[asterisk-users] Asterisk pjsip registration issues
Hey all I am hoping someone can assist I have now spent over a week trying to figure out what is going on with PJSIP registrations. I am able to register handsets against an asterisk 13 server running pjsip, but I am not able to get pjsip to register out to an older chan_sip asterisk server. If I drop the registration I can make things work, but when I have to register the asterisk - pjsip server against another server the registration completes, but I can not send any calls across the registration, nor will it handle options correctly as well. We keep getting ... No auth credentials for realm(s) 'aster...@xxx.xxx.xxx.xxx' in challenge. in one form or another, and I have been unable to find any definitive documentation on what is at cause for this. In some areas I have seen responses saying it is an issue with realms so I have tried with and without but no success. I really need some direction on this. This is the last issue I know of that is holding up us from moving to pjsip. If I can't get asterisk / pjsip to register and send authenticated messages than it can't work for replacing chan_sip in all situations. What am I doing wrong. [zktech_trunk] type=registration endpoint=zktech_trunk transport=udp-nat outbound_auth=zktech_trunk server_uri=sip:acct.8...@xxx.xxx.xxx.xxx client_uri=sip:acct.8...@xxx.xxx.xxx.xxx contact_user=zktech_trunk retry_interval=60 forbidden_retry_interval=600 expiration=3600 line=yes [zktech_trunk] type=auth auth_type=userpass password=rossi72v8qr username=ACCT.8009 realm=aster...@xxx.xxx.xxx.xxx [zktech_trunk] type=aor max_contacts=1 contact=sip:acct.8...@privxxx.xxx.xxx.xxx:5060 qualify_frequency=60 Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk pjsip registration issues
On Tue, Sep 26, 2017, at 10:33 AM, Bryant Zimmerman wrote: > Hey all > > I am hoping someone can assist I have now spent over a week trying to > figure out what is going on with PJSIP registrations. > > I am able to register handsets against an asterisk 13 server running > pjsip, but I am not able to get pjsip to register out to an older > chan_sip > asterisk server. > If I drop the registration I can make things work, but when I have to > register the asterisk - pjsip server against another server the > registration completes, but I can not send any calls across the > registration, nor will it handle options correctly as well. > > We keep getting ... No auth credentials for realm(s) > 'aster...@xxx.xxx.xxx.xxx' in challenge. > in one form or another, and I have been unable to find any definitive > documentation on what is at cause for this. In some areas I have seen > responses saying it is an issue with realms so I have tried with and > without but no success. > > I really need some direction on this. This is the last issue I know of > that is holding up us from moving to pjsip. If I can't get asterisk / > pjsip > to register and send authenticated messages than it can't work for > replacing chan_sip in all situations. > > What am I doing wrong. Break this down further because you have some conflicting and confusing information. Does the outbound registration work or not work? Does it show as registered in PJSIP? If you leave out the "realm" option what happens? When you say "can't send any calls across the registration" what does that mean? Are you referring to inbound calls or outbound calls? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk pjsip registration issues
> Hey all > > I am trying to register a PJSIP server on our office to an Asterisk 11 > chan_sip server in a datacenter. > > I keep getting > WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178 > digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060': > Unable to create request with auth. No auth credentials for realm(s) > 'asterisk' in challenge. > > Any insights would be appreciated I have been banging my head for several > days now. I ran into a very similar problem when I tried to switch my PJSIP service with Vitelity from "fixed IP address" to "registration-based". I would try to place a call, and it would simply time out and then get a "busy here" error from Vitelity. Calls to a similar Vitelity sub-account from a Zoiper soft-phone worked just fine. I wiresharked the sessions and found that the critical difference seemed to be in the From: and Contact: headers. Zoiper set these to the Vitelity sub-account name (the registration name) while PJSIP just set them to "asterisk". I checked the PJSIP wizard file, and found that the outbound authentication object had the right username information in it, so that wasn't the problem. After stumbling around for hours, I found that it's necessary to set the "from_user" parameter in the endpoint object to match the username in the outbound authentication object. This causes PJSIP to send this value (rather than "asterisk") in the From and Contact fields of the INVITE, and this apparently gives the far end the information it needs to issue a proper credentials challenge. Once I added this one line to my definition and restarted, outbound calls worked like a charm. So, in pjsip_wizard, one would write something like [peername] type = wizard transport = transport-udp remote_hosts = outbound.peer.com sends_auth = yes endpoint/context = outbound endpoint/from_user = MYNAME outbound_auth/username = MYNAME outbound_auth/password = MYPASSWORD Modify and embellish as required. If you're writing your PJSIP objects individually rather than via the wizard, just set the fields in those objects appropriately. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk pjsip as voip client with multiple registrations
hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:3...@example.com:5060 client_uri=sip:3...@example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic) outbound_auth=308 aors=308 [308] type=identify endpoint=308 match=example.com my problem is contact on the other side (is same for all endpoints) Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:s@1.1.1.1:5060 all incoming calls from PBX to my Asterisk are routed to only one account (because of same ip address/port ?) how can i specify different source port or different contact address for asterisk pjsip client with registration? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations
On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: > hi, > > i want use asterisk+pjsip as voip client with multiple registrations > (perf testing) > > i'm using this example configuration for one account > > [308] > type=registration > outbound_auth=308 > server_uri=sip:3...@example.com:5060 > client_uri=sip:3...@example.com:5060 > > [308](auth-userpass) > username=308 > password=pass > > [308](aor-single-reg) > contact=sip:example.com:5060 > > [308](endpoint-basic) > outbound_auth=308 > aors=308 > > [308] > type=identify > endpoint=308 > match=example.com > > > my problem is contact on the other side (is same for all endpoints) > > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:s@1.1.1.1:5060 > > all incoming calls from PBX to my Asterisk are routed to only one > account (because of same ip address/port ?) > > how can i specify different source port or different contact address for > asterisk pjsip client with registration? The "contact_user" option configures the user portion of the Contact that is sent in the REGISTER. You can set it to a different value for each registration. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk pjsip registration issues - Solved
Dave from_user fixed the issue. Thank You Thank You Thank You I was about ready to chuck pjsip. The lack of good / complete documentation is a real problem. Man you saved me another late night. Thanks Bryant From: "Dave Platt" Sent: Tuesday, September 26, 2017 3:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk pjsip registration issues > Hey all > > I am trying to register a PJSIP server on our office to an Asterisk 11 > chan_sip server in a datacenter. > > I keep getting > WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178 > digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060': > Unable to create request with auth. No auth credentials for realm(s) > 'asterisk' in challenge. > > Any insights would be appreciated I have been banging my head for several > days now. I ran into a very similar problem when I tried to switch my PJSIP service with Vitelity from "fixed IP address" to "registration-based". I would try to place a call, and it would simply time out and then get a "busy here" error from Vitelity. Calls to a similar Vitelity sub-account from a Zoiper soft-phone worked just fine. I wiresharked the sessions and found that the critical difference seemed to be in the From: and Contact: headers. Zoiper set these to the Vitelity sub-account name (the registration name) while PJSIP just set them to "asterisk". I checked the PJSIP wizard file, and found that the outbound authentication object had the right username information in it, so that wasn't the problem. After stumbling around for hours, I found that it's necessary to set the "from_user" parameter in the endpoint object to match the username in the outbound authentication object. This causes PJSIP to send this value (rather than "asterisk") in the From and Contact fields of the INVITE, and this apparently gives the far end the information it needs to issue a proper credentials challenge. Once I added this one line to my definition and restarted, outbound calls worked like a charm. So, in pjsip_wizard, one would write something like [peername] type = wizard transport = transport-udp remote_hosts = outbound.peer.com sends_auth = yes endpoint/context = outbound endpoint/from_user = MYNAME outbound_auth/username = MYNAME outbound_auth/password = MYPASSWORD Modify and embellish as required. If you're writing your PJSIP objects individually rather than via the wizard, just set the fields in those objects appropriately. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations
Dne 26/09/2017 v 22:33 Joshua Colp napsal(a): On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:3...@example.com:5060 client_uri=sip:3...@example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic) outbound_auth=308 aors=308 [308] type=identify endpoint=308 match=example.com my problem is contact on the other side (is same for all endpoints) Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:s@1.1.1.1:5060 all incoming calls from PBX to my Asterisk are routed to only one account (because of same ip address/port ?) how can i specify different source port or different contact address for asterisk pjsip client with registration? The "contact_user" option configures the user portion of the Contact that is sent in the REGISTER. You can set it to a different value for each registration. ok i have this configuration now client - asterisk+pjsip (public ip 1.1.1.1) pjsip/307 pjsip/308 server - asterisk+chan_sip (public ip 2.2.2.2) sip/307 Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:307@1.1.1.1:5060 sip/308 Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:308@1.1.1.1:5060 now, every call from server to client is received through pjsip/307 . but i need receive call for pjsip/308 through registration of pjsip/308. is it possible? is it possible configure different source port other than 5060? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations
On Tue, Sep 26, 2017, at 05:53 PM, marek cervenka wrote: > Dne 26/09/2017 v 22:33 Joshua Colp napsal(a): > > On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: > >> hi, > >> > >> i want use asterisk+pjsip as voip client with multiple registrations > >> (perf testing) > >> > >> i'm using this example configuration for one account > >> > >> [308] > >> type=registration > >> outbound_auth=308 > >> server_uri=sip:3...@example.com:5060 > >> client_uri=sip:3...@example.com:5060 > >> > >> [308](auth-userpass) > >> username=308 > >> password=pass > >> > >> [308](aor-single-reg) > >> contact=sip:example.com:5060 > >> > >> [308](endpoint-basic) > >> outbound_auth=308 > >> aors=308 > >> > >> [308] > >> type=identify > >> endpoint=308 > >> match=example.com > >> > >> > >> my problem is contact on the other side (is same for all endpoints) > >> > >> Addr->IP : 1.1.1.1:5060 > >> Reg. Contact : sip:s@1.1.1.1:5060 > >> > >> all incoming calls from PBX to my Asterisk are routed to only one > >> account (because of same ip address/port ?) > >> > >> how can i specify different source port or different contact address for > >> asterisk pjsip client with registration? > > The "contact_user" option configures the user portion of the Contact > > that is sent in the REGISTER. You can set it to a different value for > > each registration. > > ok i have this configuration now > client - asterisk+pjsip (public ip 1.1.1.1) > pjsip/307 > pjsip/308 > > server - asterisk+chan_sip (public ip 2.2.2.2) > sip/307 > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:307@1.1.1.1:5060 > > sip/308 > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:308@1.1.1.1:5060 > > > now, every call from server to client is received through pjsip/307 . > but i need receive call for pjsip/308 through registration of pjsip/308. > is it possible? > is it possible configure different source port other than 5060? There is no ability to match to an endpoint currently based on the transport traffic comes in on. You can try enabling the line option[1] which may allow the inbound calls to be directed to an alternative endpoint. If this doesn't work you'll need to match all incoming to a single endpoint and direct things appropriately in the dialplan based on the dialed number. [1] http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
