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)" > 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
[asterisk-users] SIP registration issues
Hi, Having problems with a client trying to login to Asterisk 1.6.2 from behind a DSL router. The account can be accessed perfectly from other clients. Would appreciate if you could look at the the attached log and see if you spot any glaring issues. The user is very infrequently available for discussion and testing, so please try to batch questions in one mail itself! Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: Contact: ;q=1, ;q=0.667, ;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 <-> --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) <--- Transmitting (no NAT) to CLIENT-IP:49153 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d To: ;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09c83283" Content-Length: 0 <> Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:CLIENT-IP:49152 ---> REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: Contact: ;q=1, ;q=0.667, ;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 <-> --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) <--- Transmitting (no NAT) to CLIENT-IP:49153 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d To: ;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09c83283" Content-Length: 0 <> Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:CLIENT-IP:49152 ---> REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: Contact: ;q=1, ;q=0.667, ;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 <-> --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) <--- Transmitting (no NAT) to CLIENT-IP:49153 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d To: ;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09c83283" Content-Length: 0 <> Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:CLIENT-IP:49152 ---> REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: Contact: ;q=1, ;q=0.667, ;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 <-> --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) <--- Transmitting (no NAT) to CLIENT-IP:49153 ---> SIP/2.0 401 Unauthorized V
[Asterisk-Users] SIP registration issues
Hello. Is there any know issue with Asterisk 1.0.9 concerning intermittent SIP registration issues. My SIP hard phone (aastra 9133i) and soft phone (xlite) keep losing registration so calls to them go direct to VM although calling to other phones from them works fine. The logs show 'Transmitting (no NAT): SIP/2.0 403 Forbidden' which doesn't occur when they miraculously start working/registering. Asterisk seems to lose the user. Sep 9 11:47:36 VERBOSE[2444]: 12 headers, 0 lines Sep 9 11:47:36 VERBOSE[2444]: Using latest request as basis request Sep 9 11:47:36 VERBOSE[2444]: Sending to 192.168.1.100 : 5060 (non-NAT) Sep 9 11:47:36 VERBOSE[2444]: Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK289a5fe76 From: Martin ;tag=d6d383eca9b6910 To: Martin ;tag=as3c7c47f1 Call-ID: [EMAIL PROTECTED] CSeq: 54943697 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.1.100:5060 Sep 9 11:47:36 NOTICE[2444]: Registration from 'Martin ' failed for '192.168.1.100' Sep 9 11:47:36 VERBOSE[2444]: Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sep 9 11:47:36 VERBOSE[2444]: Sip read: REGISTER sip:192.168.1.50:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKd88070866 Max-Forwards: 70 Content-Length: 0 To: No User From: No User ;tag=0e8bc4f3c760bc2 Call-ID: [EMAIL PROTECTED] CSeq: 535959059 REGISTER Contact: No User Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 But then, some period of time later, they will start working at random times with no changes. Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
[Asterisk-Users] SIP Registration issues
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 have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any ideas? Thanks Andy 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.nixhelp.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: ;tag=as72c0d7da To: Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: Event: registration Content-Length: 0 (no NAT) to 192.168.1.2:5060 Sip read: REGISTER sip:sip.nixhelp.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: ;tag=as72c0d7da To: Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: Event: registration Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.2 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: ;tag=as72c0d7da To: ;tag=as72c0d7da Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: ontent-Length: 0 to 192.168.1.2:5060 Jul 20 23:46:40 NOTICE[81930]: chan_sip.c:7320 handle_request: Registration from '' failed for '192.168.1.2' Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sip read: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: ;tag=as72c0d7da To: ;tag=as72c0d7da Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 10 headers, 0 lines -- Got SIP response 403 "Forbidden" back from 192.168.1.2 Destroying call '[EMAIL PROTECTED]' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP registration issues - Ugly workaround
Hello everyone, I'm currently attempting to get Asterisk properly registering through a NAT proxy. Here's the twist, the provider does not permit direct SIP messages to the sip registry, instead they want registration to be done by their nat traversal proxy, and when you send-out the registration messages to the nat traversal, they must be sent as if they were originally sent directly to the sip proxy. To make a story short, I had to override the resolution of the SIP proxy hostname from my /etc/hosts to point to the nat traversal, and finally, after 4 weeks, I got asterisk to register. My registration line in /etc/asterisk/sip.conf looks like this: register => userid:password:[EMAIL PROTECTED]:5065/401 where userid is a numerical 8 digit value. Is there a more elegant method on accomplishing this? Thanks Kris ___ 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
