Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no register => gateway:123@188.208.12.37/2222 [gateway] type=friend callerid="sip" <2222> username=gateway host=188.208.12.37 secret=123 My extensions.conf exten => 2222,1,dial,SIP/2222@188.208.12.37|60|rTt exten => 2222,2,Hangup I'm going crazy with this...I think that I'm not doing well the registration but I can't find why!! 188.208.12.237 is the IP of the asterisk and 188.208.12.37 is the IP of the SIP gateway. 2222 is one of the phones of the SIP Gateway...Anyone can help????Please! Thanks very very much Michelle ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/
I'm afraid I have no idea what your goal is here. Do you have a phone somewhere in this configuration? I don't see it. Please explain what it is you are trying to do. From what I see (though much data is missing from your explanatin) anytime you place a call, it will result in a loop. While you're at it, include the following information: sip show peers sip show registry sip debug (and wait for a cycle of SIP messages to go by) JT>Hi! >I have a new problem with my SIP device.I have done some changes and >now I receive continuosly a SIP message: "501" "Not impelmented" back >from the SIP Gateway. I can see that it doesn't register to Asterisk. >I have in the SIP device: > >Registrar 1: UnRegistered to: 2222 >registrar: 188.208.12.237 5060 expires: 2000 >name: gateway passwd: 123 > > >My sip.conf: > >[general] >port = 5060 >bindaddr = 0.0.0.0 >context = default >transfer = yes >threewaycalling = yes >usecallerid = yes >hidecallerid = no >register => gateway:123@188.208.12.37/2222 > >[gateway] >type=friend >callerid="sip" <2222> >username=gateway >host=188.208.12.37 >secret=123 > >My extensions.conf > >exten => 2222,1,dial,SIP/2222@188.208.12.37|60|rTt >exten => 2222,2,Hangup > >I'm going crazy with this...I think that I'm not doing well the >registration but I >can't find why!! 188.208.12.237 is the IP of the asterisk and 188.208.12.37 >is the IP of the SIP gateway. 2222 is one of the phones of the SIP >Gateway...Anyone can help????Please! >Thanks very very much >Michelle > >----- >Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam >http://mixmail.ya.com >Ya.com ADSL Home 24h, M?dem + Alta + 1 mes Gratis >http://acceso.ya.com/adslhome24h/ > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users
I have a SIP Gateway with 2 phones, and a MGCP Gateway with other two. I want all the phones to call to the PSTN and to call between them. MGCP goes OK but SIP doesn't. I can call them but the can't call. I'm starting to think that is a problem of the SIP device, not an Asterisk problem. michelle >I'm afraid I have no idea what your goal is here. Do you have a >phone somewhere in this configuration? I don't see it. Please >explain what it is you are trying to do. From what I see (though >much data is missing from your explanatin) anytime you place a call, >it will result in a loop. >While you're at it, include the following information: >sip show peers >sip show registry >sip debug (and wait for a cycle of SIP messages to go by) >JT >>Hi! >>I have a new problem with my SIP device.I have done some changes and >>now I receive continuosly a SIP message: "501" "Not impelmented" back >>from the SIP Gateway. I can see that it doesn't register to Asterisk. >>I have in the SIP device: >> >>Registrar 1: UnRegistered to: 2222 >>registrar: 188.208.12.237 5060 expires: 2000 >>name: gateway passwd: 123 >> >> >>My sip.conf: >> >>[general] >>port = 5060 >>bindaddr = 0.0.0.0 >>context = default >>transfer = yes >>threewaycalling = yes >>usecallerid = yes >>hidecallerid = no >>register => <A href="javascript:sendMsg ('gateway:123@188.208.12.37/2222');">gateway:123@188.208.12.37/2222< /A> >> >>[gateway] >>type=friend >>callerid="sip" <2222> >>username=gateway >>host=188.208.12.37 >>secret=123 >> >>My extensions.conf >> >>exten => <A href="javascript:sendMsg ('2222,1,dial,SIP/2222@188.208.12.37|60|rTt');">2222,1,dial,SIP/2222@188.2 08.12.37|60|rTt</A> >>exten => 2222,2,Hangup >> >>I'm going crazy with this...I think that I'm not doing well the >>registration but I >>can't find why!! 188.208.12.237 is the IP of the asterisk and 188.208.12.37 >>is the IP of the SIP gateway. 2222 is one of the phones of the SIP >>Gateway...Anyone can help????Please! >>Thanks very very much >>Michelle >> >>----- >>Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam >><A href="http://mixmail.ya.com/app/message? l=es&o=8&url=http%3A%2F%2Fmixmail%2Eya%2Ecom" target=_blank>http://mixmail.ya.com</A> >>Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis >><A href="http://mixmail.ya.com/app/message?l=es&o=8&url=http%3A% 2F%2Facceso%2Eya%2Ecom%2Fadslhome24h%2F" target=_blank>http://acceso.ya.com/adslhome24h/</A> >> >>_______________________________________________ >>Asterisk-Users mailing list ><A href="javascript:sendMsg ('>Asterisk-Users@lists.digium.com');">>Asterisk- Users@lists.digium.com</A> >><A href="http://mixmail.ya.com/app/message?l=es&o=8&url=http%3A% 2F%2Flists%2Edigium%2Ecom%2Fmailman%2Flistinfo%2Fasterisk% 2Dusers" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk- users</A> >_______________________________________________ >Asterisk-Users mailing list ><A href="javascript:sendMsg('Asterisk- Users@lists.digium.com>http://lists.digium.com/mailman/listinfo/asterisk- users');">Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users</A> ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/
Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -----snip---- SEND >> provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number <sip:pstn-number@provider.com> To: pstn-number <sip:pstn-number@provider.com> Contact: "pstn-number" <sip:pstn-numer@ip.ip.ip.ip:5060> Call-ID: 4EE3811126954242ACEC131F369A30A6@telia.com CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username="pstn-number",realm="provider.com",nonce="MTA3NzAyOTUwMjk2NWI2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0",response="f833201fd4a8719ea9a2e c505debbd56",uri="sip:provider.com",opaque="dd5d790f90d0307c7390cdb8f6e9 4cc8",qop=auth,cnonce="4B86525A67C646469656D90AD4C1273C",nc=00000002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE << provider.ip.ip.ip:5060 SIP/2.0 200 OK --------- end snip ------------- This is Asterisk (does not work) ------snip------------ Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: <sip:pstn-number@provider.com>;tag=as017cdd56 To: <sip:pstn-number@provider.com> Call-ID: 0597f20209eb7890424ba0d70eeb08b7@ip.ip.ip CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: <sip:s@ip.ip.ip.ip> Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI> Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says "From: number <number@provider.com>" asterisk only says "From: <number@provider.com>" How do I tell my Asterisk to send the registration as xlite ? /Mike
Mike, That is the only difference you see? Check out the Digest line. I would have thought that you would be seeing a 401 instead of 403 though (auth required instead of forbidden). The phone number part should be a meaningless string. You can test by telneting directly to your provider and entering the header information with the phone number and see what happens. T -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Micke Andersson Sent: Tuesday, February 17, 2004 12:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP REGISTER Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -----snip---- SEND >> provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number <sip:pstn-number@provider.com> To: pstn-number <sip:pstn-number@provider.com> Contact: "pstn-number" <sip:pstn-numer@ip.ip.ip.ip:5060> Call-ID: 4EE3811126954242ACEC131F369A30A6@telia.com CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username="pstn-number",realm="provider.com",nonce="MTA3NzAyOTUwMjk2NWI2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0",response="f833201fd4a8719ea9a2e c505debbd56",uri="sip:provider.com",opaque="dd5d790f90d0307c7390cdb8f6e9 4cc8",qop=auth,cnonce="4B86525A67C646469656D90AD4C1273C",nc=00000002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE << provider.ip.ip.ip:5060 SIP/2.0 200 OK --------- end snip ------------- This is Asterisk (does not work) ------snip------------ Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: <sip:pstn-number@provider.com>;tag=as017cdd56 To: <sip:pstn-number@provider.com> Call-ID: 0597f20209eb7890424ba0d70eeb08b7@ip.ip.ip CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: <sip:s@ip.ip.ip.ip> Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI> Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says "From: number <number@provider.com>" asterisk only says "From: <number@provider.com>" How do I tell my Asterisk to send the registration as xlite ? /Mike _______________________________________________ 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 ------------------------------------------------------------------------------ Notice: This e-mail message, together with any attachments, contains information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. ------------------------------------------------------------------------------
We have many Asterisk systems connected to us - we provider them with worldwide origination and termination and we have no problems. It could be provider configuration. If you can find out what kind of GW is in use at provider and version I can probably tell you how to configure it properly. Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 aram@hi-teck.com tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Micke Andersson Sent: Tuesday, February 17, 2004 9:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP REGISTER Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -----snip---- SEND >> provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number <sip:pstn-number@provider.com> To: pstn-number <sip:pstn-number@provider.com> Contact: "pstn-number" <sip:pstn-numer@ip.ip.ip.ip:5060> Call-ID: 4EE3811126954242ACEC131F369A30A6@telia.com CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username="pstn-number",realm="provider.com",nonce="MTA3NzAyOTUwMjk2NWI2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0",response="f833201fd4a8719ea9a2e c505debbd56",uri="sip:provider.com",opaque="dd5d790f90d0307c7390cdb8f6e9 4cc8",qop=auth,cnonce="4B86525A67C646469656D90AD4C1273C",nc=00000002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE << provider.ip.ip.ip:5060 SIP/2.0 200 OK --------- end snip ------------- This is Asterisk (does not work) ------snip------------ Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: <sip:pstn-number@provider.com>;tag=as017cdd56 To: <sip:pstn-number@provider.com> Call-ID: 0597f20209eb7890424ba0d70eeb08b7@ip.ip.ip CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: <sip:s@ip.ip.ip.ip> Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI> Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says "From: number <number@provider.com>" asterisk only says "From: <number@provider.com>" How do I tell my Asterisk to send the registration as xlite ? /Mike _______________________________________________ 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 3344 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040603/c74b3a50/winmail.bin
I'm having trouble making calls over my VoIP provider. I do successfully register, and when I try to establish a phone call Asterisk sends wrong username and password. Instead of sending username and pass that I have provided, he send username and pass of the SIP phone that is registered to * (the phone from which I try to make a call). What have I done wrong? This is my sip.conf [general] context=sip port=5060 bindaddr=0.0.0.0 srvlookup=no tos=184 maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=alaw allow=gsm musicclass=default useragent=PBX Lama nat=no externip = 200.200.200.200 ; my external IP localnet = 10.0.0.0/255.255.255.0 realm=lama.hr register => myusername:mypass@sip.iskon.hr canreinvite=no [iskon1] type=friend username=myusername secret=mypass host=sip.iskon.hr nat=yes canreinvite=no [214] callerid=Vice Lacmanovic <214> type=friend username=214 secret=vice host=dynamic mailbox=214 canreinvite=no dtmfmode=inband And this is part of my extensions.conf - the line I use for calling out. exten => _8.,1,Dial(SIP/${EXTEN:1}@iskon1) Again, problem is that Asterisk to my VoIP provider sends username 214 and pass vice (data of my SIP phone) and not the data that I have provide to it (myusername and mypassword for that VoIP provider). Thank you for your time. -- Tomislav Parcina tparcina#lama.hr
Subject: RE: SIP Register From: Tomislav Parcina <tparcina@lama.hr> In article <026e01c631b2$5fc9fad0$060117ac@musicroom>, mark@switchnet.com.au says...> First impressions telling me you want to check your phone settings. What > phone are you using and what are the config settings?Hi Mark, thank you for your reply. I'm using Cisco 7905 with SIP version 1.3.1(050608A). This phone has tone of settings (few pages). What exactly would you need? Why do you think it's phone problem and not Asterisk? Asterisk is the one that contents my provider. * is the one who should decide what information's to send to my VoIP provider... Anyway, I'm inexperienced with this and I'm just trying to understand what is happening and where could be the problem. -- Tomislav Parcina name.surname@email.t-com.hr
> Why do you think it's phone problem and not Asterisk? Asterisk is the > one that contents my provider. * is the one who should decide what > information's to send to my VoIP provider... Anyway, I'm inexperienced > with this and I'm just trying to understand what is happening and where > could be the problem.One more thing. Now I have tried with softphone. I have the same problem. Asterisk sends user and password of SIP account (SIP phone) that is making a call but not the account information's that I have received from my service provider. Question: How to configure Asterisk so he sends right user information's? -- Tomislav Parcina name.surname@email.t-com.hr