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
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
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
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
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
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
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
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
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:
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
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
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
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
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
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
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
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
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
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
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
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
Re: [Asterisk-Users] SIP registration problem
In the Grandstream setup, turn off subscribe to message waiting indication. ...or upgrade to CVS head, where I've fixed this problem with SUBSCRIBE. Best regards, /Olle ___ 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
I have this problem for 2 days and i dont understand
I am behind a nat
my sip.conf is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
disallow = all
allow= gsm
allow= ilbc
allow= ulaw
allow= alaw
;
;
localnet = 172.27.254.0/255.255.255.0 ; intern network ip address
;localmask = 255.255.255.0 ;
externip =193.49.116.12 ; my public ip address
;
maxexpirey=180
defaultexpirey=160
;
register = 560793:[EMAIL PROTECTED]/6002
;
[fwd]
type=friend
secret=mypasswd
username=fayafibun
host=fwd.pulver.com
fromdomain=fwd.pulver.com
insecure=very
context = from-sip
;
;
;
;
[bombaclaat]
callerid=(bombaclaat 6009)
type=friend
secret=mypasswd
host=dynamic
auth=md5
defaultip=172.27.254.14
context=internal
reinvite=no
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=all
mailbox=bombaclaat
qualify=1000
nat=yes
;
;
[6002]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
;context=internal
context = from-sip
mailbox=6002
;
[6000]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
context=internal
mailbox=6000
;
[bloodclaat]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
context=internal
mailbox=bloodclaat
;
;
my extension.conf
[general]
static=yes
writeprotect=no
[globals]
;
; The name to use on callerid
;
BOMBA=SIP/bombaclaat
OTRE=SIP/6002
FWDUSERID=560793
FWDUSERNAME=fayafibun
PHONE1=6002
PHONE1VM=voicemail(6002)
FWDEXTEND=6002
;EVRYONE=${BOMBA}${OTRE}
;
[internal]
;
; local extensions
;
exten = bombaclaat,1,Dial(SIP/bombaclaat,60) ; call SIP extension
bombaclaat for 60 seconds, if extension bombaclaat is called
exten = bombaclaat,2,Voicemail(ubombaclaat) ; if we cant connect
to bombaclaat or after seconds go to the unavail VM
exten = bombaclaat,102,Voicemail(bbombaclaat); if busy, go to the busy VM
exten = 6002,1,Dial(SIP/6002,60) ; call SIP extension
bombaclaat for 60 seconds, if extension bombaclaat is called
exten = 6002,2,Voicemail(u6002) ; if we cant connect
to bombaclaat or after seconds go to the unavail VM
exten = 6002,102,Voicemail(b6002); if busy, go to the busy VM
exten = bloodclaat,1,Dial(SIP/bloodclaat,60)
exten = bloodclaat,2,Voicemail(ubloodclaat)
exten = bloodclaat,103,Voicemail(bbloodclaat)
exten = 6000,1,Dial(SIP/6000,60)
exten = 6000,2,Voicemail(u6000)
exten = 6000,103,Voicemail(b6000)
exten = _[123456789],1,NoOp(callfor${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},40,tr)
exten = _[123456789],3,Congestion
exten = 1312605133,1,Dial(${FIPC}/${EXTEN:1},60) ; call SIP
extension bombaclaat for 60 seconds, if extensio$
exten = 1312605133,2,Voicemail(ubombaclaat) ; if we cant connect
to bombaclaat or after seconds go to t$
exten = bombaclaat,104,Voicemail(bbombaclaat);;
;
;appeler le 2500 de n importe kel phone pour contacter le voicemail system
exten = 2500,1,VoicemailMain
exten = 2500,2,Hangup
;
;
; Voicemail System
;
exten = 123,1,Answer
exten = 123,2,Playback(tt-weasels)
exten = 123,3,Voicemail(6002)
exten = 123,4,Hangup
;
;
;exten = ,1,VoiceMailMain(${CALLERIDNUM}) ; extension is
the VM system,
; go directly to callers VM
;exten = ,2,Hangup
;
;[outbound-internal]
;
; include local extensions
;
; include = internal
;
;
; include SIP accounts
;
; include = 6002
; include = bombaclaat
; include = 6000
; include = bloodclaat
[default]
;
; include from-sip for default. We dont use it, but it might be a good idea
;
;include = internal
;Extension Description
;101 Mark Spencer
;102 Wil Meadows
;0 Operator
include = from-sip
include = fwd-out
[fwd-out]
exten = _7.,1,SetCIDNum(${FWDUSERID})
exten = _7.,2,SetCIDName(${FWDUSERNAME})
exten = _7.,3,Dial(SIP/fwd-outgoin/${EXTEN:1})
exten = _7.,4,Playback(invalid)
exten = _7.,5,Hangup
[from-sip]
exten = ${FWDEXTEN},1,Dial(${PHONE1},30)
exten = ${FWDEXTEN},2,Voicemail(u${PHONE1VM})
exten = ${FWDEXTEN},3,Hangup
exten = ${FWDEXTEN},102,Voicemail(b${PHONE1VM})
exten = ${FWDEXTEN},103,Hangup
I have those errors
Jan 20 11:30:18 NOTICE[98310]: chan_sip.c:4053 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
Jan 20 11:30:24
Re: [Asterisk-Users] sip registration fails
I have tried uncommenting the section for xlite included in the sample configuration file sip.conf and I can't register. [xlite1] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend regexten=1234 ; When they register, create extension 1234 username=tito callerid=yo 5678 host=dynamic nat=yes ; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw AM Hello, AM I am trying to register in asterisk with a softphone (x-lite) and I am AM getting the following message: AM Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: AM Registration from 'tito sip:[EMAIL PROTECTED]' failed for AM '192.168.1.5' AM In the sip.conf file I have included the following. Does I need to AM include another change to allow the user to register? AM [phone1] AM type=friend AM host=dynamic AM defaultip=192.168.1.5 AM username=tito AM secret=tito AM dtmfmode=rfc2833 AM mailbox=1000 AM context=sip AM callerid=Tito 2124 AM I get the following message too and I don't know what does that means: AM Jan 19 13:26:41 WARNING[3343]: chan_sip.c:685 retrans_pkt: AM Maximum retries exceeded on call AM [EMAIL PROTECTED] for seqno 102 AM (Non-critical Request) AM ___ AM Asterisk-Users mailing list AM Asterisk-Users@lists.digium.com AM http://lists.digium.com/mailman/listinfo/asterisk-users AM To UNSUBSCRIBE or update options visit: AMhttp://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 fails
Alberto Martínez wrote: Hello, I am trying to register in asterisk with a softphone (x-lite) and I am getting the following message: Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: Registration from 'tito sip:[EMAIL PROTECTED]' failed for '192.168.1.5' Just a guess, but the ip's don't match up. [...] I get the following message too and I don't know what does that means: Jan 19 13:26:41 WARNING[3343]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) I'm getting this too. Using sip debug shows some sort of message notification attempt repeating itself for a sip client even though the client isn't online. The series of repeats ends with the error message that you are seeing. Dave CAUTION: This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of Winstone Pulp International Ltd. ___ 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/dialing problem.
On Nov 3, 2004, at 4:16 PM, Ben Greear wrote: Hello! I have a Grandstream and a Cisco SIP phone, and I'm trying to make a call between them. I added this to my sip.conf: ; Grandstream [1001] type=friend host=dynamic ; cisco phone [1002] type=friend host=dynamic First, what's in your extensions.conf? That controls the flow of calls once they get into the system. There should be a context that has extensions for 1001 and 1002, and sip.conf should direct calls into that extension via a 'context =' line. Running an Asterisk console in verbose mode (asterisk -vr will connect to a running server) provides a lot of useful debugging information. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration/dialing problem.
Scott Laird wrote: First, what's in your extensions.conf? That controls the flow of calls once they get into the system. There should be a context that has extensions for 1001 and 1002, and sip.conf should direct calls into that extension via a 'context =' line. Indeed, I had not changed the extensions.conf at all. After adding some (at least mostly correct) values, I was able to make calls between my sip phones, as well as between a soft-phone based on VOCAL and a SIP phone. So, I'm quite satisfied with it now, though I have barely started to scratch the surface of the feature set. Thanks, Ben -- Ben Greear [EMAIL PROTECTED] Candela Technologies Inc http://www.candelatech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration with public dynamic ip address
I set up my own STUN server and turned reinvite off. Lyle - Original Message - From: [EMAIL PROTECTED] To: '[EMAIL PROTECTED]' Sent: Tuesday, August 31, 2004 8:53 AM Subject: [Asterisk-Users] SIP registration with public dynamic ip address Hi, I'm trying to configure a natted budgetone phone to a asterisk server as described in wiki using port forwarding. I successfully make call from the client but it seems it does not register the client ip address and when I try to recall it is not reacheable. Asterisk can manage natted sip client with dynamic ip address ? Bye ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration issues
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell [EMAIL PROTECTED] wrote: Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I'm using * with an IX66 and no issues, with CVS head I suggest you have a configuration error somewhere it looks like the IX66 is trying to authorise the clients, and no * have you set the IX66 to forward all sip requests for your domain to * ? Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Registration seems to timeout
Hi. Thanks for tipping me off with the new firmware. I installed it and tested the codec. Has more delay but seems to be better quality than what I was using before. Anyways, that didn't fix the SIP Registration Failure that I am getting. Any ideas? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Richard Neese Sent: Wednesday, June 09, 2004 7:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Registration seems to timeout try changing your codec to ilbc and make sure that his gs has the latest flash to support it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration seems to timeout
try changing your codec to ilbc and make sure that his gs has the latest flash to support it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration Problem
Brian Rathman wrote: I am using snom200 phones registering with Asterisk via SIP. I can see where the phone registers without a problem, and then when you try and make a call I get a proxy authentication required message on the phone and failed to authenticate user error in the Asterisk messages file. Then the next call you make from the phone goes through without a problem. Nothing changes between these two events, but it is almost like the phone is using two different passwords for the same account. Has anyone else seen a problem like this? I am using an Asterisk CVS version from early March, not sure if upgrading will help as well. Thanks, Brian Please don't start a new thread by replying to an exisiting post - threaded mailreaders list it as a reply to that post (even if you change the subject, as theading is done by messageid). You're also less likely to get a response due to the post being inside an existing converstion rather than as listed as a new topic. regards, Julien ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote: This is also closely related to Asterisk SIP's lack of proper [user section] authentication/recognition for incoming calls. We've seen a lot of posts here where new users have problems with this, but the real problem is usually not acknowledged. So tell me what's wrong with the user authentication/recognition ? I'm working on that part in chan_sip2 now. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote: Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or not, Asterisk doesn't do it correctly either. The host should respond with 200/OK if the call could succeed theoretically if it were an INVITE or else it should send a 404 or maybe a 487(? hmm, have to look) see the RFC for details. Interesting, didn't know that. Where in the RFC? I removed the qualify lines and sip reload [ed]. The extension still showed up as UNREACHABLE instead of UNMONITORED. I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? If a peer is registred at reload/sip reload, it will not change. You have to unload the sip module and reload it or restart asterisk to change the configuration of a registred, i.e. active, peer. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip Registration Problem
How will this effect a live system? No new calls? Or will it terminate exisiting calls? I'll have a chat with the vendor regarding the OPTIONS reply.. It certainly does sesem like it should reply with something.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Tuesday, May 25, 2004 1:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip Registration Problem Karl Brose wrote: Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or not, Asterisk doesn't do it correctly either. The host should respond with 200/OK if the call could succeed theoretically if it were an INVITE or else it should send a 404 or maybe a 487(? hmm, have to look) see the RFC for details. Interesting, didn't know that. Where in the RFC? I removed the qualify lines and sip reload [ed]. The extension still showed up as UNREACHABLE instead of UNMONITORED. I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? If a peer is registred at reload/sip reload, it will not change. You have to unload the sip module and reload it or restart asterisk to change the configuration of a registred, i.e. active, peer. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Registration Problem
I removed the qualify lines and sip reload [ed]. The extension still showed up as UNREACHABLE instead of UNMONITORED. I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? If a peer is registred at reload/sip reload, it will not change. You have to unload the sip module and reload it or restart asterisk to change the configuration of a registred, i.e. active, peer. /O Brett Nemeroff wrote: How will this effect a live system? No new calls? Or will it terminate exisiting calls? Unloading SIP module will terminate all SIP calls Restarting Asterisk will terminate all calls :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Registration Problem
RFC 3261 states: 11.2 Processing of OPTIONS Request The response to an OPTIONS is constructed using the standard rules for a SIP response as discussed in Section 8.2.6. The response code chosen MUST be the same that would have been chosen had the request been an INVITE. That is, a 200 (OK) would be returned if the UAS is ready to accept a call, a 486 (Busy Here) would be returned if the UAS is busy, etc. This allows an OPTIONS request to be used to determine the basic state of a UAS, which can be an indication of whether the UAS will accept an INVITE request. An OPTIONS request received within a dialog generates a 200 (OK) response that is identical to one constructed outside a dialog and does not have any impact on the dialog. This use of OPTIONS has limitations due to the differences in proxy handling of OPTIONS and INVITE requests. While a forked INVITE can result in multiple 200 (OK) responses being returned, a forked OPTIONS will only result in a single 200 (OK) response, since it is treated by proxies using the non-INVITE handling. See Section 16.7 for the normative details. If the response to an OPTIONS is generated by a proxy server, the proxy returns a 200 (OK), listing the capabilities of the server. The response does not contain a message body. Allow, Accept, Accept-Encoding, Accept-Language, and Supported header fields SHOULD be present in a 200 (OK) response to an OPTIONS request. If the response is generated by a proxy, the Allow header field SHOULD be omitted as it is ambiguous since a proxy is method agnostic. Contact header fields MAY be present in a 200 (OK) response and have the same semantics as in a 3xx response. That is, they may list a set of alternative names and methods of reaching the user. A Warning header field MAY be present. A message body MAY be sent, the type of which is determined by the Accept header field in the OPTIONS request (application/sdp is the default if the Accept header field is not present). If the types include one that can describe media capabilities, the UAS SHOULD include a body in the response for that purpose. Details on the construction of such a body in the case of application/sdp are described in [13]. Brett Nemeroff wrote: How will this effect a live system? No new calls? Or will it terminate exisiting calls? I'll have a chat with the vendor regarding the OPTIONS reply.. It certainly does sesem like it should reply with something.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Tuesday, May 25, 2004 1:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip Registration Problem Karl Brose wrote: Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or not, Asterisk doesn't do it correctly either. The host should respond with 200/OK if the call could succeed theoretically if it were an INVITE or else it should send a 404 or maybe a 487(? hmm, have to look) see the RFC for details. Interesting, didn't know that. Where in the RFC? I removed the qualify lines and sip reload [ed]. The extension still showed up as UNREACHABLE instead of UNMONITORED. I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? If a peer is registred at reload/sip reload, it will not change. You have to unload the sip module and reload it or restart asterisk to change the configuration of a registred, i.e. active, peer. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Registration Problem
for those who want to patch their SIP, here is a quck fix to make
Asterisk do a little better:
--- chan_sip.c 2004-05-16 01:33:06.0 -0400
+++ chan_sip.c_OPTIONS 2004-05-17 14:30:36.0 -0400
@@ -5916,6 +5916,7 @@
/* Initialize the context if it hasn't been already */
if (!strcasecmp(cmd, OPTIONS)) {
+ check_user(p, req, cmd, e, 0, sin, 0);
res = get_destination(p, req);
build_contact(p);
/* XXX Should we authenticate OPTIONS? XXX */
Olle E. Johansson wrote:
Karl Brose wrote:
Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
not, Asterisk doesn't do it correctly either.
The host should respond with 200/OK if the call could succeed
theoretically if it were an INVITE or else it should send a
404 or maybe a 487(? hmm, have to look) see the RFC for details.
Interesting, didn't know that. Where in the RFC?
___
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Re: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote: If the response to an OPTIONS is generated by a proxy server, the proxy returns a 200 (OK), listing the capabilities of the server. The response does not contain a message body. Allow, Accept, Accept-Encoding, Accept-Language, and Supported header fields SHOULD be present in a 200 (OK) response to an OPTIONS request. If the response is generated by a proxy, the Allow header field SHOULD be omitted as it is ambiguous since a proxy is method agnostic. Contact header fields MAY be present in a 200 (OK) response and have the same semantics as in a 3xx response. That is, they may list a set of alternative names and methods of reaching the user. A Warning header field MAY be present. This is what asterisk is doing, or? Please explain where and how you think Asterisk is not following the RFC, and I'll look into it. The other alternative would be to act as a UAS, but that may be confusing. Is any phone using this for checking if an URL is busy or not? In dialogue or out of dialogue? Just want to know if there's anything out there to test with. Thank you for looking this up. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Registration Problem
It's a bug in Asterisk. I believe it's still open also on the bugtracker. There are a few reported senarios with these kind of problems. Some of them where solved with the recent 'ast_gethostbyname' fix. Are you running a recent version? Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or not, Asterisk doesn't do it correctly either. The host should respond with 200/OK if the call could succeed theoretically if it were an INVITE or else it should send a 404 or maybe a 487(? hmm, have to look) see the RFC for details. Brett Nemeroff wrote: Hi All, I had an unusual problem today; I'm sure it's a configuration problem. I had 2 phones behind a nat device and I had qualify=300 in both extensions config. The device I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was ignoring the OPTIONS messages that * was sending, and thus * interpreted that as the extensions being down. I removed the qualify lines and sip reload [ed]. The extension still showed up as UNREACHABLE instead of UNMONITORED. I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? Thanks all, Brett ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration Errors
Hi! Registration only works if you have set host=dynamic for the client! In case of a static host registration makes no sense, anyway! The only purpose of registration is to tell the server at which IP address the phone can be found. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration Errors
Larry Keyes wrote: Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and outbound to an X100p card. Any ideas regarding the config file would be appreciated. -- Larry NOTICE[1125350192]: File chan_sip.c, Line 5297 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.162' NOTICE[1125350192]: File chan_sip.c, Line 3557 (register_verify): Peer '1001' isn't dynamic Read what it says. Peer '1001' is defined as a fixed IP address, not dynamic. So it is not allowed to register. The host= setting defines how we're going to contact the peer when we want to deliver a call to the phone. host=dynamic - Make the device register with asterisk so we know the current IP address host=ip address - No registration, we already know the IP address and the address doesn't change. For mobile devices, like soft phones on a laptop, use registration. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration Errors
Larry Keyes wrote: Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and outbound to an X100p card. I saw the same problem with both CVS version with SJPhone and X-lite. I do not own a Grandstream those your settings may be slightly different. To fix the problem I : - changed my SIP definition - stop and started asterisk (IMPORTANT: reload did not work) The [] part is really used as the username on the phone . I am not even sure the username= is used for anything !! Make sure as well you do not mix phone name and extension number. My phone is SIP/thomas, my extension is 1505 [thomas] type=friend host=dynamic dtmfmode=inband ; your dtmf mode may be right for your phone ... No idea. username=thomas secret=supersecret callerid=Thomas Mangin 1505 context=default mailbox=1505 ;auth=md5 ;reinvite=no ;canreinvite=no ;qualify=1000 ;defaultip=10.0.0.10 ;restrictcid=no *CLI sip show users Username Secret Authen Def.Context A/C thomas supersecret md5,plaintextdefault No Try this and if it work change one thing at the time and RESTART * as reload can cause some surprise. I can not recall if host=dynamic and defaultip are compatible but I think there are. Hope it helps, I only have few hours experience with * myself. Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip registration send out by asterisk
- Original Message - From: SW [EMAIL PROTECTED] To: [EMAIL PROTECTED] Digium. Com [EMAIL PROTECTED] Sent: Tuesday, December 16, 2003 1:47 PM Subject: [Asterisk-Users] sip registration send out by asterisk Hi friends, I've noticed that first register message sent by * always get rejected by the destination sip server. Then * sends a second registration message ( with Autherization section, and that get accepted by the destination host). Why is this ? Isnt there a way to tell * to send with Autothorization message the first attempt ? Asterisk sends this first 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipauth.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a From: sip:[EMAIL PROTECTED];tag=as3e96887d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 106 REGISTER User-Agent: Asterisk PBX Expires: 160 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 After 401 unautorized from iconnect asterisk sends this 8 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sipauth.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a From: sip:[EMAIL PROTECTED];tag=as3e96887d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=1510xxx, realm=deltathree.com, algorithm=MD5, uri=sip:sipauth.deltathree.com, nonce=3fdecbbf, response=49558c95bc3383bcbf76a26376e1614a Expires: 160 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 Cheers SW I could be wrong, but I believe there is a challenge token sent back with the Unauthorized message that is used to build the properly Authenticated request. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration Difficulties
Dave Cotton wrote: I have SIP registrations working correctly for FWD and Sipphone, but it is impossible to connect to Sipcall or Nikotel, I saw that someone on the list has problems with ICH. Does this imply that it will work even in a NAT environment? I have watched the list like a hawk for evidence of FWD working for machines placed behind NAT, but so far haven't seen that anyone could actually get it going. If so, that would be waay good news for us NAT-captives. . Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration Difficulties
On Tue, 2003-09-30 at 20:21, Brian Capouch wrote: Does this imply that it will work even in a NAT environment? I have watched the list like a hawk for evidence of FWD working for machines placed behind NAT, but so far haven't seen that anyone could actually get it going. If so, that would be waay good news for us NAT-captives. . Yes, it is behind a NAT, I have an Intertex IX66 with ADSL modem and wireless card for a review see http://www.adslguide.org.uk/hardware/reviews/2003/q3/intertex_ix66-airsip.asp or http://www.modemhelp.org/reviews/january2003.html The verdict is really why tear your hair out? When I've ironed out the sipcall problem I put up my configs. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration between *'s
Here are a few outgoing gateway configs that work for me. [vocal] type=friend host=1.1.1.7 insecure=1 port=5065 accountcode=memrtr ;dtmfmode=info [cisco] type=friend host=1.1.1.3 insecure=1 canreinvite=no port=5060 dtmfmode=info accountcode=memrtr Xisco wrote: That's true if always there to connect two asterisk servers, but I'm doing some proves in order to connect one asterisk server with another SIP server. That's the matter. - Original Message - From: Jamie Carl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 12:12 PM Subject: Re: [Asterisk-Users] SIP registration between *'s ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration
I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration
Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration
Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC ON: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
No, it is not something you can fix by tweaking the configuration files, you should complain to the authors of the user agent. Anyway, it is a minor problem and I guess that most implementations can overcome it, but you should at least report it to the authors. Jan. On 19-09 09:17, Sergio Serrano Revuelto wrote: Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo
Re: [Asterisk-Users] SIP registration
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 18 September 2003 19:04, Hielke Christian Braun wrote: try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. It didn't. And now something else is weird. Asterisk fails sending audio to my SIP phone. Found this in my logs: Sep 19 11:08:52 WARNING[950291]: File channel.c, Line 1819 (ast_channel_make_compatible): No path to translate from SIP/sc.sc.sc.sc-de54( 4) to H323/ip$hc.hc.hc.hc:1244/14060(8) Sep 19 11:08:58 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [hex]@ as.as.as.as for seqno 102 (Request) Sep 19 11:09:04 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [hex]@ as.as.as.as for seqno 102 (Request) What on earth is this? Codec? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/as2r2TEAILET3McRAtIaAJ9Hpa3k/a7giiB62pwn7qw17jck/ACeJLdH fzoRqSVrEMfgAfzE5BOogoU= =N4hn -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration between *'s
Why? Use IAX2, it is s much better... J On Fri, 19 Sep 2003 11:54:23 +0200 Xisco [EMAIL PROTECTED] wrote: Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =usuario1:pass1@public_ip_2 In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=public_ip_1 dtmfmode=inband Logs in * are the followings In * one logs: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK488fe503;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as504a35d0 To: sip:usuario1@public_ip_2;tag=as2a0e47ce Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 103 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 (no NAT) topublic_ip_2:5060 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2;tag=as13445743 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 In * two logs: NOTICE[81926]: File chan_sip.c, Line 4816 (handle_request): Registration from 'sip:usuario1@public_ip_2' failed for 'public_ip_1' Sip read: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 11 headers, 0 lines Using latest request as basis request Sending to public_ip_1: 5060 (NAT) Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2;tag=as1538b8a6 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 Any idea to fix the problem Any special configuration in sip.conf Thanks a lot. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration between *'s
That's true if always there to connect two asterisk servers, but I'm doing some proves in order to connect one asterisk server with another SIP server. That's the matter. - Original Message - From: Jamie Carl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 12:12 PM Subject: Re: [Asterisk-Users] SIP registration between *'s Why? Use IAX2, it is s much better... J On Fri, 19 Sep 2003 11:54:23 +0200 Xisco [EMAIL PROTECTED] wrote: Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =usuario1:pass1@public_ip_2 In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=public_ip_1 dtmfmode=inband Logs in * are the followings In * one logs: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK488fe503;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as504a35d0 To: sip:usuario1@public_ip_2;tag=as2a0e47ce Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 103 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 (no NAT) topublic_ip_2:5060 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2;tag=as13445743 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 In * two logs: NOTICE[81926]: File chan_sip.c, Line 4816 (handle_request): Registration from 'sip:usuario1@public_ip_2' failed for 'public_ip_1' Sip read: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 11 headers, 0 lines Using latest request as basis request Sending to public_ip_1: 5060 (NAT) Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2;tag=as1538b8a6 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 Any idea to fix the problem Any special configuration in sip.conf Thanks a lot. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration between *'s
Doesn't matter it should still work. Here is a hint.. dont use passwords/secrets it will then work! bkw On Fri, 19 Sep 2003, Xisco wrote: That's true if always there to connect two asterisk servers, but I'm doing some proves in order to connect one asterisk server with another SIP server. That's the matter. - Original Message - From: Jamie Carl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 12:12 PM Subject: Re: [Asterisk-Users] SIP registration between *'s Why? Use IAX2, it is s much better... J On Fri, 19 Sep 2003 11:54:23 +0200 Xisco [EMAIL PROTECTED] wrote: Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =usuario1:pass1@public_ip_2 In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=public_ip_1 dtmfmode=inband Logs in * are the followings In * one logs: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK488fe503;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as504a35d0 To: sip:usuario1@public_ip_2;tag=as2a0e47ce Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 103 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 (no NAT) topublic_ip_2:5060 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK59f913b2;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as4f879ac7 To: sip:usuario1@public_ip_2;tag=as13445743 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 In * two logs: NOTICE[81926]: File chan_sip.c, Line 4816 (handle_request): Registration from 'sip:usuario1@public_ip_2' failed for 'public_ip_1' Sip read: REGISTER sip:public_ip_2SIP/2.0 Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:s@public_ip_1 Event: registration Content-length: 0 11 headers, 0 lines Using latest request as basis request Sending to public_ip_1: 5060 (NAT) Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP public_ip_1:5060;branch=z9hG4bK0f194106;received=public_ip_1 From: sip:usuario1@public_ip_2;tag=as35957f60 To: sip:usuario1@public_ip_2;tag=as1538b8a6 Call-ID: 77064d8f2fdfe4746d509dc2488fe503@public_ip_1 CSeq: 119 REGISTER User-Agent: Asterisk PBX Contact: sip:usuario1@public_ip_2 Content-Length: 0 Any idea to fix the problem Any special configuration in sip.conf Thanks a lot. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration
sip show registry is when asterisk registers with some gateway. you want to look at sip show peers or sip show users. regards Martin On Thu, 31 Jul 2003, Steve Woolley wrote: I am trying to get SIP registrations to work within Asterisk. From my snom 200 phone (and on my SJPhone soft client) I can dial via extension. Example: To Dial extension 1110 on my asterisk1 server: I can simply enter SIP:[EMAIL PROTECTED] and the call goes through just like it should. As I understand it (and I probably don't), once my SIP device has established communication with the asterisk server, it registers the device name (in the sip registry) and thus I can dial the phone by entering: SIP:[EMAIL PROTECTED] (providing of course snom1 is the context for my sip phone in sip.conf) In fact I do see the following on the sip console when I make a call from snom1: asterisk1*CLI -- Registered SIP 'snom1' at 172.16.14.11 port 5060 expires 3600 -- Executing Macro(SIP/snom1-a17d, oneline|Zap/4) in new stack -- Executing Dial(SIP/snom1-a17d, Zap/4|20) in new stack -- Called 4 -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Zap/4-1 is ringing -- Zap/4-1 is ringing I haven't found much documentation on sip registration in asterisk, but I kind of assumed that entering sip show registry on the console would show me the registrations, but only the following is returned by this command: asterisk1*CLI sip show registry Host Username Refresh State Anyone have any ideas? -- Steve Woolley ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, FL 32746 (407)682-6226 x1110 http://www.adstelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
