds and solutions are all welcome! Help? Thanks, Steve ___ -- 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 ___ -- 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
Re: [asterisk-users] SIP Registration and INVITE question
Thanks for the reply - Perhaps I was not clear. On the register= line, if I set /extension to be /12345, then this just replaces 's' with 12345, and ALL calls, regardless of their destination number will be routed on the INVITE line to 12...@x.x.x.x, and the actual destination is specified in the To: header. Not particularly useful, and I'd prefer not to have to go fumbling through the SIP headers to find what was really dialled :) Looking at the SIP RFC, the idea is that you specify a set of What I will accept details with each registration in the Contact: headers, which is intended to include _multiple_ possible destination addresses. The Registrar will then only ever send calls addressed to that list of destinations. Sadly, the RFC authors did not think to consider private point-to-point links where you can usefully say send me anything, you know best. Asterisk fills by defaulting to a single s...@x.x.x.x, where the 's' can be replaced by any single number. Most ITSPs work around this by assuming that they know best, and routing numbers even if they are missing from the registration. The odd exception does not do this. I suspect that the solution will be to register with a /extension of /pedanticitsp, and then have a dialplan which pulls and parses the SIP To: header. Other suggestions are still welcome. Regards, Steve 2009/4/6 Martin asteriskl...@callthem.info: Have you looked at the syntax of register = keyword ? register = [transport://]user[:secret[:authuse...@host[:port][/extension] ; If no extension is given, the 's' extension is used. There you have it ... Contact: sip:s set the extension and you should be fine Martin On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies davies...@gmail.com wrote: I have an ITSP we are trying to work with that has an Unusual way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: sip:s...@x.x.x.x This then causes ALL calls from the ITSP inbound to us to be addressed: INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] In fact, whatever we send in the Contact: header is reflected in the INVITE for inbound calls, and the actual number dialled is always placed in the To: header. What 99.9% of our ITSPs would send is: INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] As you can see, the correct destination number is placed into the INVITE header AND the To: header, and Asterisk routes it correctly based on the INVITE. My questions: - Is there a way of telling chan_sip to register with multiple Contact: headers in the registration request, so that all of the acceptable DDI numbers can be presented to the ITSP (This is what the RFC seems to suggest is the correct way to operate.) - Alternatively, has anyone encountered this previously, and perhaps created an s extension that then digs into the To: header, and routes according to that? Examples, workarounds and solutions are all welcome! Help? Thanks, Steve ___ -- 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 ___ -- 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 ___ -- 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 Registration
Hi, I have this weird problem i cant explain. i have two asterisk, i'm using realtime table for my sip/user accounts. my database is on a mysql cluster. my prob is if i register on phone on asterisk 1 it is ok, but on second asterisk it can't, Registration from '122144 sip:[EMAIL PROTECTED]:5060' failed for '12.34.56.78' - Wrong password but both asterisk talks to a single mysql cluster. i defined this on my sip.conf domain=10.10.10.130 domain=10.10.10.131 domain=my.domain.com any ides? TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 registration timeout/expiration
Hi, If I set maxexpirey=60 in sip.conf and also set a registration timeout=60 on client software, doesn't this mean that the SIP user (an ATA connected phone) should be forced to re-register every minute? If I look at the CLI when the SIP user registers I do see a statement regarding a 60 second timeout. However, after 1 minute I don't see it unregister and register again (debug is on). I'm asking this because in my LAN I have a DNS server which is dynamically updated (via a script) with both A and SRV records with very short TTLs. The idea is that the LAN SIP clients (both softphones and ATA-connected phones) switch from one failing (or down for maintenance) server to another active box. This part seems to work fine. However, I'm having trouble getting the SIP registrations back to the first server when the latter is back on-line. The only way I found to do this within a minute is to kill asterisk on box 2 and all accounts will register on box 1 (even if the 5-second-TTL A records have been updated and/or the SRV entries give box1 a much higher priority). How can I make them move to box 1 without bringing down box 2? It seems as though maxexpirey is not taken into account. The extensions will stay on box 2 and will move to box 1 only if: - box 2 dies - or I wait around 30 minutes (I don't what this timeout could be) I've tried it on Asterisk 1.4.21.2 and 1.2.30. Any ideas? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 registration timeout/expiration
you have this option on major phones also, try that. 2008/7/31 Vieri [EMAIL PROTECTED] Hi, If I set maxexpirey=60 in sip.conf and also set a registration timeout=60 on client software, doesn't this mean that the SIP user (an ATA connected phone) should be forced to re-register every minute? If I look at the CLI when the SIP user registers I do see a statement regarding a 60 second timeout. However, after 1 minute I don't see it unregister and register again (debug is on). I'm asking this because in my LAN I have a DNS server which is dynamically updated (via a script) with both A and SRV records with very short TTLs. The idea is that the LAN SIP clients (both softphones and ATA-connected phones) switch from one failing (or down for maintenance) server to another active box. This part seems to work fine. However, I'm having trouble getting the SIP registrations back to the first server when the latter is back on-line. The only way I found to do this within a minute is to kill asterisk on box 2 and all accounts will register on box 1 (even if the 5-second-TTL A records have been updated and/or the SRV entries give box1 a much higher priority). How can I make them move to box 1 without bringing down box 2? It seems as though maxexpirey is not taken into account. The extensions will stay on box 2 and will move to box 1 only if: - box 2 dies - or I wait around 30 minutes (I don't what this timeout could be) I've tried it on Asterisk 1.4.21.2 and 1.2.30. Any ideas? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 registration
Hi, I have this weird problem i cant explain. i have two asterisk, i'm using realtime table for my sip/user accounts. my database is on a mysql cluster. my prob is if i register on phone on asterisk 1 it is ok, but on second asterisk it can't, Registration from '122144 sip:[EMAIL PROTECTED]:5060' failed for '12.34.56.78' - Wrong password but both asterisk talks to a single mysql cluster. i defined this on my sip.conf domain=10.10.10.130 domain=10.10.10.131 domain=my.domain.com any ides? TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 Registration!
Hi, I have been using asterisk-1.4.17 version. Have a SIP registration from bandtel sip providers. Use DID numbers for the incoming calls which works fine when i dont use any peer setting in my sip.conf file. But when i use a peer and make calls thru the DID number it doesn't reach asterisk at all. Doesnt give me any errors as well. peer in my sip conf is as given below: [proxy2_bandtel] type=peer username=206**1 secret=*** fromdomain=206**1 host=proxy1.bandtel.com qualify=yes outboundproxy=proxy1.bandtel.com If i dont use the above code in sip.conf file the DID number reaches asterisk and completes the incoming call. But i need to have the above code to register SIP, which should give me the status as ok when i run the command 'sip show peers'. Can you please let me know why this doesnt work with the above code. Thanks and appreciate your response. Regards, Naveen.Palani “Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.” This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus. For more details about our company, visit http://www.Quinnox.com ___ -- 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
Re: [asterisk-users] SIP registration problem
I have seen this issue where there were internet connectivity issues. Asterisk registers every so often with the ITS. For some reason or another (it can be many reasons such as DNS, internet, ISP has issue etc). asterisk cant re-register so it keeps trying. As far as the so context if you have a simple register line in sip.conf (such as register= axe:[EMAIL PROTECTED]) then asterisk will tell the server that it is registering it with to send all calls to the s extension in your default context. - Original Message - From: Michelle Dupuis To: asterisk-users@lists.digium.com Sent: Saturday, May 05, 2007 4:08 PM Subject: [asterisk-users] SIP registration problem I've reposted with a more meaningful subject - hopefully someone will replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to [EMAIL PROTECTED] REGISTER attempt 2 to [EMAIL PROTECTED] Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why is our asterisk box being associated with the entryunauthorized context, not the entryinternal context? (See below) 3. Why is the contact sip:[EMAIL PROTECTED]:5060 in our SIP messages, why s@ anything? Thanks MD -- Contents of sip.conf at ITSP: [999] context=entryinternal ; I know this context exists! This is the right context. type=friend username=999 secret= callerid=Test 999 host=dynamic nat=no canreinvite=no allow=ulaw allow=alaw dtmfmode=rfc2833 --- Console log from ITSP show strange SIP traffic: --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms pbx*CLI pbx*CLI -- SIP read from 123.183.86.231:5060: REGISTER sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;rport From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=999, realm=pbx.itsp.com, algorithm=MD5, uri=sip:pbx.itsp.com, nonce=5cec66c0, response=6451967016fc38f896efeb7247523fe1, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060 Event: registration Content-Length: 0 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 123.183.86.231 : 5060 (NAT) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=5060 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=5060 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED];tag=as7d680d48 Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;expires=120 Date: Fri, 04 May 2007 19:27:58 GMT ontent-Length: 0 -- SIP read from 123.183.86.231:5060: OPTIONS sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 19:38:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (12 headers 0 lines) --- Looking for s in entryunauthorized (domain pbx.itsp.com) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;received=123.183.86.231;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com;tag=as51d476cd Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:74.110.57.25 Accept: application/sdp Content-Length: 0 -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
[asterisk-users] SIP registration problem
I've reposted with a more meaningful subject - hopefully someone will replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to [EMAIL PROTECTED] REGISTER attempt 2 to [EMAIL PROTECTED] Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why is our asterisk box being associated with the entryunauthorized context, not the entryinternal context? (See below) 3. Why is the contact sip:[EMAIL PROTECTED]:5060 in our SIP messages, why s@ anything? Thanks MD -- Contents of sip.conf at ITSP: [999] context=entryinternal ; I know this context exists! This is the right context. type=friend username=999 secret= callerid=Test 999 host=dynamic nat=no canreinvite=no allow=ulaw allow=alaw dtmfmode=rfc2833 --- Console log from ITSP show strange SIP traffic: --- Scheduling destruction of call mailto:'[EMAIL PROTECTED]' '[EMAIL PROTECTED]' in 15000 ms pbx*CLI pbx*CLI -- SIP read from 123.183.86.231:5060: REGISTER sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;rport From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=999, realm=pbx.itsp.com, algorithm=MD5, uri=sip:pbx.itsp.com, nonce=5cec66c0, response=6451967016fc38f896efeb7247523fe1, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060 Event: registration Content-Length: 0 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 123.183.86.231 : 5060 (NAT) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED];tag=as7d680d48 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;expires=120 Date: Fri, 04 May 2007 19:27:58 GMT ontent-Length: 0 -- SIP read from 123.183.86.231:5060: OPTIONS sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 19:38:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (12 headers 0 lines) --- Looking for s in entryunauthorized (domain pbx.itsp.com) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;received=123.183.86.231;rport=506 0 From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com;tag=as51d476cd Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:74.110.57.25 Accept: application/sdp Content-Length: 0 ___ --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] SIP REGISTRATION TIME OUT
Hi, now i can log in ok on my xlite, somebody calls me and everythink its okey. i hear and the caller hear. (the pc with the xlite have DMZ). But now i close xlite and put the same extension on a grandstream 286 (dont have DMZ). When somebody calls me the caller can hear me. but i cant hear! whats the problem? with other providers i can talk using my grandstream 286 without give it dmz or changing the configuration on my router. i hopes somebody can help me! 2007/4/14, dave cantera [EMAIL PROTECTED]: hello, I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I loaded 1.4 over *NOW because the gui regenerates files that, well, don't seem to work very well. it seems to me the gui creates the users.conf file, and then a script creates or uses the users.conf to create the dialplan... here is the users.conf file from *NOW... as you can see, this file does not conform to either sip.conf or extensions.conf, so that is my reasoning that it is source for some other generator... daveC ;! ;! Automatically generated configuration file ;! Filename: users.conf (/etc/asterisk/users.conf) ;! Generator: Manager ;! Creation Date: Sun Jan 21 15:41:42 2007 ;! [general] ; ; Full name of a user ; fullname = New User ; ; Starting point of allocation of extensions ; userbase = 6000 ; ; Create voicemail mailbox and use use macro-stdexten ; hasvoicemail = yes ; ; Create SIP Peer ; hassip = yes ; ; Create IAX friend ; hasiax = yes ; ; Create H.323 friend ; ;hash323 = yes ; ; Create manager entry ; hasmanager = no ; ; Remaining options are not specific to users.conf entries but are general. ; callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 host = dynamic localextenlength = 4 ;[6000] ;fullname = Joe User ;email = [EMAIL PROTECTED] ;secret = 1234 ;zapchan = 1 ;hasvoicemail = yes ;hassip = yes ;hasiax = no ;hash323 = no ;hasmanager = no ;callwaiting = no ;context = international Nicholas Campion wrote: The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 http://192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.conf in the general section. Give that a try and see what your result is. Nick On 4/13/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: mmm are you sure that asterisk-gui generate it on the sip.conf file? cause i see a new file called users.conf, and i can see the sip users on it. Anybody uses asterisk now and can check it please?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 08:34 PM -- 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] SIP REGISTRATION TIME OUT
hi, to get it work i change under sip.conf nat: route Allow RTP reinvite:update with that i can hear, without dmz... but... why? 2007/4/19, Manolet Gmail [EMAIL PROTECTED]: Hi, now i can log in ok on my xlite, somebody calls me and everythink its okey. i hear and the caller hear. (the pc with the xlite have DMZ). But now i close xlite and put the same extension on a grandstream 286 (dont have DMZ). When somebody calls me the caller can hear me. but i cant hear! whats the problem? with other providers i can talk using my grandstream 286 without give it dmz or changing the configuration on my router. i hopes somebody can help me! 2007/4/14, dave cantera [EMAIL PROTECTED]: hello, I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I loaded 1.4 over *NOW because the gui regenerates files that, well, don't seem to work very well. it seems to me the gui creates the users.conf file, and then a script creates or uses the users.conf to create the dialplan... here is the users.conf file from *NOW... as you can see, this file does not conform to either sip.conf or extensions.conf, so that is my reasoning that it is source for some other generator... daveC ;! ;! Automatically generated configuration file ;! Filename: users.conf (/etc/asterisk/users.conf) ;! Generator: Manager ;! Creation Date: Sun Jan 21 15:41:42 2007 ;! [general] ; ; Full name of a user ; fullname = New User ; ; Starting point of allocation of extensions ; userbase = 6000 ; ; Create voicemail mailbox and use use macro-stdexten ; hasvoicemail = yes ; ; Create SIP Peer ; hassip = yes ; ; Create IAX friend ; hasiax = yes ; ; Create H.323 friend ; ;hash323 = yes ; ; Create manager entry ; hasmanager = no ; ; Remaining options are not specific to users.conf entries but are general. ; callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 host = dynamic localextenlength = 4 ;[6000] ;fullname = Joe User ;email = [EMAIL PROTECTED] ;secret = 1234 ;zapchan = 1 ;hasvoicemail = yes ;hassip = yes ;hasiax = no ;hash323 = no ;hasmanager = no ;callwaiting = no ;context = international Nicholas Campion wrote: The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 http://192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.conf in the general section. Give that a try and see what your result is. Nick On 4/13/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: mmm are you sure that asterisk-gui generate it on the sip.conf file? cause i see a new file called users.conf, and i can see the sip users on it. Anybody uses asterisk now and can check it please?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 08:34 PM -- 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
Re: [asterisk-users] SIP REGISTRATION TIME OUT
hello, I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I loaded 1.4 over *NOW because the gui regenerates files that, well, don't seem to work very well. it seems to me the gui creates the users.conf file, and then a script creates or uses the users.conf to create the dialplan... here is the users.conf file from *NOW... as you can see, this file does not conform to either sip.conf or extensions.conf, so that is my reasoning that it is source for some other generator... daveC ;! ;! Automatically generated configuration file ;! Filename: users.conf (/etc/asterisk/users.conf) ;! Generator: Manager ;! Creation Date: Sun Jan 21 15:41:42 2007 ;! [general] ; ; Full name of a user ; fullname = New User ; ; Starting point of allocation of extensions ; userbase = 6000 ; ; Create voicemail mailbox and use use macro-stdexten ; hasvoicemail = yes ; ; Create SIP Peer ; hassip = yes ; ; Create IAX friend ; hasiax = yes ; ; Create H.323 friend ; ;hash323 = yes ; ; Create manager entry ; hasmanager = no ; ; Remaining options are not specific to users.conf entries but are general. ; callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 host = dynamic localextenlength = 4 ;[6000] ;fullname = Joe User ;email = [EMAIL PROTECTED] ;secret = 1234 ;zapchan = 1 ;hasvoicemail = yes ;hassip = yes ;hasiax = no ;hash323 = no ;hasmanager = no ;callwaiting = no ;context = international Nicholas Campion wrote: The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100 http://192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 http://192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.conf in the general section. Give that a try and see what your result is. Nick On 4/13/07, *Alex Balashov* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: mmm are you sure that asterisk-gui generate it on the sip.conf file? cause i see a new file called users.conf, and i can see the sip users on it. Anybody uses asterisk now and can check it please?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.0.0/752 - Release Date: 04/08/2007 08:34 PM -- 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] SIP REGISTRATION TIME OUT
hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default extension with SIP. is 600 and password 1234 so now i download xlite y configure it on the next way: user: 600 pass: 1234 auth user: 600 domain: pbxa.com nothing appers on the CLI, and after a 30 seconds i recieve a message on the xlite: Registration Error: 408- Request Timeout. (ping pbxa.com works fine), and btw, if i try with a user that doesnt exist (for example 601) on xlite i receive this on CLI: *CLI [Apr 13 11:32:02] NOTICE[12896]: chan_sip.c:14530 handle_request_register: Registration from '601sip:[EMAIL PROTECTED]' failed for '200.118.190.39' - No matching peer found i really dont get why i can register my SIP softphone, i try uninstalling and installing asterisk about 3 times and always is the same any ideas...? thanks you in advanced... ___ --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] SIP REGISTRATION TIME OUT
Hi Manolet, Can you provide your sip.conf? Thanks! -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP REGISTRATION TIME OUT
of course, download it from here: http://contelecltda.com/sip.conf but i dont edit the sip.conf, is the default make samples sip.conf file. i just use the asterisk gui interface to add the user... 2007/4/13, Alex Balashov [EMAIL PROTECTED]: Hi Manolet, Can you provide your sip.conf? Thanks! -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP REGISTRATION TIME OUT
On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: of course, download it from here: http://contelecltda.com/sip.conf but i dont edit the sip.conf, is the default make samples sip.conf file. i just use the asterisk gui interface to add the user... Well, then my conjecture would be that the GUI interface is broken, because there are no definitions for that or any other peer in there, nor hooks to include any other files generated by the GUI interface that might conceivably have them. Someone else have more insights? -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP REGISTRATION TIME OUT
mmm are you sure that asterisk-gui generate it on the sip.conf file? cause i see a new file called users.conf, and i can see the sip users on it. Anybody uses asterisk now and can check it please?? 2007/4/13, Alex Balashov [EMAIL PROTECTED]: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: of course, download it from here: http://contelecltda.com/sip.conf but i dont edit the sip.conf, is the default make samples sip.conf file. i just use the asterisk gui interface to add the user... Well, then my conjecture would be that the GUI interface is broken, because there are no definitions for that or any other peer in there, nor hooks to include any other files generated by the GUI interface that might conceivably have them. Someone else have more insights? -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP REGISTRATION TIME OUT
On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: mmm are you sure that asterisk-gui generate it on the sip.conf file? cause i see a new file called users.conf, and i can see the sip users on it. Anybody uses asterisk now and can check it please?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP REGISTRATION TIME OUT
The quick way to check if a user is defined is to go to the asterisk console and type sip show users which will list all the defined users and passwords. You say that it isn't a networking issue, but the fact that you are behind a NAT (your local ip is 192.168.0.100) is causing the problem (i think). All of your packets are reaching the server, but when it tries to respond it is sending the packets to 192.168.0.100 which is (obviously) not what you want to happen. The solution to this (typically) is to add NAT=yes to sip.confin the general section. Give that a try and see what your result is. Nick On 4/13/07, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, 13 Apr 2007, Manolet Gmail said something to this effect: mmm are you sure that asterisk-gui generate it on the sip.conf file? cause i see a new file called users.conf, and i can see the sip users on it. Anybody uses asterisk now and can check it please?? Hmm. I use 1.4.x here and installed the stock config file samples bundle, and there's no trace of users.conf. But then again, I have never used any GUI configurator, so I'm not in the best position to know what sort of structure and metadata it generates. -- Alex Balashov [EMAIL PROTECTED] ___ --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
[asterisk-users] SIP registration
When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.3.2' - Not a local SIP domain In sip.conf I have this for my global settings: [general] context=from-sip; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=actarg.com; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet autodomain=yes ; Turn this on to have Asterisk add local host ; name and local IP to domain list. ; and multiline formatted headers for strict localnet=192.168.2.0/23 qualify=no And this for my local settings: [201] type=friend; Friends place calls and receive calls context=from-sip ; Context for incoming calls from this user secret=asteriskpassword host=dynamic ; This peer register with us callerid=John Doe 201 disallow=all allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! progressinband=no ; Polycom phones don't work properly with never dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info nat=no ; there is not NAT between phone and Asterisk canreinvite=no ; disallow RTP voice traffic to bypass Asterisk Is there a better way to do this? Am I missing something obvious? Nathan Bell IT Engineer Action Target, Inc. ___ --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] SIP registration
Hi Nathan - I just saw this post about having trouble registering your phone ;-) When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.3.2' - Not a local SIP domain sip.conf autodomain=yes localnet=192.168.2.0/23 You might try expanding the scope of your localnet. Maybe this would work: localnet=192.168.0.0/255.255.0.0 Also, it seems like it should be covered by autodomain, but you might try explicitly adding: domain=192.168.3.2 - Noah ___ --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] SIP registration
That doesn't seem to make any difference. I still get the Not a local SIP domain and I get this from the CLI: ast*CLI sip show peers Name/username HostDyn Nat ACL Port Status 202(Unspecified)D 0Unmonitored 201(Unspecified)D 0Unmonitored 2 sip peers [2 online , 0 offline] ast*CLI sip show users Username Secret Accountcode Def.Context ACL NAT 202 *** from-sip No RFC3581 201 *** from-sip No RFC3581 Noah Miller wrote: Hi Nathan - I just saw this post about having trouble registering your phone ;-) When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.3.2' - Not a local SIP domain sip.conf autodomain=yes localnet=192.168.2.0/23 You might try expanding the scope of your localnet. Maybe this would work: localnet=192.168.0.0/255.255.0.0 Also, it seems like it should be covered by autodomain, but you might try explicitly adding: domain=192.168.3.2 - Noah ___ --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] SIP registration
The problem was on the polycom provisioning setup. In my dhcp settings I wasn't giving it the correct domain-name-servers option. I changed that and I changed the phones to use [EMAIL PROTECTED] instead of [EMAIL PROTECTED] and that seems to have taken care of it. Thanks for the help. Nathan Bell IT Engineer Du Jour Noah Miller wrote: Hi Nathan - I just saw this post about having trouble registering your phone ;-) When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.3.2' - Not a local SIP domain sip.conf autodomain=yes localnet=192.168.2.0/23 You might try expanding the scope of your localnet. Maybe this would work: localnet=192.168.0.0/255.255.0.0 Also, it seems like it should be covered by autodomain, but you might try explicitly adding: domain=192.168.3.2 - Noah ___ --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] SIP registration problem w/ SBC
Thanks Andrew, I see the resolved bug report. I'll get the patch fix. Sorry for the unnecessary mail. -Tom On 1/20/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: http://www.google.com/search?q=423+%22Interval+Too+Brief%22start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official Hint: Who develops Asterisk? On 1/20/07, Thomas Madler [EMAIL PROTECTED] wrote: Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED] ' timed out, trying again (Attempt #9) Is there something I can tweak on my end to fix this? TIA, -Tom ___ --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 ___ --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 registration problem w/ SBC
Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED] ' timed out, trying again (Attempt #9) Is there something I can tweak on my end to fix this? TIA, -Tom ___ --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] SIP registration problem w/ SBC
http://www.google.com/search?q=423+%22Interval+Too+Brief%22start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official Hint: Who develops Asterisk? On 1/20/07, Thomas Madler [EMAIL PROTECTED] wrote: Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED] ' timed out, trying again (Attempt #9) Is there something I can tweak on my end to fix this? TIA, -Tom ___ --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
[asterisk-users] SIP Registration conundrum
I have a customer on one of my Asterisk boxes that wants a small number of DIDs in Hong Kong. Referring to voip-info.org, we found the provider HKBN and their 2b service at www.2b.com.hk. Following the information at http://www.voip-info.org/wiki/index.php?page=asterisk+settings+HKBN+2b we successfully set up one DID, with Asterisk configured to receive incoming calls from it, as follows (in sip.conf): [general] register = 3999hk:[EMAIL PROTECTED]:5060/3999 ;--- Hong Kong Broadband incoming. Section name must match register statements. [s2hkbntel.net] type=user context=from-hkbn canreinvite=no insecure=very disallow=all allow=alaw dtmfmode=rfc2833 nat=no And in /etc/hosts we have the entries: #203.80.89.135 s2hkbntel.net s21.hkbntel.net 203.80.89.139 s2hkbntel.net s22.hkbntel.net As I said, this works fine. We registered a second number and added a register statement, which also worked fine: register = 3888hk:[EMAIL PROTECTED]:5060/3888 The problem came when we registered a third number. I added another register statement in the same way, but when Asterisk tried to register this number, it received a 301 Moved Permanently response, indicating that this third account was on the s21 proxy instead of the s22 one. This system has been in production for over a year, and is running the v1-0 CVS from April 2005. The first question is: does either the 1.2 or trunk version of Asterisk support 301 redirection to automatically re-attempt registration at the specified new IP address? The second question is: how can I set up my sip.conf to register these numbers with different proxies? I've tried using different proxy names and type=friend sections: register = 3999hk:[EMAIL PROTECTED]:5060/3999 register = 3777hk:[EMAIL PROTECTED]:5060/3777 [hkbn1] type=friend host=s22.hkbntel.net fromdomain=s2hkbntel.net context=from-hkbn canreinvite=no insecure=very disallow=all allow=alaw dtmfmode=rfc2833 nat=no [hkbn2] type=friend host=s21.hkbntel.net fromdomain=s2hkbntel.net context=from-hkbn canreinvite=no insecure=very disallow=all allow=alaw dtmfmode=rfc2833 nat=no However, this doesn't work. The SIP REGISTER packet looks like this: REGISTER sip:hkbn2 SIP/2.0 Via: SIP/2.0/UDP 194.nn.nnn.n:5060;branch=z9hG4bK0d1223ab From: sip:[EMAIL PROTECTED];tag=as1ec60bd6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 And the response that comes back is: SIP/2.0 403 Registrations to foreigndomains are forbidden So it evidently doesn't like either the hkbn2 in the REGISTER line or the s21.hkbntel.net in the From and/or To header. Originally, when one number was working, all those values were s2hkbntel.net. If you're still with me, thanks! I would appreciate any advice from either a SIP expert or someone who has successfully done this with HKBN. I can update to a newer Asterisk if necessary, but only if it will help with this issue. Thanks in advance, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 registration fails with 404
Can anyone give me any direction as to why I'm getting a 404 during the registration process. Sip Debug is: -- SIP read from 192.168.99.110:5060: REGISTER sip:asterisk1.rightsolve.com SIP/2.0 Via: SIP/2.0/UDP 192.168.99.110:5060;branch=123456789 To: sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=12345 CSeq: 1 REGISTER Call-ID: f97f33fb6c6a82c2efc0a16258e93d6b Max-Forwards: 70 User-Agent: VCS Contact: sip:[EMAIL PROTECTED] Expires: 600 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.99.110 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.99.110:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.99.110:5060;branch=123456789;received=192.168.99.110 From: sip:[EMAIL PROTECTED];tag=12345 To: sip:[EMAIL PROTECTED];tag=as5106c249 Call-ID: f97f33fb6c6a82c2efc0a16258e93d6b CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 TIA, routerguy This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --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 registration on Sipura 841
Hi, I'm a user of [EMAIL PROTECTED] for a couple of months now and been playing around with it for a while. I'm facing a strange situation which i am not able to solve. I have my * server and a SIPURA 841 phone both behind my router at home (No NAT between them). My * server is registered to 192.168.2.XXX and my phone is at 192.168.2.XXX with a port 5060. Initially i see the registration fine and everything works well for a minute. Just after a while i see that the phone registration expires and its take an IP 217.195.32.11 with port 5385. I am not able to identify this issue. As a result i am not able to receive calls on this device but i am able to call out. Please help Thanks in advance... Dan ___ --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 registration question
I am a newbie and am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 registration question
I am a newbie and am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 Registration Problem
I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server: Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip:[EMAIL PROTECTED];tag=12e8dd0080754148To: sip:[EMAIL PROTECTED];tag=as2383b1dfCall-ID: [EMAIL PROTECTED]CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290Content-Length: 0 After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] SIP Registration from Verizon DSL
I have a client who is unable to register her SJPhone on my Asterisk server. She is using a Westell DSL router connected to Verizon. Others in her group, using cable modems, are able to register. The group is located in the Dallas area. Is Verizon still blocking SIP registrations? Is there something about the Westell that needs to be changed? From what the client says, outbound traffic is unlimited. In sip.conf, I have nat=yes and qualify=yes. Thanks -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Sip registration Failure
Hi List, I am very new to asterisk. I downloaded asterisk from CVS head yesterday and compiled it in Redhat linux 9. I created a sip account for testing and configured it in the Firefly. While Firefly try to connect to the asterisk server i am getting an error as below and failing the registration. Oct 1 08:00:57 NOTICE[23415]: chan_sip.c:10646 handle_request_register: Registration from '200 sip:[EMAIL PROTECTED]' failed for '192.168.10.200' - Not a local SIP domain My sip configuration is as below, [general] context=default ; Default context for incoming calls bindport=5060 bindaddr=0.0.0.0 [test] type=peer secret=200 username=200 host=dynamic nat=no #disallow=all allow=all context=default Please help me to find out the problem in my configuration. Thanks in advance Rgds Anil ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] SIP registration issues
Hello. Is there any know issue with Asterisk 1.0.9 concerning intermittent SIP registration issues. My SIP hard phone (aastra 9133i) and soft phone (xlite) keep losing registration so calls to them go direct to VM although calling to other phones from them works fine. The logs show 'Transmitting (no NAT): SIP/2.0 403 Forbidden' which doesn't occur when they miraculously start working/registering. Asterisk seems to lose the user. Sep 9 11:47:36 VERBOSE[2444]: 12 headers, 0 lines Sep 9 11:47:36 VERBOSE[2444]: Using latest request as basis request Sep 9 11:47:36 VERBOSE[2444]: Sending to 192.168.1.100 : 5060 (non-NAT) Sep 9 11:47:36 VERBOSE[2444]: Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK289a5fe76 From: Martin sip:[EMAIL PROTECTED]:5060;tag=d6d383eca9b6910 To: Martin sip:[EMAIL PROTECTED]:5060;tag=as3c7c47f1 Call-ID: [EMAIL PROTECTED] CSeq: 54943697 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.1.100:5060 Sep 9 11:47:36 NOTICE[2444]: Registration from 'Martin sip:[EMAIL PROTECTED]:5060' failed for '192.168.1.100' Sep 9 11:47:36 VERBOSE[2444]: Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sep 9 11:47:36 VERBOSE[2444]: Sip read: REGISTER sip:192.168.1.50:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKd88070866 Max-Forwards: 70 Content-Length: 0 To: No User sip:[EMAIL PROTECTED]:5060 From: No User sip:[EMAIL PROTECTED]:5060;tag=0e8bc4f3c760bc2 Call-ID: [EMAIL PROTECTED] CSeq: 535959059 REGISTER Contact: No User sip:[EMAIL PROTECTED] Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 But then, some period of time later, they will start working at random times with no changes. Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] SIP Registration resets
Hi, We get this problem with some of our overseas SIP clients that they get un-registered from our Asterisk server, but still able to make calls. After sometime they get registered again. What can be causing this problem, any ideas? Zeeshan ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] SIP Registration failure
Hi list, I'm in central-europe and signed yesterday a broadvoice account. My Asterisk box is CVS 2005-08-25. Problem I face is: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 2) then Registration for '[EMAIL PROTECTED]' timed out and finaly Giving up forever to register '[EMAIL PROTECTED]' If I do keewi*CLI sip show peers Name/username Host Dyn Nat ACL Mask Port Status broadvoice/phonenumber 147.135.20.128 255.255.255.255 5060 OK (213 ms) keewi*CLI sip show registry Host Username Refresh State sip.broadvoice.com:5060 phonenumber 120 Failed In log files with sip debug I found Aug 27 16:38:09 VERBOSE[10641] logger.c: -- SIP read from 147.135.0.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 83.30.64.252:5060;branch=z9hG4bK70ac947c From: asterisk sip:[EMAIL PROTECTED];tag=as0de7b82c To: sip:sip.broadvoice.com;tag=SD384v999- Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Accept: application/sdp,application/broadsoft,text/plain Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE,NOTIFY,UPDATE Supported: 100rel Content-Length: 0 I'm registered with few others suppliers, SIP and IAX2 without any problem. Have someone an idea on what's happend? My ping time is 140 ms. Thanks for any hint. -- Daniel ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.
Steve Gladden wrote: You also want to look at the registertimeout and registerattempts Yes!!!, thank you VERY much this is what I needed. Where are these options documented at? I'm guessing the source code? Or is there a better place to find this stuff? A search on the wiki for registertimeout or registerattempts reveals absolutely nothing. I had been searching ealier for things like SIP register timeout and Giving up forever all to no avail. You should always check configs/sip.conf.sample in your source code directory. We update docs/ and configs/ very often. We recently updated the behaviour on authentication for INVITEs as well in CVS head, the base for 1.2. We will now give up if we can't authenticate, so the call goes back to the dialplan with CONGESTION instead of trying forever and ever. /Olle --- Astricon 2005 - wanna speak? Check http://www.astricon.net/2005/speakers Looking for call center, business and service providers business cases! ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.
On Wed, 24 Aug 2005, Steve Gladden wrote: I'm looking for some help in how to keep asterisk from doing this. If we loose Internet or routing to our upstream provider even for only a few short minutes asterisk quickly gives up never tries again. I have to do a manual reload to get it to register with my sip provider(s) again before incoming calls are accepted. This is really bad as it causes us to loose the ability to get incoming calls now then. Not at all what we want in a phone system. Won't you just start by updating your Asterisk IIRC, we patched a bug a couple of weeks back. If it still times out too quick, drop another line and we'll look further. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.
I updated 2 weeks ago and am due to update again... So Yes I will update It seems that the giving up forever feature is by design, As I had seen a post about it awhile back... But I would rather not have asterisk give up (forever) if it can't see a sip server. I feel retries should certainly back off in fact back way off like to once per some configurable time figure But not give up forever! In a single (non-redundant) phone system one wants it to come back and register back in unattended even if the Internet were down for several hours. :-) Actually I just needed the two settings that were mentioned previously... Not sure if the mentioned bug was of our concern, as my problem was not just with the fact that it timed out fast, but the fact that it could time out period and never try to re-register. I also would like to know where I could have found documentation of those two settings (registertimeout or registerattempts) As I had not been able to find those on my own or in the wiki. Thanks! Steve ) On Wed, 24 Aug 2005, Steve Gladden wrote: I'm looking for some help in how to keep asterisk from doing this. If we loose Internet or routing to our upstream provider even for only a few short minutes asterisk quickly gives up never tries again. I have to do a manual reload to get it to register with my sip provider(s) again before incoming calls are accepted. This is really bad as it causes us to loose the ability to get incoming calls now then. Not at all what we want in a phone system. Won't you just start by updating your Asterisk IIRC, we patched a bug a couple of weeks back. If it still times out too quick, drop another line and we'll look further. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] SIP Registration --Giving up forever after very short network outage.
I'm looking for some help in how to keep asterisk from doing this. If we loose Internet or routing to our upstream provider even for only a few short minutes asterisk quickly gives up never tries again. I have to do a manual reload to get it to register with my sip provider(s) again before incoming calls are accepted. This is really bad as it causes us to loose the ability to get incoming calls now then. Not at all what we want in a phone system. How can I get asterisk to stop giving up so soon or better yet not give up at all... this is after all a phone system... I would really like it to register back in if the Internet goes down then comes back up 10 minutes later! I'm running CVS-HEAD (about two weeks old) Aug 24 19:08:13 NOTICE[7124]: chan_sip.c:4701 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #8) Aug 24 19:08:16 NOTICE[7124]: chan_sip.c:4701 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #11) Aug 24 19:08:16 NOTICE[7124]: chan_sip.c:4719 sip_reg_timeout:-- Giving up forever trying to register '[EMAIL PROTECTED]' Thanks for your help !!! Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.
Steve Gladden wrote: I'm looking for some help in how to keep asterisk from doing this. If we loose Internet or routing to our upstream provider even for only a few short minutes asterisk quickly gives up never tries again. I have to do a manual reload to get it to register with my sip provider(s) again before incoming calls are accepted. Try using IP addresses instead of hostnames in sip.conf. Asterisk's DNS support is supposed to be improved in CVS-HEAD, but you should still try it. However, using an IP address instread of a hostname in your host= line could have issues with some ways a provider might do failover and load balancing. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.
On 8/24/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Try using IP addresses instead of hostnames in sip.conf. Asterisk's DNS support is supposed to be improved in CVS-HEAD, but you should still try it. However, using an IP address instread of a hostname in your host= line could have issues with some ways a provider might do failover and load balancing. You also want to look at the registertimeout and registerattempts options for your sip.conf. I had lots of problem staying registered with various providers, so now I'm running with registerattempts=0, IOW try forever to (re-)register. In conjunction with the registertimeout you have some control over how often you retry. (IIRC, both options are CVS-HEAD only, not available in stable. But so is the Giving up forever error. At least I think that's the case.) -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be lamitated! ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.
You also want to look at the registertimeout and registerattempts Yes!!!, thank you VERY much this is what I needed. Where are these options documented at? I'm guessing the source code? Or is there a better place to find this stuff? A search on the wiki for registertimeout or registerattempts reveals absolutely nothing. I had been searching ealier for things like SIP register timeout and Giving up forever all to no avail. Steve On 8/24/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Try using IP addresses instead of hostnames in sip.conf. Asterisk's DNS support is supposed to be improved in CVS-HEAD, but you should still try it. However, using an IP address instread of a hostname in your host= line could have issues with some ways a provider might do failover and load balancing. You also want to look at the registertimeout and registerattempts options for your sip.conf. I had lots of problem staying registered with various providers, so now I'm running with registerattempts=0, IOW try forever to (re-)register. In conjunction with the registertimeout you have some control over how often you retry. (IIRC, both options are CVS-HEAD only, not available in stable. But so is the Giving up forever error. At least I think that's the case.) -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be lamitated! ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Sip registration question
Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they think packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that from a phone this works fine, and that our config must be at issue. The claim is that Asterisk isn't doing MD5 authentication right, and since I'm not an expert with SIP MD5 auth in asterisk, may be true. Right now, I'm trying to get the registration happening. On a test server, we've been able to put through a call w/o registration, so it seems some of this can be compatible. I'm wondering if I can use md5secret with a register = statement. The current busted config: [general] ;register = userid:pass:[EMAIL PROTECTED]:5069 [myipsolution] type=friend authuser=acctid username=userid secret=pass md5secret=XXXMD5HASH of userid:asterisk:pass X nat=yes host=voipprovider.com port=5069 insecure=very canreinvite=no The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' The password is right, as given and verified by the provider. Any suggestions would be great. Thanks, J. ___ 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
Re: [Asterisk-Users] Sip registration question
On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they think packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that from a phone this works fine, and that our config must be at issue. The claim is that Asterisk isn't doing MD5 authentication right, and since I'm not an expert with SIP MD5 auth in asterisk, may be true. Right now, I'm trying to get the registration happening. On a test server, we've been able to put through a call w/o registration, so it seems some of this can be compatible. I'm wondering if I can use md5secret with a register = statement. The current busted config: [general] ;register = userid:pass:[EMAIL PROTECTED]:5069 [myipsolution] type=friend authuser=acctid username=userid secret=pass md5secret=XXXMD5HASH of userid:asterisk:pass X nat=yes host=voipprovider.com port=5069 insecure=very canreinvite=no The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' The password is right, as given and verified by the provider. Any suggestions would be great. Hi, Did you try to put the md5 encoded password in your register= line ? -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ 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
Re: [Asterisk-Users] Sip registration question
Hi, Quoting Michiel van Baak [EMAIL PROTECTED]: On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' Did you try to put the md5 encoded password in your register= line ? I didn't before (I wasn't sure that was a valid syntax) ... but I have tried now, same error. Is there something to tell asterisk to try an MD5 auth, either in the password or on the registration line? Thanks for your quick response. J. ___ 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
Re: [Asterisk-Users] Sip registration question
On 17:01, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: Hi, Quoting Michiel van Baak [EMAIL PROTECTED]: On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' Did you try to put the md5 encoded password in your register= line ? I didn't before (I wasn't sure that was a valid syntax) ... but I have tried now, same error. Is there something to tell asterisk to try an MD5 auth, either in the password or on the registration line? Thanks for your quick response. J. Hi, I don't think it is possible to use md5auth on register= lines. Have a look at: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf The one line that makes me think it is impossible is right below the Asterisk as a SIP client examples: Agreed, it's not very good to have a lot of cleartext passwords in this text file, but that's how it works now. If you find out I'm wrong, please send me or the list a reply -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ 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
[Asterisk-Users] SIP registration fails with realtime
I have set up realtime for Asterisk just as the instruction provide. Everything works, except it apearer that SIP devices do not regisert correctly. I can place a call from a SIP device, but not place a call to a SIP device. If a I use sip.conf everything seems to work.I have not posted all the configurations here because I'm just looking for aset of checks to follow. I noted that several other people on different lists have the same issue, but I have found no answer I understand.__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - FIXED :-)
I finally figured it out ... working with BT100 you need to make a little voodoo ritual first :-) ... so follow the steps --exactly-- if you have trouble This is my working configuration behind Linksys WRT54G router: - Upgrade firmware 1.0.5.23 - Reset BT100 to factory defaults - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - DTMF: SIP INFO - Reboot BTW ... this is exactly what I tried 100x before but without the exact order of steps. I think especially step #2 about resetting to factory defaults before you do any re-configuration is critical. Don't trust the web interface always start fresh. Strangely, I had no problems whenever I was behind any other router than Linksys ... didn't have to do all this voodoo stuff ... makes me uncomfortable since I feel like I'll plug the phones in tomorrow and I'll be back where I started. Maybe the secret was not changing my underwear in the morning :-) LOL On the Asterisk side it's just the usual: Nat = yes Qualify = yes Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - URI? I think I'm getting closer to figuring this out ... I just tried Linksys PAP2 and it registered just fine. I looked at the SIP packets captured by ethereal and I discovered that the real problem will probably be the uri in the authorization. For the working Linksys PAP2 and X-Lite I get: Authorization: DIGEST ... uri=sip:asterisk.mydomain.com ... For the BT100 which doesn't register (403 Forbidden) I get: Authorization: DIGEST ... uri=sip:wan-ip-of-the-router ... ... this kind of makes sense ... that looks like the wrong uri to send. So for some reason BT100 sends the wrong URI ... how can I fix this?? Again the weird thing is that if I plug in the BT100 behind any other router then Linksys WRT54G everything works fine. I'm trying my BT100 with the following config: - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - Nat travelsal: no - Local sip port: 5060 - Use NAT ip: no - Proxy require: no And in my sip.conf I have Nat=yes Qualify=yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:04 PM To: 'Pedro'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http
[Asterisk-Users] SIP registration behind Linksys WRT54G
Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ 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
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ 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 ___ 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
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
I have tried several, dlink doesn't seem to have the same issue and a more intelligent firewall is not having any problems. We are working with the Sipura 1001 and 2000 units on this issue. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri="sip:asterisk.mydomain.com", ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri="sip 66.x.x.166:5060", ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ 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 ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
I've got a 7960 behind a Linksys wireless box and its working just fine with nat=yes in the sip.conf. Has been for over a year. Not sure of the model though. Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. ___ 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
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
I have a whole Asterisk server behind a wtr54gs. We have SPA-2000's registering from the Internet into it with no problems. Actually, we don't have it at the moment but did for several months. Not sure if this helps any or just adds to the confusion. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 10:24 PM Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G I've got a 7960 behind a Linksys wireless box and its working just fine with nat=yes in the sip.conf. Has been for over a year. Not sure of the model though. Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. ___ 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 ___ 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
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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 ___ 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
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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 ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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 ___ 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 ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - URI?
I think I'm getting closer to figuring this out ... I just tried Linksys PAP2 and it registered just fine. I looked at the SIP packets captured by ethereal and I discovered that the real problem will probably be the uri in the authorization. For the working Linksys PAP2 and X-Lite I get: Authorization: DIGEST ... uri=sip:asterisk.mydomain.com ... For the BT100 which doesn't register (403 Forbidden) I get: Authorization: DIGEST ... uri=sip:wan-ip-of-the-router ... ... this kind of makes sense ... that looks like the wrong uri to send. So for some reason BT100 sends the wrong URI ... how can I fix this?? Again the weird thing is that if I plug in the BT100 behind any other router then Linksys WRT54G everything works fine. I'm trying my BT100 with the following config: - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - Nat travelsal: no - Local sip port: 5060 - Use NAT ip: no - Proxy require: no And in my sip.conf I have Nat=yes Qualify=yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:04 PM To: 'Pedro'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ 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 ___ 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 ___ 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 ___ 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 ___ 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 ___ 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
[Asterisk-Users] SIP registration fails
Title: SIP registration fails Hello List ;) I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions. First of all the relevant part of my sip.conf: cut sip.conf -- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to srvlookup=yes nat=yes localnet=192.168.11.0/255.255.255.0 externip=myexternaldyndnsname realm=myrealm context = from-sip ; Default for incoming calls insecure=very tos=0x18 dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw register = mysipid:mysippass@sip.web.de/mysipid [webde] type=friend username=mysipid secret=mysippass host=sip.web.de fromuser=mysipid fromdomain=sip.web.de nat=no canreinvite=no insecure=very qualify=400 dtmfmode=info cut sip.conf -- My questions on this are: a) why is SIP registration failing? b) how is mapping between register= and [webde] done? many 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
RE: [Asterisk-Users] SIP registration fails
Title: SIP registration fails You may better look at example sip.conf files you will be able to find on WIKI as there appears to be several incosnsistencies in your sip.conf. My suggestion is get rid off what you dont need and use only those what is barely essential. When you are using NAT Ip you need to have entries like: host=dynamic Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William MarksSent: Wednesday, April 13, 2005 10:57 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP registration fails Hello List ;) I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions. First of all the relevant part of my sip.conf: cut sip.conf -- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to srvlookup=yes nat=yes localnet=192.168.11.0/255.255.255.0 externip=myexternaldyndnsname realm=myrealm context = from-sip ; Default for incoming calls insecure=very tos=0x18 dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw register = mysipid:mysippass@sip.web.de/mysipid [webde] type=friend username=mysipid secret=mysippass host=sip.web.de fromuser=mysipid fromdomain=sip.web.de nat=no canreinvite=no insecure=very qualify=400 dtmfmode=info cut sip.conf -- My questions on this are: a) why is SIP registration failing? b) how is mapping between "register=" and [webde] done? many thanks. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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
AW: [Asterisk-Users] SIP registration fails
Title: AW: [Asterisk-Users] SIP registration fails Hi Seshu, that's where I started off. But most of them are not working (at least not for me). My desired setup (for now) is very simple: SIP provider(web.de) -- * -- 2 SIP phones But none of the examples explains how the register statement and the corresponding host-entry are linked to each other. Could you help? Will -Ursprüngliche Nachricht- Von: Kanuri, Seshu (Company IT) [mailto:[EMAIL PROTECTED]] Gesendet: Mittwoch, 13. April 2005 20:11 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] SIP registration fails You may better look at example sip.conf files you will be able to find on WIKI as there appears to be several incosnsistencies in your sip.conf. My suggestion is get rid off what you dont need and use only those what is barely essential. When you are using NAT Ip you need to have entries like: host=dynamic Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of William Marks Sent: Wednesday, April 13, 2005 10:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP registration fails Hello List ;) I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions. First of all the relevant part of my sip.conf: cut sip.conf -- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to srvlookup=yes nat=yes localnet=192.168.11.0/255.255.255.0 externip=myexternaldyndnsname realm=myrealm context = from-sip ; Default for incoming calls insecure=very tos=0x18 dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw register = mysipid:mysippass@sip.web.de/mysipid [webde] type=friend username=mysipid secret=mysippass host=sip.web.de fromuser=mysipid fromdomain=sip.web.de nat=no canreinvite=no insecure=very qualify=400 dtmfmode=info cut sip.conf -- My questions on this are: a) why is SIP registration failing? b) how is mapping between register= and [webde] done? many thanks. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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
Re: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
Hi I have a Zyxel P2002 (ATA) with this config. Registration works but i cant call inn. Outgoing works fine. Any clue? Thore - Original Message - From: Paul Dracevich [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 6:51 AM Subject: RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi ya I have also three of these phone, here is my entry in my sip.conf [4701721] type=friend username=4701721 secret=password721 host=dynamic canreinvite=no context=internal disallow=all allow=g729 dtmfmode=rfc2833 qualify=4 permit=0.0.0.0/0.0.0.0 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugur GUNCER Sent: Sunday, 3 April 2005 4:37 p.m. To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi all, I bougth zyxel wifi phone but i cant register when i want to register phone to asterisk i recieve These errors I spend 6 hours to fix regist problem but i cant find the solution [9875] type=friend username=9875 secret=5789 host=dynamic context=default callerid=Ugur Guncer 9875 canreinvite=no dtmfmode=rfc2833 nat=no Sip read: REGISTER sip:213.139.225.82:5060 SIP/2.0 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: sip:[EMAIL PROTECTED]:43956;transport=udp Expires: 300 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 85.99.110.143 : 43956 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 85.99.110.143:43956 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0f3403ce Content-Length: ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
You need to upgrade these phones to the latest firmware for it to work well with asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thore Sent: Sunday, April 03, 2005 3:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi I have a Zyxel P2002 (ATA) with this config. Registration works but i cant call inn. Outgoing works fine. Any clue? Thore - Original Message - From: Paul Dracevich [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 6:51 AM Subject: RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi ya I have also three of these phone, here is my entry in my sip.conf [4701721] type=friend username=4701721 secret=password721 host=dynamic canreinvite=no context=internal disallow=all allow=g729 dtmfmode=rfc2833 qualify=4 permit=0.0.0.0/0.0.0.0 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugur GUNCER Sent: Sunday, 3 April 2005 4:37 p.m. To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi all, I bougth zyxel wifi phone but i cant register when i want to register phone to asterisk i recieve These errors I spend 6 hours to fix regist problem but i cant find the solution [9875] type=friend username=9875 secret=5789 host=dynamic context=default callerid=Ugur Guncer 9875 canreinvite=no dtmfmode=rfc2833 nat=no Sip read: REGISTER sip:213.139.225.82:5060 SIP/2.0 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: sip:[EMAIL PROTECTED]:43956;transport=udp Expires: 300 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 85.99.110.143 : 43956 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 85.99.110.143:43956 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0f3403ce Content-Length: ___ 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 ___ 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 ___ 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 ___ 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
[Asterisk-Users] Sip registration Problems With Zyxel P2000W
Hi all, I bougth zyxel wifi phone but i cant register when i want to register phone to asterisk i recieve These errors I spend 6 hours to fix regist problem but i cant find the solution [9875] type=friend username=9875 secret=5789 host=dynamic context=default callerid=Ugur Guncer 9875 canreinvite=no dtmfmode=rfc2833 nat=no Sip read: REGISTER sip:213.139.225.82:5060 SIP/2.0 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: sip:[EMAIL PROTECTED]:43956;transport=udp Expires: 300 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 85.99.110.143 : 43956 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 85.99.110.143:43956 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0f3403ce Content-Length: ___ 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
RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
Hi ya I have also three of these phone, here is my entry in my sip.conf [4701721] type=friend username=4701721 secret=password721 host=dynamic canreinvite=no context=internal disallow=all allow=g729 dtmfmode=rfc2833 qualify=4 permit=0.0.0.0/0.0.0.0 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugur GUNCER Sent: Sunday, 3 April 2005 4:37 p.m. To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi all, I bougth zyxel wifi phone but i cant register when i want to register phone to asterisk i recieve These errors I spend 6 hours to fix regist problem but i cant find the solution [9875] type=friend username=9875 secret=5789 host=dynamic context=default callerid=Ugur Guncer 9875 canreinvite=no dtmfmode=rfc2833 nat=no Sip read: REGISTER sip:213.139.225.82:5060 SIP/2.0 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: sip:[EMAIL PROTECTED]:43956;transport=udp Expires: 300 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 85.99.110.143 : 43956 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 85.99.110.143:43956 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0f3403ce Content-Length: ___ 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 ___ 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
[Asterisk-Users] SIP registration problem
Hi, I am adding phones to my asterisk setup, until now i worked with some softphones, with no problem, I got some Grandstream BT100 phones, and see something strange in the log, the on the phone's screen, This is from the log : Found peer '122' Looking for 122 in default Transmitting (no NAT): SIP/2.0 404 Not Found This happends when the action is SUBSCRIBE , Now, this is a SIP client, defined in the sip.conf, as [122] context=default ... and also the exten is in the default context in the extension conf file, Right after the the peer seems to be registered, and the phone seems to work, but from time to time, i see 404 on the phone's display, and need to touch it to make it change (dial something, or just pick up and hangup) I couldnt find why this is happening, i searched, and found some with the same problem, but no solution, If you have any idea why this is happening, i will be glad to hear it. Thanks. Marco. ___ 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
