sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

[Serusers] Asterisk Authenticating as UAC ..

by Felipe Martins

Hi everybody, I have a SER Server (Proxy/REGISTRAR) and Asterisk (PSTN/UAC), I need asterisk to act as a UAC to register at a certain account at another server in order to do external calls. At this point I can make my users auth my SER Server normally, and I can also make every or desired calls to be forwarded to asterisk, but there is a problem. I need my clients to make and receive calls, none of them are working outbound because a misconfiguration of my own, and outbound by the simple fact that when asterisk gets the forwarded packets from SER, its not authenticating at my External Voip Provider to get a line. I need Asterisk to Automatically auth any user which have theier packets forwarded by ser to it, at my external voip provider. Do anybody know what to do, or have made it before ? I'm having some problems with that. Thanks in Advance -- Felipe Martins TEP Solution & New Technologies Mundivox Communications fmartins(a)mundivox.com Site: www.mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844

19 years, 9 months

Re: [Serusers] FW: SER Asterisk Voicemail

by Aisling O'Driscoll

Would someone be so kind as to give me a few guidelines for "dial-in" access to asterisk so that the user can access voicemail, based on the below email. I think that is what is happening, the caller is leaving a message which is being saved on Asterisk buts ince the users are registered with ser they cant access the voicemail. I already have an account for each user in sip.conf,extensions.conf and voicemail.conf on Asterisk as suggested in the archives. Thanks a million, Aisling. ---- Original Message ---- From: blairs(a)isc.upenn.edu To: ashling.odriscoll(a)cit.ie Subject: Re: [Serusers] FW: SER Asterisk Voicemail Date: Mon, 14 Feb 2005 13:22:38 -0500 > >If the message is only sent as an email attachment >(delete=yes,attach=yes) then >the user must listen to it by playing the attached wav file on their >pc. > >If the message is saved on the Asterisk server then you need to >provide >"dial-in" access to Asterisk that sends the caller to VoiceMailMain. > From there >they can access their mailbox and manage messages. > >_Steve > >Aisling O'Driscoll wrote: > >>Any more ideas on my below mail? If a user is registered with SER >and >>leaves a voicemail message with asterisk (by using rewritehostport >>etc in ser.cfg), then how is the user supposed to listen to the >>message afterwards? Is there any other way other than the MWI >method?? >> >>Thnaksm >>Aisling. >> >>---- Original Message ---- >>From: ashling.odriscoll(a)cit.ie >>To: asterisk-users(a)lists.digium.com >>Subject: FW: SER Asterisk Voicemail >>Date: Thu, 10 Feb 2005 16:45:53 -0000 >> >>Hi all, >> >>I have SER and Asterisk set up together with ser handling user >>registrations and asterisk providing voicemail services. When I ring >>a phone and it doesnt answer after a designated amount of time, the >>request is forwarded to asterisk, and I can leave a message. >> >>Now, this may seem a ridiculous question but how can I listen to my >>message afterwards? I have read about a solution by Java Rockx using >>sipsak for sending mwi sip notify messages to the phone but is there >>a simpler way which I am blindly ignoring?? >> >>Thank you in advance, >>Aisling. >> >> >>-------------------Legal >Disclaimer--------------------------------------- >> >>The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)iptel.org >>http://mail.iptel.org/mailman/listinfo/serusers >> >> > >-- > >ISC Network Engineering >The University of Pennsylvania >3401 Walnut Street, Suite 221A >Philadelphia, PA 19104 > > >voice: 215-573-8396 > > 215-746-8001 > >fax: 215-898-9348 > >sip:blairs@upenn.edu > > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

19 years, 9 months

[Serusers] Aastra Intelligate Registration Problem

by Martin Koenig

Hello, we have a problem with the SIP trunk of an Aastra Intelligate PBX. Registration fails with the SER error message "pre_auth(): Credentials received are not filled properly". SER is 0.8.14. See ngrep: # U 2005/02/15 11:40:46.093312 aastra_intelligate:5060 -> toplink_proxy:5060 REGISTER sip:toplink-voice.de SIP/2.0. Via: SIP/2.0/UDP aastra_intelligate:5060;branch=fc15d6ace7866108222849a9dd6303d8. To: username<sip:username@toplink-voice.de:5060>. From: username<sip:username@toplink-voice.de:5060>;tag=f52ad23f5a30a9cd. Call-ID: 182a55ff8fb00e0d31a6f7cb9b8c22b9@aastra_intelligate. CSeq: 2289 REGISTER. Max-Forwards: 70. Expires: 3000. Contact: <sip:username@aastra_intelligate>. Allow: ACK,BYE,CANCEL,INVITE. User-Agent: Aastra Intelligate. Content-Length: 0. . # U 2005/02/15 11:40:46.093883 toplink_proxy:5060 -> aastra_intelligate:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP aastra_intelligate:5060;branch=fc15d6ace7866108222849a9dd6303d8. To: username<sip:username@toplink-voice.de:5060>;tag=16ac3fc2258766c821c391b58b08db64.9f29. From: username<sip:username@toplink-voice.de:5060>;tag=f52ad23f5a30a9cd. Call-ID: 182a55ff8fb00e0d31a6f7cb9b8c22b9@aastra_intelligate. CSeq: 2289 REGISTER. WWW-Authenticate: Digest realm="toplink-voice.de", nonce="4211d2da1728b0bd58773cf042217a138e8508ca", qop="auth". Content-Length: 0. . # U 2005/02/15 11:40:46.321069 aastra_intelligate:5060 -> toplink_proxy:5060 REGISTER sip:toplink-voice.de SIP/2.0. Via: SIP/2.0/UDP aastra_intelligate:5060;branch=c46c24632f85f6b001dca195835600a4. To: username<sip:username@toplink-voice.de:5060>. From: username<sip:username@toplink-voice.de:5060>;tag=f52ad23f5a30a9cd. Call-ID: 182a55ff8fb00e0d31a6f7cb9b8c22b9@aastra_intelligate. CSeq: 2290 REGISTER. Max-Forwards: 70. Expires: 3000. Contact: <sip:username@aastra_intelligate>. Allow: ACK,BYE,CANCEL,INVITE. Authorization: Digest nc=00000001,nonce="4211d2da1728b0bd58773cf042217a138e8508ca",qop=auth,realm="toplink-voice.de",response="62989172348871cf1fd92b4bc9bc3be2",uri="sip:toplink-voice.de",username="username". User-Agent: Aastra Intelligate. Content-Length: 0. . # U 2005/02/15 11:40:46.321559 toplink_proxy:5060 -> aastra_intelligate:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP aastra_intelligate:5060;branch=c46c24632f85f6b001dca195835600a4. To: username<sip:username@toplink-voice.de:5060>;tag=16ac3fc2258766c821c391b58b08db64.f64f. From: username<sip:username@toplink-voice.de:5060>;tag=f52ad23f5a30a9cd. Call-ID: 182a55ff8fb00e0d31a6f7cb9b8c22b9@aastra_intelligate. CSeq: 2290 REGISTER. Content-Length: 0. When I take a look at the Authorization Header of the PBX: Authorization: Digest nc=00000001, nonce="4211d2da1728b0bd58773cf042217a138e8508ca", qop=auth, realm="toplink-voice.de", response="62989172348871cf1fd92b4bc9bc3be2", uri="sip:toplink-voice.de", username="username" It is obvious that the cnonce is missing. According to RFC2617 it should be present, right? Quote RFC2617: "cnonce This MUST be specified if a qop directive is sent (see above), and MUST NOT be specified if the server did not send a qop directive in the WWW-Authenticate header field. The cnonce-value is an opaque quoted string value provided by the client and used by both client and server to avoid chosen plaintext attacks, to provide mutual authentication, and to provide some message integrity protection. See the descriptions below of the calculation of the response- digest and request-digest values." Could anyone please verify this? Testing with the SIPgate.de SER proxy, registration works. How is this possible if PBX is not sending RFC2617 compilant Authorization headers? With best regards, Martin Koenig

19 years, 9 months

[Serusers] Question regarding SER/Asterisk functionality

by "Geir O. Høgberg"

Hi all, I'm currently looking for a VoIP platform to support the following features: Caller ID Call Waiting with caller ID Call Hold/Retrieve Three-way conference Calling Line Identity Presentation Call back last missed call Last called number redial User line locking/Call Barring (all current levels) Itemised bill Call Forward Call Forward on No Reply Call Forward on Busy Call Forward Unconditional VoiceMail - Message Waiting Indicator - Call back option Number Portability Secret Number Legal intercept Emergency Number Routing Anonymous Caller Rejection Directory Service update Feature code activation I have looked into both SER and Asterisk, and found out that SER should be the most appropriate platform since most of the users will use a SIP adapter/endpoint and be located on the Internet (like most of the VoIP services these days). So my question to the list is to gather some experiences and possible remarks/comments on which direction I should go :D Any help would be gladly appreciated :-) -- Geir

19 years, 9 months

RE: [Serusers] INVITE retransmission

by Zappasodi Daniele

Thanks Marian, this is the function that I'm looking for. Now my ser.cfg works correctly. regards Daniele -----Original Message----- From: Marian Dumitru [mailto:marian.dumitru@voice-sistem.ro] Sent: Tue 2/15/2005 10:42 AM To: Zappasodi Daniele Cc: Klaus Darilion; serusers(a)iptel.org Subject: Re: [Serusers] INVITE retransmission Hi Daniele, if the transaction already exists (as created by t_newtran()), use t_forward_nonack_uri() instead of t_relay(). Best regards, Marian Zappasodi Daniele wrote: > Hi Klaus, > if I use t_newtran() and then I invoke the t_relay() to transmit the INVITE, I encounter this error message: > t_newtran: transaction already in process > and the proxy responds with > 500 : I'm terribly sorry, server error occurred. > > Regards > Daniele > > -----Messaggio originale----- > Da: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Inviato: martedì 15 febbraio 2005 9.33 > A: Zappasodi Daniele > Cc: serusers(a)iptel.org > Oggetto: Re: [Serusers] INVITE retransmission > > > Hi! > > If the first INVITE is processed by a function of the tm module, the > retransmissions will be absorbed by ser automatically. > > take a look at: > http://www.iptel.org/ser/doc/seruser/seruser.html#STATEFULUA > > regards, > klaus > > Zappasodi Daniele wrote: > > >>Hi, >>how can I distinguish, in ser.cfg, a retransmitted INVITE from the new INVITE? >>I need to invoke some functions only for the new INVITEs before transmit them. >> >>Regards >>Daniele >> >> >>********************************************************************** >>The information in this message is confidential and may be legally >>privileged. It is intended solely for the addressee. Access to this message >>by anyone else is unauthorized. If you are not the intended recipient, any >>disclosure, copying, or distribution of the message, or any action or >>omission taken by you in reliance on it, is prohibited and may be unlawful. >>Please immediately contact the sender if you have received this message in >>error. >> >>********************************************************************** >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)iptel.org >>http://mail.iptel.org/mailman/listinfo/serusers > > > > ********************************************************************** > The information in this message is confidential and may be legally > privileged. It is intended solely for the addressee. Access to this message > by anyone else is unauthorized. If you are not the intended recipient, any > disclosure, copying, or distribution of the message, or any action or > omission taken by you in reliance on it, is prohibited and may be unlawful. > Please immediately contact the sender if you have received this message in > error. > > ********************************************************************** > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers -- Voice System http://www.voice-system.ro ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

19 years, 9 months

[Serusers] serweb

by Petri Nevalainen

Hi every one.. Can someone send me instructors or note how to I can configure Serweb in my laptop(Fedora core 3)..or link where I can find easy notes for dummy's ;) because I have some problems.. thanks.. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

19 years, 9 months

Re: [Serusers] INVITE retransmission

by Zappasodi Daniele

Hi Klaus, if I use t_newtran() and then I invoke the t_relay() to transmit the INVITE, I encounter this error message: t_newtran: transaction already in process and the proxy responds with 500 : I'm terribly sorry, server error occurred. Regards Daniele -----Messaggio originale----- Da: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Inviato: martedì 15 febbraio 2005 9.33 A: Zappasodi Daniele Cc: serusers(a)iptel.org Oggetto: Re: [Serusers] INVITE retransmission Hi! If the first INVITE is processed by a function of the tm module, the retransmissions will be absorbed by ser automatically. take a look at: http://www.iptel.org/ser/doc/seruser/seruser.html#STATEFULUA regards, klaus Zappasodi Daniele wrote: > Hi, > how can I distinguish, in ser.cfg, a retransmitted INVITE from the new INVITE? > I need to invoke some functions only for the new INVITEs before transmit them. > > Regards > Daniele > > > ********************************************************************** > The information in this message is confidential and may be legally > privileged. It is intended solely for the addressee. Access to this message > by anyone else is unauthorized. If you are not the intended recipient, any > disclosure, copying, or distribution of the message, or any action or > omission taken by you in reliance on it, is prohibited and may be unlawful. > Please immediately contact the sender if you have received this message in > error. > > ********************************************************************** > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

19 years, 9 months

[Serusers] PIX operating system SIP bug

by Luca Corti

Hello, This is just to confirm that if you have a Cisco PIX Firewall sitting between your SIP Proxy (eg SER) and your PSTN gateway (in my case a Cisco AS5350) you need at least version 6.3 of the PIX IOS. There's a nasty bug in versions <= 6.2 which prevents transactions to be correctly established by blocking SDP packets. Maybe this will save someone else time Luca

19 years, 9 months

[Serusers] INVITE retransmission

by Zappasodi Daniele

Hi, how can I distinguish, in ser.cfg, a retransmitted INVITE from the new INVITE? I need to invoke some functions only for the new INVITEs before transmit them. Regards Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

19 years, 9 months

[Serusers] Disconnecting a call.

by Tom Gaudasinski

Greetings, Another question from me. I need to be able to liberally disconnect a call that's in progress at any time. Is there any way to do this. I want to implement a pseudo prepaid billing, and I need something that will allow for call termination once credits have been used up. Is there any way i can send SER a message or anything else? Or maybe set a timer for maximum call duration so that the call will be killed after it passes? Thank you.

19 years, 9 months

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sr-users February 2005