Re: [asterisk-users] WebRTC demo phones
Hello David, I'd recommend trying http://sipjs.com/ , it's similar to sipjs but you can choose which kind of media it uses via a configuration object: http://sipjs.com/guides/make-call/ Check out the guides, they are extremely clear and informative: http://sipjs.com/guides/ cheers, Olli 2015-03-12 9:20 GMT+02:00 Mitul Limbani mi...@enterux.in: Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the enable video checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with Rejecting secure video stream without encryption details. - sipML5, but it won't register, perhaps something to do with not using the Asterisk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains RTP/AVP not RTP/SAVP, and Asterisk rejects it with We are requesting SRTP for audio, but they responded without it! Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 -- _ -- 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] Question regarding custom announcements used by several Asterisk servers
Oops, quite right, how typoful of me! Thanks for the excellent points, I'll look into gluster and puppet and see may way onwards from there. cheers, Olli 2015-02-06 12:32 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk: On 06/02/15 07:54, Olli Heiskanen wrote: My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? I assume you mean NFS. Yes you can do that although using NFS you will then have a single point of failure and in the standard NFS client configuration if you try to access a file which is on NFS but it is unavailable then the file access will hang. So you might be better off having the files copied onto each of the asterisks servers local file storage or use a redundant file system such as gluster. -- _ -- 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] Question regarding custom announcements used by several Asterisk servers
Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- 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] Problem with odbc connector with cdr
Hello, I'm stuck with getting cdr records stored in MySql database. I have a working realtime environment and have verified that the db connection works fine when used via res_config_mysql.conf. I'd appreciate Your help on how to get the odbc connector working as I think there's something wrong with its configuration. The problem presented itself as an error when making a call that was supposed to be stored in mysql cdr: Unable to retrieve database handle. CDR failed. This kept popping up every time I was trying to have a call be stored in my mysql cdr table, and I traced the problem to the odbc connector, where my knowhow is about to end. One problem may be that odbcinst -j outputs wrong ini files, I believe I need these to be under /etc/ instead of /usr/local/etc/, but I'm unsure about how to change those permanently. # odbcinst -j unixODBC 2.3.2 DRIVERS: /usr/local/etc/odbcinst.ini SYSTEM DATA SOURCES: /usr/local/etc/odbc.ini FILE DATA SOURCES..: /usr/local/etc/ODBCDataSources USER DATA SOURCES..: /root/.odbc.ini SQLULEN Size...: 4 SQLLEN Size: 4 SQLSETPOSIROW Size.: 2 Odbcinst does see the connection, it reads it from /usr/local/etc/odbcinst.ini. # odbcinst -q -d [MySQL] I tried copying my connection information onto those files listed above and I'm not getting any errors in Asterisk logs when restarting Asterisk. However: # echo select 1 | isql -v MySQL [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect Here's the contents of odbc.ini: [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = serverip USER = myuser PASSWORD = mypass PORT = 3306 DATABASE = asterisk And odbcinst.ini: [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc5.so Setup = /usr/lib/libodbcmyS.so Driver64= /usr/lib64/libmyodbc5.so Setup64 = /usr/lib64/libodbcmyS.so FileUsage = 1 And here I hit a wall, how can I fix the configuration? MySQL-asterisk is referred to in res_odbc.conf, whose label is referred to in cdr_odbc.conf. When the odbc connector starts working, is this the correct way to configure the cdr db connection in Asterisk? Thanks, Olli -- _ -- 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] Problem with odbc connector with cdr
Thanks John, At first got an error using MySQL-asterisk, but then I removed /etc/ ini files and used the DSN in /usr/local/etc/odbc.ini, that did the trick for isql. I must have created the files /etc/ while following a guide online. Nice! After some meddling with the Asterisk conf files to have the correct values I got the cdrs working, thanks dude! cheers, Olli 2015-02-03 18:04 GMT+02:00 John Kiniston johnkinis...@gmail.com: I notice you have MySQL-asterisk as your definition in your odbc.ini but you are trying to connect to simply 'MySQL' with your 'isql' command. Does isql work with 'MySQL-asterisk' as the DSN instead of simply 'MySQL' ? I have machines that use /etc/odbc.ini and machines that use /usr/local/etc/odbc.ini depending on if I used a package to instal ODBC or if I compiled ODBC myself. On Tue, Feb 3, 2015 at 1:35 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, I'm stuck with getting cdr records stored in MySql database. I have a working realtime environment and have verified that the db connection works fine when used via res_config_mysql.conf. I'd appreciate Your help on how to get the odbc connector working as I think there's something wrong with its configuration. The problem presented itself as an error when making a call that was supposed to be stored in mysql cdr: Unable to retrieve database handle. CDR failed. This kept popping up every time I was trying to have a call be stored in my mysql cdr table, and I traced the problem to the odbc connector, where my knowhow is about to end. One problem may be that odbcinst -j outputs wrong ini files, I believe I need these to be under /etc/ instead of /usr/local/etc/, but I'm unsure about how to change those permanently. # odbcinst -j unixODBC 2.3.2 DRIVERS: /usr/local/etc/odbcinst.ini SYSTEM DATA SOURCES: /usr/local/etc/odbc.ini FILE DATA SOURCES..: /usr/local/etc/ODBCDataSources USER DATA SOURCES..: /root/.odbc.ini SQLULEN Size...: 4 SQLLEN Size: 4 SQLSETPOSIROW Size.: 2 Odbcinst does see the connection, it reads it from /usr/local/etc/odbcinst.ini. # odbcinst -q -d [MySQL] I tried copying my connection information onto those files listed above and I'm not getting any errors in Asterisk logs when restarting Asterisk. However: # echo select 1 | isql -v MySQL [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect Here's the contents of odbc.ini: [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = serverip USER = myuser PASSWORD = mypass PORT = 3306 DATABASE = asterisk And odbcinst.ini: [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc5.so Setup = /usr/lib/libodbcmyS.so Driver64= /usr/lib64/libmyodbc5.so Setup64 = /usr/lib64/libodbcmyS.so FileUsage = 1 And here I hit a wall, how can I fix the configuration? MySQL-asterisk is referred to in res_odbc.conf, whose label is referred to in cdr_odbc.conf. When the odbc connector starts working, is this the correct way to configure the cdr db connection in Asterisk? Thanks, Olli -- _ -- 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] Asterisk removes a charachter from sip peer name
Perfect, that's it! Thank you Paddy for pointing that out to me, I had
totally missed it!
Thanks,
Olli
2015-01-05 15:15 GMT+02:00 Paddy Grice pa...@wizaner.com:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olli Heiskanen
*Sent:* 03 January 2015 08:04
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Asterisk removes a charachter from sip peer
name
Hello all,
Just wondering on a behavior I noticed while testing with realtime sip
peers with names like 111@mydomain.com. Using Kamailio as outbound
proxy, it sends Asterisk a sip message where To header value is
sip:111@mydomain.com and From header has value username
sip:111@mydomain.com;transport=UDP;tag=fc609171. When Asterisk sends
out the sip message, the To header is as it was but as for From header,
Asterisk removes the . charachter from the user part of the sip uri, thus
resulting in 111333. Also the Contact header is affected the same way.
I was wondering what might be causing this? Does Asterisk not allow dots
in the peer names? The call itself connects so it's not much of an issue
but it would be good to know about this, as of course there's a chance I've
just missed something relevant.
cheers,
Olli
Sounds a bit like
From sip.conf
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and
'-' not
; in square brackets. For example, the caller id value 555. becomes
555
; when this option is enabled. Disabling this option results in no
modification
; of the caller id value, which is necessary when the caller id represents
something
; that must be preserved. This option can only be used in the [general]
section.
; By default this option is on.
;
;shrinkcallerid=yes ; on by default
Paddy
--
_
-- 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] Asterisk removes a charachter from sip peer name
Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111@mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is sip:111@mydomain.com and From header has value username sip:111@mydomain.com;transport=UDP;tag=fc609171. When Asterisk sends out the sip message, the To header is as it was but as for From header, Asterisk removes the . charachter from the user part of the sip uri, thus resulting in 111333. Also the Contact header is affected the same way. I was wondering what might be causing this? Does Asterisk not allow dots in the peer names? The call itself connects so it's not much of an issue but it would be good to know about this, as of course there's a chance I've just missed something relevant. cheers, Olli -- _ -- 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] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions ( http://sipjs.com/guides/server-configuration/asterisk/). Calls between these work nicely without problems. Now when I call from outside, from an external Asterisk 11.5 server, I end up having problems calling from a sip client to a webrtc client. The Asterisk I have on my main testing server is the latest current 11.14.1. When there's an internal call, Asterisk changes the sdp in the INVITE message and handles the rtp nicely, but it does not do so when the call comes from outside. Why not? Instead, it sends back 488 Not acceptable here. If I react to that in Kamailio and use rtpengine to rewrite the sdp, Asterisk accepts the INVITE and sends it to the websocket peer, but the sdp contains a very simple sdp with RTP/AVP profile. This I'd consider invalid behavior, since Asterisk knows the called party is webrtc and the INVITE already contains valid sdp with RTP/SAVPF profile. It's likely I have something wrong in my setup, or maybe I've overlooked something relevant? Question is, what is causing this behavior and what can I do to fix it? Either I'd need Asterisk to handle the sdp and rtp like it does for internal calls (which would be preferable in this case) or after the 488 sent by Asterisk I'd need Asterisk to relay the sdp offered by Kamailio/rtpengine as such without rewriting it. Here the call works fine (internal call from sip peer 771 to webrtc peer 660): INVITE that Asterisk (at port 5070) receives: PU.BL.IC.IP:5060 PU.BL.IC.IP:5070: SIP, length: 1046 INVITE sip:6...@testers.com;transport=UDP SIP/2.0 Record-Route: sip:PU.BL.IC.IP;lr=on;ftag=41030177 Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 Via: SIP/2.0/UDP AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z- Max-Forwards: 69 Contact: sip:7...@ast.er.isk.ip:38699;transport=UDP To: sip:6...@testers.com;transport=UDP From: 771sip:7...@testers.com;transport=UDP;tag=41030177 Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Length: 239 v=0 o=Z 0 0 IN IP4 AST.ER.ISK.IP s=Z c=IN IP4 AST.ER.ISK.IP t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Invite that Asterisk sends: PU.BL.IC.IP:5070 PU.BL.IC.IP:5060: SIP, length: 1238 INVITE sip:6...@pu.bl.ic.ip:5060 SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK26a2386a;rport Max-Forwards: 70 From: 771 win8 minipc sip:7...@testers.com:5070;tag=as05e60cc6 To: sip:6...@pu.bl.ic.ip:5060 Contact: sip:7...@pu.bl.ic.ip:5070 Call-ID: 7985f7161fcf1a6824b8388d451be...@testers.com CSeq: 102 INVITE User-Agent: I Am the Devil Date: Fri, 05 Dec 2014 15:50:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 663 v=0 o=root 777617621 777617621 IN IP4 PU.BL.IC.IP s=Asterisk PBX 11.14.1 c=IN IP4 PU.BL.IC.IP t=0 0 m=audio 15662 RTP/SAVPF 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:75a7e84431d15f682bd728ee10bd867d a=ice-pwd:028c19574216643c12188a8530f278f8 a=candidate:H5bdd423d 1 UDP 2130706431 PU.BL.IC.IP 15662 typ host a=candidate:H5bdd423d 2 UDP 2130706430 PU.BL.IC.IP 15663 typ host a=connection:new a=setup:actpass a=fingerprint:SHA-256 CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05 a=sendrecv Here the call fails (sip peer 201 calls from outside the server to webrtc peer 660): Invite that Asterisk receives: PU.BL.IC.IP:5060 PU.BL.IC.IP:5070: SIP, length: 1345 INVITE sip:660%40testers@pu.bl.ic.ip SIP/2.0 Record-Route: sip:PU.BL.IC.IP;lr=on;ftag=as4647f03c;nat=yes Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3264.8a896801756c527f2496fdc14e3f30ad.0 Via: SIP/2.0/UDP 192.168.0.201:5060 ;rport=5060;received=AST.ER.ISK.IP;branch=z9hG4bK56f5698e Max-Forwards: 69 From: Pirjo Ahvenainen sip:201@192.168.0.201;tag=as4647f03c To: sip:660%40testers@pu.bl.ic.ip Contact: sip:201@192.168.0.201:5060;alias=AST.ER.ISK.IP~5060~1 Call-ID: 69e66f05330de0063b5eba760191da6c@192.168.0.201:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.5.0 Date: Tue, 02 Dec 2014 08:34:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
Re: [asterisk-users] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello, Thanks Gareth for your reply. I assume you're referring to the first INVITE in my message, which is from the call that works. I don't know why the sdp displays only iLBC and speex at that point but the Zoiper client that's making the call is configured to support gsm, speex, ulaw, alaw, and iLBC, and the call works fine, audio and all, as the sdp that leaves Asterisk (thus reaches the called peer) actually contains ulaw, gsm and alaw. In the failing case Asterisk sends the INVITE via Kamailio to the called webrtc client, and in this message the rtp profile is m=audio 12902 RTP/AVP 0 3 8 101. Kamailio sends the INVITE to the client, which responds with 488. Kamailio notices this and uses rtpengine to handle the rtp, but: the client will not accept a second INVITE even though the sdp is correct this time: the client responds with 482 Loop Detected because the Call-ID is the same as the previous INVITE it got. This is why I can't handle the rtp using rtpengine, and here things have already gone wrong. So I need the INVITE to contain correct sdp when it leaves Asterisk, so sdp conversion and rtpengine would net be needed. Wonder if there's any way to do that? cheers, Olli 2014-12-05 18:53 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk: On 05/12/14 16:46, Olli Heiskanen wrote: INVITE that Asterisk (at port 5070) receives: PU.BL.IC.IP:5060 PU.BL.IC.IP:5070: SIP, length: 1046 INVITE sip:6...@testers.com;transport=UDP SIP/2.0 Record-Route: sip:PU.BL.IC.IP;lr=on;ftag=41030177 Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 Via: SIP/2.0/UDP AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z- Max-Forwards: 69 Contact: sip:7...@ast.er.isk.ip:38699;transport=UDP sip:7...@ast.er.isk.ip:38699;transport=UDP To: sip:6...@testers.com;transport=UDP From: 771sip:7...@testers.com;transport=UDP;tag=41030177 Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Length: 239 v=0 o=Z 0 0 IN IP4 AST.ER.ISK.IP s=Z c=IN IP4 AST.ER.ISK.IP t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv This client is saying it only supports speex and iLBC and would prefer them in that order. Your sip.conf appears to only permit alaw, ulaw and gsm so there is no mutual supported codec and hence the call fails. -- _ -- 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] Asterisk removes ice lines in sdp when calling between webrtc clients
Hi, Thanks Matthew for trying to reproduce the problem, I appreciate your efforts very much. There must be something off in my setup in one way or another. I could just discard this server and build a new one, but I think it's not good practice to leave a problem unsolved, so I'll continue trying to figure this out. One thing I noticed - don't know if it's relevant or not - due to a repo mismatch, I had problems with updating libgdiplus and libgdiplus-devel package, had to disable a repo and reinstall those and my mono installation (which is making me lose my hair). Is there a way to debug Asterisk itself? Or find the code that parses the outbound sdp? I figured there must be an if statement or more that determines whether or not to parse the ice lines into the sdp body. Finding that/those statements that produce the kind of sdp I'm seeing Asterisk send out, might tell something about what's wrong with my setup. As my c is not exactly fluent I wasn't sure which code files to search, can you guys help out with that? cheers, Olli 2014-10-03 11:31 GMT+03:00 Matthew Jordan mjor...@digium.com: On Thu, Oct 2, 2014 at 10:18 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hi, Thanks Eric for your reply, yes I know Asterisk replaces the sdp, however it should create ice lines when calling to a webrtc client, which it is currently not doing. To recap my problem (check previous messages for details); I have 2 webrtc clients (sip.js on chrome) with realtime information that appears to be correct. When calling from A to B, INVITE coming to Asterisk contains correct sdp, but when the INVITE leaves Asterisk, the sdp lacks ice lines. Unfortunately, I can't reproduce this. We've been running a lot of tests with a variety of SIP clients over the past week here at SIPit - both with and without ICE - and I haven't had a single instance of Asterisk failing to provide any ICE candidates when it is properly configured to do so. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Asterisk removes ice lines in sdp when calling between webrtc clients
Hi Joshua, Excellent! I didn't even remember to consider newer versions of asterisk as 11.11 was the latest one when I started building on. I have had libuuid and libuuid-devel installed the whole time, but perhaps 11.11 just did not see it there. I just installed 11.13 and it works perfectly. Thank you sir, I will raise a drink for you next time I'm out. cheers, Olli 2014-10-07 16:55 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Hi, Thanks Matthew for trying to reproduce the problem, I appreciate your efforts very much. There must be something off in my setup in one way or another. I could just discard this server and build a new one, but I think it's not good practice to leave a problem unsolved, so I'll continue trying to figure this out. One thing I noticed - don't know if it's relevant or not - due to a repo mismatch, I had problems with updating libgdiplus and libgdiplus-devel package, had to disable a repo and reinstall those and my mono installation (which is making me lose my hair). I would suggest using the latest version of 11 (as older versions will not work with current browsers). As well do you have the uuid development library installed? If not pjproject won't be built and you won't have ICE support which will yield exactly this result. -- 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 -- 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] Asterisk removes ice lines in sdp when calling between webrtc clients
Hi, Is there anything I can do with this problem? Re-installing Asterisk does not solve this and the problem still persists. Or is there any other logs or configurations I can provide to help figure out why Asterisk is removing lines from the sdp? Any ideas would be greatly appreciated! I also tried removing everything under /etc/asterisk/ and make samples to restore any errors I could have had in my configurations, then restoring my minimal configuration: asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and sip.conf. This did not help. (in case this message comes double, I just canceled posting of previous similar one as it was too big) cheers, Olli -- _ -- 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] Asterisk removes ice lines in sdp when calling between webrtc clients
Hi, Thanks Eric for your reply, yes I know Asterisk replaces the sdp, however it should create ice lines when calling to a webrtc client, which it is currently not doing. To recap my problem (check previous messages for details); I have 2 webrtc clients (sip.js on chrome) with realtime information that appears to be correct. When calling from A to B, INVITE coming to Asterisk contains correct sdp, but when the INVITE leaves Asterisk, the sdp lacks ice lines. cheers, Olli 2014-10-02 18:13 GMT+03:00 Eric Wieling ewiel...@nyigc.com: Asterisk is not a SIP Proxy. It is a B2BUA and will **always** replace the SDP with its own. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olli Heiskanen *Sent:* Thursday, October 02, 2014 9:06 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients Hi, Is there anything I can do with this problem? Re-installing Asterisk does not solve this and the problem still persists. Or is there any other logs or configurations I can provide to help figure out why Asterisk is removing lines from the sdp? Any ideas would be greatly appreciated! I also tried removing everything under /etc/asterisk/ and make samples to restore any errors I could have had in my configurations, then restoring my minimal configuration: asterisk.conf, extconfig.conf, extensions.conf, res_mysql.conf and sip.conf. This did not help. (in case this message comes double, I just canceled posting of previous similar one as it was too big) cheers, Olli -- _ -- 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] Asterisk removes ice lines in sdp when calling between webrtc clients
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all fields except
the number of the peer.
There's probably something I've changed that causes this behavior. Can
anyone tell me what's wrong in my configuration?
res_rtp_asterisk is included in the compilation and uuid-devel is
installed, Asterisk version is 11.11.0. Ice is enabled in rtp.conf as well
as in both clients in the realtime sip peer table.
Here's my realtime peer data:
*CLI realtime load sippeers name 660
Column Name Column Value
id 4
type friend
name 660
host dynamic
secret
encryption yes
avpf yes
icesupport yes ICE is enabled
ipaddr PU.BL.IC.IP
port 5060
regseconds 1410185500
defaultuser 660
fullcontact sip:6...@pu.bl.ic.ip:5060
lastms 0
useragent
context default
directmedia no
deny 0.0.0.0/0.0.0.0
permit PU.BL.IC.IP
nat force_rport,comedia
language
disallow
allow
force_avp yes
callerid
amaflags
mailbox
regexten
regserver
fromdomain testers.com
videosupport no
contactpermit
contactdeny
fullname 660 win8
hasvoicemail
subscribemwi
dtlsenable yes
dtlsverify no
dtlscertfile /etc/asterisk/keys/asterisk.pem
dtlsprivatekey /etc/asterisk/keys/asterisk.pem
dtlssetup actpass
sippasswd md5pwd
rpid
domain testers.com
sippasswd2
and my sip.conf:
[general]
bindport = 5070
bindaddr = PU.BL.IC.IP
udpbindaddr = PU.BL.IC.IP
tcpenable = yes
limitonpeers = yes
rtcachefriends = no
tos_sip=cs3
tos_audio=ef
realm = testers.com
autodomain=yes
domain=PU.BL.IC.IP
domain=testers.com
transport=ws,wss,udp
outboundproxy=PU.BL.IC.IP:5060
I'd appreciate Your advice.
cheers,
Olli
--
_
-- 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] Asterisk removes ice lines in sdp when calling between webrtc clients
) in new stack -- Executing [661@default:2] Dial(SIP/660-0007, SIP/661,3600,rt) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 18366 Adding codec 13 (ulaw) to SDP Adding codec 12 (gsm) to SDP Adding codec 14 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to PU.BL.IC.IP:5060: INVITE sip:6...@pu.bl.ic.ip:5060 SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport Max-Forwards: 70 From: 660 win8 sip:6...@testers.com;tag=as73376885 To: sip:6...@pu.bl.ic.ip:5060 Contact: sip:6...@pu.bl.ic.ip:5070 Call-ID: 2f70cc9567be50a46ba2879d4391a...@testers.com CSeq: 102 INVITE User-Agent: I Am the Devil Date: Mon, 08 Sep 2014 15:15:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 437 v=0 o=root 630896079 630896079 IN IP4 PU.BL.IC.IP s=Asterisk PBX 11.11.0 c=IN IP4 PU.BL.IC.IP t=0 0 m=audio 18366 RTP/SAVPF 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=connection:new a=setup:actpass a=fingerprint:SHA-256 CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05 a=sendrecv --- --- SIP read from UDP:PU.BL.IC.IP:5060 --- SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport=5070 From: 660 win8 sip:6...@testers.com;tag=as73376885 To: sip:6...@pu.bl.ic.ip:5060 Call-ID: 2f70cc9567be50a46ba2879d4391a...@testers.com CSeq: 102 INVITE Content-Length: 0 - --- (7 headers 0 lines) --- -- Called SIP/661 --- Transmitting (NAT) to PU.BL.IC.IP:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060 Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 Record-Route: sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes Record-Route: sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes From: 660 sip:6...@testers.com;tag=856i7ei98p To: sip:6...@testers.com;tag=as4298ec2e Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 INVITE Server: I Am the Devil Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:6...@pu.bl.ic.ip:5070 Content-Length: 0 --- SIP read from UDP:PU.BL.IC.IP:5060 --- SIP/2.0 404 No destinations Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport=5070 From: 660 win8 sip:6...@testers.com;tag=as73376885 To: sip:6...@pu.bl.ic.ip:5060;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78 Call-ID: 2f70cc9567be50a46ba2879d4391a...@testers.com CSeq: 102 INVITE Content-Length: 0 - --- (7 headers 0 lines) --- Transmitting (NAT) to PU.BL.IC.IP:5060: ACK sip:6...@pu.bl.ic.ip:5060 SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport Max-Forwards: 70 From: 660 win8 sip:6...@testers.com;tag=as73376885 To: sip:6...@pu.bl.ic.ip:5060;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78 Contact: sip:6...@pu.bl.ic.ip:5070 Call-ID: 2f70cc9567be50a46ba2879d4391a...@testers.com CSeq: 102 ACK User-Agent: I Am the Devil Content-Length: 0 --- Scheduling destruction of SIP dialog ' 2f70cc9567be50a46ba2879d4391a...@testers.com' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [661@default:3] Hangup(SIP/660-0007, ) in new stack == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-0007' Scheduling destruction of SIP dialog 'oc0ppijresm05k2emsgt' in 32000 ms (Method: INVITE) --- Reliably Transmitting (NAT) to PU.BL.IC.IP:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060 Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 From: 660 sip:6...@testers.com;tag=856i7ei98p To: sip:6...@testers.com;tag=as4298ec2e Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 INVITE Server: I Am the Devil Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- SIP read from UDP:PU.BL.IC.IP:5060 --- ACK sip:6...@testers.com SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0 Max-Forwards: 69 To: sip:6...@testers.com;tag=as4298ec2e From: 660 sip:6...@testers.com;tag=856i7ei98p Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 ACK Content-Length: 0 - --- (8 headers 0 lines) --- u363id562*CLI 2014-09-08 17:57 GMT+03:00 Matthew Jordan mjor...@digium.com: On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes
Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients
Daylight Time) | sip.transport | sending WebSocket message: SIP/2.0 100 Trying Via: SIP/2.0/WS PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.1 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070 To: sip:6...@pu.bl.ic.ip:5060 From: 660 win8 sip:6...@testers.com;tag=as40c12073 Call-ID: 597260a76cb0cb9155392f3a3c0be...@testers.com CSeq: 102 INVITE Supported: gruu,outbound Content-Length: 0 ... Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.inviteservercontext | invalid SDP sip-0.6.2.js:2655 Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.inviteservercontext | Failed to set remote offer sdp: Called with SDP without ice-ufrag and ice-pwd. sip-0.6.2.js:2655 Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | sending WebSocket message: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070 To: sip:6...@pu.bl.ic.ip:5060;tag=cl8lmb52gl From: 660 win8 sip:6...@testers.com;tag=as40c12073 Call-ID: 597260a76cb0cb9155392f3a3c0be...@testers.com CSeq: 102 INVITE Supported: gruu,outbound Content-Length: 0 ... Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | received WebSocket text message: ACK sip:v1vbuq35@0i03dp4lli27.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.0 Max-Forwards: 69 From: 660 win8 sip:6...@testers.com;tag=as40c12073 To: sip:6...@pu.bl.ic.ip:5060;tag=cl8lmb52gl Call-ID: 597260a76cb0cb9155392f3a3c0be...@testers.com CSeq: 102 ACK Content-Length: 0 Thanks, Olli 2014-09-08 18:50 GMT+03:00 Matthew Jordan mjor...@digium.com: On Mon, Sep 8, 2014 at 10:19 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hi Matthew, Here's the debug output: --- SIP read from UDP:PU.BL.IC.IP:5060 --- INVITE sip:6...@testers.com SIP/2.0 Record-Route: sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes Record-Route: sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0 Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 Max-Forwards: 69 To: sip:6...@testers.com From: 660 sip:6...@testers.com;tag=856i7ei98p Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 INVITE Contact: sip:6...@testers.com ;gr=urn:uuid:81780308-9304-4e37-984f-a2e864b17bd3;alias=CL.IE.NT.IP~47184~5;alias=CL.IE.NT.IP~47184~5 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Content-Type: application/sdp Supported: gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 1862 v=0 o=- 9082254022026432015 2 IN IP4 PU.BL.IC.IP s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx m=audio 10862 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 PU.BL.IC.IP a=candidate:3350409123 1 udp 2122194687 192.168.0.101 65339 typ host generation 0 a=candidate:3350409123 2 udp 2122194687 192.168.0.101 65339 typ host generation 0 a=candidate:2301678419 1 tcp 1518214911 192.168.0.101 0 typ host generation 0 a=candidate:2301678419 2 tcp 1518214911 192.168.0.101 0 typ host generation 0 a=candidate:1190865175 1 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0 a=candidate:1190865175 2 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0 a=ice-ufrag:7N23UxBo9XUgx9pJ a=ice-pwd:jL7AIeiJD5byGDSapfSftPRl a=ice-options:google-ice a=fingerprint:sha-256 93:E7:FC:E6:C2:74:71:2F:4F:81:43:7D:0C:A1:0F:C9:FC:3B:85:E6:44:2F:5A:39:05:79:DD:A6:0B:05:49:80 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:1842567493 cname:aJCyVX5iKPNU6Gf8 a=ssrc:1842567493 msid:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx 01a46fec-8a85-412d-9905-dcbefb8952b6 a=ssrc:1842567493 mslabel:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx a=ssrc:1842567493 label:01a46fec-8a85-412d-9905-dcbefb8952b6 a=sendrecv a=rtcp:10863 a=rtcp-mux a=candidate:GjpvUWJxlvHL7PZ6 1 UDP 1518214655 PU.BL.IC.IP 10862 typ host a=candidate:GjpvUWJxlvHL7PZ6 2 UDP 1518214654 PU.BL.IC.IP 10863 typ host - --- (16 headers 42 lines) --- Sending to PU.BL.IC.IP:5060 (no NAT) Sending to PU.BL.IC.IP:5060 (no NAT) Using INVITE request as basis request - oc0ppijresm05k2emsgt Found peer '660' for '660' from PU.BL.IC.IP:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 111 Found RTP audio format 103 Found RTP audio format 104 Found RTP audio
[asterisk-users] Asterisk rejects sdp from webrtc client
Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine, first one came directly from the client. I defined my clients according to the sip.js guide: http://sipjs.com/guides/server-configuration/asterisk/ So this was rejected: (I marked the extra lines with '//' to ease looking through the differences) v=0 o=- 9046935681162021751 2 IN IP4 91.221.66.61 s=- t=0 0 a=group:BUNDLE audio // a=msid-semantic: WMS Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF m=audio 11076 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 91.221.66.61 a=candidate:2999745851 1 udp 2122260223 192.168.56.1 52820 typ host generation 0 // a=candidate:2999745851 2 udp 2122260223 192.168.56.1 52820 typ host generation 0 // a=candidate:3350409123 1 udp 2122194687 192.168.0.101 52821 typ host generation 0 // a=candidate:3350409123 2 udp 2122194687 192.168.0.101 52821 typ host generation 0 // a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0 // a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0 // a=candidate:2301678419 1 tcp 1518214911 192.168.0.101 0 typ host generation 0 // a=candidate:2301678419 2 tcp 1518214911 192.168.0.101 0 typ host generation 0 // a=candidate:1190865175 1 udp 1685987071 91.145.67.22 52821 typ srflx raddr 192.168.0.101 rport 52821 generation 0 // a=candidate:1190865175 2 udp 1685987071 91.145.67.22 52821 typ srflx raddr 192.168.0.101 rport 52821 generation 0 // a=ice-ufrag:QJy1Fslu8ITGYl/d // a=ice-pwd:Q8N6+0PPj4CUG6leGAie7zaL // a=ice-options:google-ice // a=fingerprint:sha-256 CF:30:A7:7F:71:11:D4:5E:B0:E7:E3:F9:D8:C2:F4:60:2A:D0:76:46:F8:3A:97:01:C9:0C:5A:F7:B8:7D:C1:43 a=setup:actpass a=mid:audio // a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level // a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time // a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2179369454 cname:SvzCJjIAujxHGm9P a=ssrc:2179369454 msid:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF add6e533-c83d-42f2-b487-fcac8646ad32 a=ssrc:2179369454 mslabel:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF a=ssrc:2179369454 label:add6e533-c83d-42f2-b487-fcac8646ad32 a=sendrecv a=rtcp:11077 a=rtcp-mux a=candidate:hhsCc0ehS5qzXXxS 1 UDP 1518214655 91.221.66.61 11076 typ host a=candidate:hhsCc0ehS5qzXXxS 2 UDP 1518214654 91.221.66.61 11077 typ host And this was accepted as such: v=0 o=- 9046935681162021751 2 IN IP4 91.221.66.61 s=- t=0 0 a=msid-semantic: WMS Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF m=audio 11080 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 91.221.66.61 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2179369454 cname:SvzCJjIAujxHGm9P a=ssrc:2179369454 msid:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF add6e533-c83d-42f2-b487-fcac8646ad32 a=ssrc:2179369454 mslabel:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF a=ssrc:2179369454 label:add6e533-c83d-42f2-b487-fcac8646ad32 a=sendrecv a=rtcp:11081 a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RV9RgRP59zI6AoZKhGT4iq0Fj6A5tVbLy+zzj9JB a=setup:actpass a=fingerprint:sha-1 C2:D0:75:69:46:19:83:17:22:08:D4:8F:46:39:C7:AD:06:6A:CD:CC cheers, Olli -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
Hello, After having thought this through a bit I have some thoughts I'd like to share. In this case where the rtp profile is RTP/AVP Asterisk accepts and handles the call normally. If a webrtc client calls a sip client, or even another webrtc client, rtpengine is needed to step in (in my setup most of the clients would indeed be webrtc, but some of them might be sip). I think it would be better to use RTP/SAVPF throughout the process if both clients are webrtc (or otherwise speak RTP/SAVPF), but currently there is no way to accomplish this? Is it possible to configure Asterisk to only accept the RTP/SAVPF profile, and send 488 to all others? If it's not possible to force Asterisk to ignore rtp profiles (thus allowing the sdp be handled by rtpengine entirely), I'd prefer to use RTP/SAVPF or RTP/SAVP in the communication between Kamailio and Asterisk servers and use rtpengine to bridge to RTP/AVP and RTP/AVPF only if the client cannot speak securely. I'd very much like to hear opinions and thoughts on these. cheers, Olli 2014-08-13 20:39 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com: Aaa now I understood better, thanks! That's the instruction I used originally to write my Kamailio config, but I wasn't sure on how the sdp was supposed to be altered at which places in the whole SIP flow. I was thinking the original INVITE with the original sdp would go all the way to the receiving client, which return 488, which Kamailio would pick up and use rtpengine to alter the sdp at that point. So I'll need to alter the sdp every time before sending it to the Asterisk servers altogether and so avoid all the hassle I've been having with Asterisk. cheers, Olli 2014-08-13 20:07 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com: On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hi, Wow, thanks Paul, realizing the problem makes a lot of sense. So I setup Kamailio as a peer, but if I disable chan_sip module completely, I can't do it in sip.conf like I'd otherwise assume to do. I tried to rebuild Asterisk without chan_sip, but I guess that's not quite the way to go? Asterisk stopped sending back any sip messages so either there is a configuration means on how to do this or I'm doing something wrong with my current setup. My next thought was to compile Asterisk normally and set rtcachefriends to no, that did not work either, when dialing the cli stated: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) which I guess says Asterisk does not know where to send the message. The inner workings of Asterisk is a bit beyond me, if you don't mind giving advice on how to proceed I'd be most grateful. I think you are still mis-understanding me. I'll try to be clearer. From the POV of asterisk, you do still need chan_sip, however the only peer asterisk needs to be away of it Kamailio. All other peers will be stored within kamailio. This was the reason for my comment about realtime sip, you don't need it. Then, within kamailio, you'll need to invoke rtpengine using (rtpproxy-ng with kamailio 4.1) to rewrite the sdp for the invite to asterisk. You'll use the rtpproxy_offer and rtp_answer functions to remove ICE when calls originate from webrtc clients. Since you are not using a websocket in asterisk, it will just be a SIP over udp, the need for ICE and SAVPF is not needed. What you are trying to do is pretty complicated, it took me about 2 weeks to get everything setup properly. There is good information[1] on the web, you just need to google for it. [1] http://www.slideshare.net/crocodilertc/webrtc-websockets -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
Thanks Paul, I appreciate your thoughts. I understand your way, it's logical in your environment. I prefer to use LTS versions of Asterisk so I'm guessing what I want to do is not quite possible with Asterisk 11. I'd prefer my setup to work like this in different cases. webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- webrtc (rtp/savpf) sip (rtp/avp) -- kamailio -- rtpengine (rtp/savpf) -- asterisk -- kamailio -- rtpengine (rtp/avp) -- sip (rtp/avp) webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- rtpengine (rtp/avp) -- sip (rtp/avp) ... essentially, using RTP/AVP only when the client does not speak securely. It appears I'll have to try out the RTP/AVP way until there is an Asterisk that can accomplish this without having to use peer-specific settings. Down-side to this is that rtpengine needs resources from the server for webrtc clients even though both ends speak the same profile. It's not so complicated now that I know more on what Asterisk supports and how it handles the sdp, I just needed to learn by doing, testing and asking. I must be a bit ahead of my time for going for a RTP/SAVPF within my architecture, but using RTP/AVP is not such a bad option as srtp is on its way anyway in future Asterisk versions and the rtp flowing between Kamailio and users' networks are far more important than internal rtp traffic. cheers, Olli 2014-08-15 18:48 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com: On Fri, Aug 15, 2014 at 10:41 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, After having thought this through a bit I have some thoughts I'd like to share. In this case where the rtp profile is RTP/AVP Asterisk accepts and handles the call normally. If a webrtc client calls a sip client, or even another webrtc client, rtpengine is needed to step in (in my setup most of the clients would indeed be webrtc, but some of them might be sip). I think it would be better to use RTP/SAVPF throughout the process if both clients are webrtc (or otherwise speak RTP/SAVPF), but currently there is no way to accomplish this? Is it possible to configure Asterisk to only accept the RTP/SAVPF profile, and send 488 to all others? If it's not possible to force Asterisk to ignore rtp profiles (thus allowing the sdp be handled by rtpengine entirely), I'd prefer to use RTP/SAVPF or RTP/SAVP in the communication between Kamailio and Asterisk servers and use rtpengine to bridge to RTP/AVP and RTP/AVPF only if the client cannot speak securely. I'd very much like to hear opinions and thoughts on these. Again, I'll only share my experiences, but we do the complete opposite. Traffic between kamailio and asterisk is only RTP/AVP since the version of asterisk we are using does not support RTP/SAVPF (1.8). However, if you want RTP/SAVPF then honestly, you should just completely remove rtpengine from the picture since newer version of asterisk support both RTP/AVP and RTP/SAVPF (asterisk 12+). What I think you should do is go back to the basics, and document everything you want to do. Right now you have too many pieces in the puzzle and making the setup complicated. Like I said before, this is a complex setup and you need to start some place. Here is a diagram of what we do. webrtc (RTP/SAVPF) - kamailio - rtpengine - asterisk (RTP/AVP) This way, only RTP/AVP is in the core of our network. Rtpengine is on the edge (where it belongs), proxing rtp traffic. And, for us, we keep RTP/SAVPF outside of asterisk since support for it has been recently added. I also believe there are some open issue with dtls + srtp too. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
Hi, Wow, thanks Paul, realizing the problem makes a lot of sense. So I setup Kamailio as a peer, but if I disable chan_sip module completely, I can't do it in sip.conf like I'd otherwise assume to do. I tried to rebuild Asterisk without chan_sip, but I guess that's not quite the way to go? Asterisk stopped sending back any sip messages so either there is a configuration means on how to do this or I'm doing something wrong with my current setup. My next thought was to compile Asterisk normally and set rtcachefriends to no, that did not work either, when dialing the cli stated: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) which I guess says Asterisk does not know where to send the message. The inner workings of Asterisk is a bit beyond me, if you don't mind giving advice on how to proceed I'd be most grateful. cheers, Olli 2014-08-12 17:40 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com: On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Thank You Paul for your reply, The registrations in my setup are not duplicated, the 'secret' field in the realtime table is empty, which causes Asterisk to not authenticate requests from my Kamailio. Kamailio handles registrations, and also routes the traffic to Asterisk using dispatcher. Also, all peers have the Kamailio ip:port as outbound proxy so all traffic goes through Kamailio. That is your issue, stop using chan_sip with realtime (using data from kamailio). The only SIP peer asterisk should know of is kamailio, and your webrtc clients should be anonymous SIP users. This way, Asterisk doesn't even need to deal with websockets and RTP/SAVPF (this is what kamailio and rtpengine) is for. In your current setup, you are bypassing the functionality of rtpengine and not even leveraging it. Looks like version 11.11 works differently, I'll try to revert back to a previous version, and see if that works. I know at least the 'force_avp' field is new to 11.11 so it's safe to assume there's some difference between versions in rtp profile handling. It would be good to know how to handle this scenario in the new versions as well, I'll probably need to upgrade ahead anyway. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
Aaa now I understood better, thanks! That's the instruction I used originally to write my Kamailio config, but I wasn't sure on how the sdp was supposed to be altered at which places in the whole SIP flow. I was thinking the original INVITE with the original sdp would go all the way to the receiving client, which return 488, which Kamailio would pick up and use rtpengine to alter the sdp at that point. So I'll need to alter the sdp every time before sending it to the Asterisk servers altogether and so avoid all the hassle I've been having with Asterisk. cheers, Olli 2014-08-13 20:07 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com: On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hi, Wow, thanks Paul, realizing the problem makes a lot of sense. So I setup Kamailio as a peer, but if I disable chan_sip module completely, I can't do it in sip.conf like I'd otherwise assume to do. I tried to rebuild Asterisk without chan_sip, but I guess that's not quite the way to go? Asterisk stopped sending back any sip messages so either there is a configuration means on how to do this or I'm doing something wrong with my current setup. My next thought was to compile Asterisk normally and set rtcachefriends to no, that did not work either, when dialing the cli stated: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) which I guess says Asterisk does not know where to send the message. The inner workings of Asterisk is a bit beyond me, if you don't mind giving advice on how to proceed I'd be most grateful. I think you are still mis-understanding me. I'll try to be clearer. From the POV of asterisk, you do still need chan_sip, however the only peer asterisk needs to be away of it Kamailio. All other peers will be stored within kamailio. This was the reason for my comment about realtime sip, you don't need it. Then, within kamailio, you'll need to invoke rtpengine using (rtpproxy-ng with kamailio 4.1) to rewrite the sdp for the invite to asterisk. You'll use the rtpproxy_offer and rtp_answer functions to remove ICE when calls originate from webrtc clients. Since you are not using a websocket in asterisk, it will just be a SIP over udp, the need for ICE and SAVPF is not needed. What you are trying to do is pretty complicated, it took me about 2 weeks to get everything setup properly. There is good information[1] on the web, you just need to google for it. [1] http://www.slideshare.net/crocodilertc/webrtc-websockets -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
Hello, Thank You Paul for your reply, The registrations in my setup are not duplicated, the 'secret' field in the realtime table is empty, which causes Asterisk to not authenticate requests from my Kamailio. Kamailio handles registrations, and also routes the traffic to Asterisk using dispatcher. Also, all peers have the Kamailio ip:port as outbound proxy so all traffic goes through Kamailio. Looks like version 11.11 works differently, I'll try to revert back to a previous version, and see if that works. I know at least the 'force_avp' field is new to 11.11 so it's safe to assume there's some difference between versions in rtp profile handling. It would be good to know how to handle this scenario in the new versions as well, I'll probably need to upgrade ahead anyway. Thanks, Olli 2014-08-12 1:56 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com: On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to configure Asterisk to ignore the rtp profile but allow calls to pass with either of those profiles (even though clients might answer with 488 which would be caught and handled by Kamailio and rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and the second goal is to be able to add peers to the db table with similar data, as in no different values based on what kind of client wants to register. I'd like to allow the user to register using which ever client they choose (in this case one of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation (http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. We are successfully using kamailio / rtpengine with websockets and asterisk 1.8. First question is why are you duplicating registrations within asterisk? Secondly, why are you using websockets in asterisk? Without knowing more about your use case, I'll tell you how we did it. Like I said, kamailio is responsible for our SIP/ws subscribers and registrations. Once within kamailio we simply dispatch traffic to asterisk via SIP/udp. RTP is handled by rtpengine (using rtproxy-ng) and that is basically it. No special configuration is needed for asterisk (in fact 1.8 has no support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a kamailio peer and away you go. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
Okay, tried reverting to Asterisk 11.10.2. I didn't change the realtime table yet, but now when calling from websocket client to another websocket client, cli says: WARNING[30620][C-]: chan_sip.c:11056 process_sdp_a_dtls: Unsupported fingerprint hash type 'sha-2' received on dialog '36ns50nk1fo04pu3m7lf' WARNING[30620][C-]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 10640 RTP/SAVPF 111 103 104 0 8 106 105 13 126 This many times, until the forking capacity of Kamailio has been reached and call fails. The clients are running on chrome, and calls have worked before... I wonder if I should revert further back and/or change or remove some realtime table fields? cheers, Olli 2014-08-12 11:17 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com: Hello, Thank You Paul for your reply, The registrations in my setup are not duplicated, the 'secret' field in the realtime table is empty, which causes Asterisk to not authenticate requests from my Kamailio. Kamailio handles registrations, and also routes the traffic to Asterisk using dispatcher. Also, all peers have the Kamailio ip:port as outbound proxy so all traffic goes through Kamailio. Looks like version 11.11 works differently, I'll try to revert back to a previous version, and see if that works. I know at least the 'force_avp' field is new to 11.11 so it's safe to assume there's some difference between versions in rtp profile handling. It would be good to know how to handle this scenario in the new versions as well, I'll probably need to upgrade ahead anyway. Thanks, Olli 2014-08-12 1:56 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com: On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to configure Asterisk to ignore the rtp profile but allow calls to pass with either of those profiles (even though clients might answer with 488 which would be caught and handled by Kamailio and rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and the second goal is to be able to add peers to the db table with similar data, as in no different values based on what kind of client wants to register. I'd like to allow the user to register using which ever client they choose (in this case one of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation (http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. We are successfully using kamailio / rtpengine with websockets and asterisk 1.8. First question is why are you duplicating registrations within asterisk? Secondly, why are you using websockets in asterisk? Without knowing more about your use case, I'll tell you how we did it. Like I said, kamailio is responsible for our SIP/ws subscribers and registrations. Once within kamailio we simply dispatch traffic to asterisk via SIP/udp. RTP is handled by rtpengine (using rtproxy-ng) and that is basically it. No special configuration is needed for asterisk (in fact 1.8 has no support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a kamailio peer and away you go. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to configure Asterisk to ignore the rtp profile but allow calls to pass with either of those profiles (even though clients might answer with 488 which would be caught and handled by Kamailio and rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and the second goal is to be able to add peers to the db table with similar data, as in no different values based on what kind of client wants to register. I'd like to allow the user to register using which ever client they choose (in this case one of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation ( http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. Here's my sip.conf: bindport = 5070 ;Kamailio is at port 5060, and it's always used as outbound proxy bindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = yes rtupdate=yes tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=no domain=testers.com allowexternaldomains=no allowguest=no ;avpf=yes ; encryption=yes transport=ws,wss,udp icesupport=yes srvlookup=yes nat=force_rport,comedia videosupport=yes directmedia=no And here's the way I've defined my websocket peer to my sippeers table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407744248 defaultuser: 660 fullcontact: sip:6...@pu.bl.ic.ip:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP secret: NULL md5secret: NULL avpf: yes force_avp: yes icesupport: yes directmedia: yes encryption: yes nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL setvar: NULL callerid: NULL amaflags: NULL videosupport: no maxcallbitrate: NULL mailbox: NULL regexten: NULL fromdomain: testers.com fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: md5ofmypwd rpid: NULL domain: testers.com sippasswd2: NULL This is how all other clients are currently defined: id: 7 name: 771 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407748788 defaultuser: 771 fullcontact: sip:7...@pu.bl.ic.ip:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP secret: NULL md5secret: NULL avpf: no force_avp: NULL icesupport: NULL directmedia: yes encryption: NULL nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL setvar: NULL callerid: NULL amaflags: NULL videosupport: NULL maxcallbitrate: NULL mailbox: NULL regexten: NULL fromdomain: testers.com fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: NULL dtlsenable: NULL dtlsverify: NULL dtlsprivatekey: NULL dtlssetup: NULL dtlscertfile: NULL dtlscafile: NULL sippasswd: 27e13af7c596313350986c58c9d24946 rpid: NULL domain: testers.com sippasswd2: NULL cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar
Re: [asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Hi, Thanks Daniel for your reply. Sorry for having been a bit obscure, it is my intention to have all clients able to call each other, regardless of which ua client software they use. I think I've realized what's going on. My goal is to use rtpengine to bridge between rtp profiles when they are different. But according to sip.js instruction, I set up my clients in a way that Asterisk took the place of rtpengine and changed the rtp profiles along the way based on the realtime table values. That got me confused but now I know at least what the problem is so I can fix it. This setup works in a way that I can make calls between websocket and sip clients, but the problem with it is that I need different values in the realtime table, according to which rtp profile the client uses. Doing this I made a wrong turn in my project, I'll need to have universal setup for each peer so the user can use a websocket client or a sip client to register and use an account. I'll still need to figure out which settings to use and which not to use, so the rtp gets handled by rtpengine, not Asterisk. But that's a question for the Asterisk list. The problem about Asterisk setting the rtp profile as UDP/TLS/RTP/SAVPF was fixed using a peer setting in the realtime table, now Asterisk accepts RTP/SAVPF I can have calls flowing as soon as I can get rtpengine to cooperate with me. I wonder, is there UDP/TLS/RTP/SAVPF handling in rtpengine/kamailio? I may have to add some kind of handling to this if I have to revert back to my previous settings. cheers, Olli 2014-08-05 16:49 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com: On 01/08/14 10:56, Olli Heiskanen wrote: Hi, I got ahead with my setup, this post helped me much: http://forums.digium.com/viewtopic.php?f=1t=90167sid= 66fdf8cc4be5d955ba584e989a23442f At least the avpf setting had to be removed from sip.conf and put in the realtime db table, defined per client. I left the encryption setting in sip.conf. I had some problems calling from SIP client to another, then had to define avpf=no for those clients. Personally I don't like to use different settings to different clients, is there a way around this? With this setup I can make calls between SIP clients but not ws clients. My client (now I use sip.js) fails to parse the sdp - including the apparently correct rtp profile UDP/TLS/RTP/SAVPF - and sends back 488, which makes the call fail. I'd like to hear opinions from you guys which would be the correct place to handle this? My setup has Asterisk Kamailio realtime integration, and I use dispatcher in Kamailio to route calls to Asterisk. Kamailio sounds like the logical place, but I'd rather find a way to not change the rtp profile along the way, at least until the clients can support that one. To understand properly, you don't want to use rtpenging for srtp(webrtc)-rtp(classic sip) gatewaying? If yes, maybe you can partition the users (classic-sip and webrtc-sip), then use two asterisk instances with routing via kamailio. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA -- _ -- 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] From and To headers contain same account in INVITEs
Hello,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.
Can you guys tell me what might be causing this? I have 6...@testers.com as
a websocket client and 7...@testers.com (caller) using a Zoiper client (db
output below). The call itself works, audio and all, only those headers are
puzzling to me. I noticed this when I tried to add a label saying '700
calling' on my web page. The same thing happens when I call from 660 to
700.
My Asterisk is 11.11.0 running on CentOS 6.5.
An INVITE is sent from my client to Kamailio and then to Asterisk:
(both Kamailio and Asterisk are at 1.1.1.1)
INVITE sip:6...@testers.com;transport=UDP SIP/2.0
Record-Route: sip:1.1.1.1;lr=on;ftag=fd070807
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKf6e9.339dda0648d95af665c91db701754d98.0
Via: SIP/2.0/UDP 2.2.2.2:37730
;rport=37730;branch=z9hG4bK-d8754z-7f27c9fc35574abb-1---d8754z-
Max-Forwards: 16
Contact: sip:700@2.2.2.2:37730;transport=UDP
To: sip:6...@testers.com;transport=UDP
From: sip:7...@testers.com;transport=UDP;tag=fd070807
Call-ID: ZDc0YjU1ZjNmMWI5YjUyYzY0YWNjN2NjN2NkODg2OTk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 239
v=0
o=Z 0 0 IN IP4 2.2.2.2
s=Z
c=IN IP4 2.2.2.2
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
... and Asterisk responds with Trying:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKf6e9.339dda0648d95af665c91db701754d98.0;received=1.1.1.1;rport=5060
Via: SIP/2.0/UDP 2.2.2.2:37730
;rport=37730;branch=z9hG4bK-d8754z-7f27c9fc35574abb-1---d8754z-
Record-Route: sip:1.1.1.1;lr=on;ftag=fd070807
From: sip:7...@testers.com;transport=UDP;tag=fd070807
To: sip:6...@testers.com;transport=UDP
Call-ID: ZDc0YjU1ZjNmMWI5YjUyYzY0YWNjN2NjN2NkODg2OTk.
CSeq: 2 INVITE
Server: I Am the Devil
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:660@1.1.1.1:5070
Content-Length: 0
And when Asterisk sends out the INVITE, From and To headers both have the
same number:
INVITE sip:660@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5070;branch=z9hG4bK75de61d0;rport
Max-Forwards: 70
From: sip:6...@testers.com;tag=as7b7c32a5
To: sip:660@1.1.1.1:5060
Contact: sip:660@1.1.1.1:5070
Call-ID: 7240b8a011890ec677f185f454858...@testers.com
CSeq: 102 INVITE
User-Agent: I Am the Devil
Date: Wed, 06 Aug 2014 09:54:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 801
v=0
o=root 969416519 969416519 IN IP4 1.1.1.1
s=Asterisk PBX 11.11.0
c=IN IP4 1.1.1.1
t=0 0
m=audio 18740 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:50d777041673316422560b90281fcd2e
a=ice-pwd:0093fdde724f8a411742661c31c90f21
a=candidate:H5bdd423d 1 UDP 2130706431 1.1.1.1 18740 typ host
a=candidate:S5bdd423d 1 UDP 1694498815 1.1.1.1 18740 typ srflx
a=candidate:H5bdd423d 2 UDP 2130706430 1.1.1.1 18741 typ host
a=candidate:S5bdd423d 2 UDP 1694498814 1.1.1.1 18742 typ srflx
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05
a=sendrecv
Here's the dialplan, nothing special:
exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
same = n,Dial(SIP/${EXTEN},3600,rt)
same = n,Hangup
And here's how the clients are set in my db:
id: 4
name: 660
ipaddr: 1.1.1.1
port: 5060
regseconds: 1407320692
defaultuser: 660
fullcontact: sip:660@1.1.1.1:5060
regserver:
useragent:
lastms: 0
host: dynamic
type: friend
context: default
deny: 0.0.0.0/0.0.0.0
permit: 1.1.1.1
secret: NULL
md5secret:
Re: [asterisk-users] From and To headers contain same account in INVITEs
Hi, There we go, that was it. Thank you Joshua! cheers, Olli 2014-08-06 15:26 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Hello, Kia ora, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 6...@testers.com mailto:6...@testers.com as a websocket client and 7...@testers.com mailto:7...@testers.com (caller) using a Zoiper client (db output below). The call itself works, audio and all, only those headers are puzzling to me. I noticed this when I tried to add a label saying '700 calling' on my web page. The same thing happens when I call from 660 to 700. Your configuration has fromuser set which explicitly sets the user portion of the From header to what you specify. This is commonly used for ITSPs as they use that to determine who you are trying to authenticate as. If you require this to be set then caller id information has to be transported in a different manner (RPID or PAI). Cheers, -- 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 -- 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] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Hi, I got ahead with my setup, this post helped me much: http://forums.digium.com/viewtopic.php?f=1t=90167sid=66fdf8cc4be5d955ba584e989a23442f At least the avpf setting had to be removed from sip.conf and put in the realtime db table, defined per client. I left the encryption setting in sip.conf. I had some problems calling from SIP client to another, then had to define avpf=no for those clients. Personally I don't like to use different settings to different clients, is there a way around this? With this setup I can make calls between SIP clients but not ws clients. My client (now I use sip.js) fails to parse the sdp - including the apparently correct rtp profile UDP/TLS/RTP/SAVPF - and sends back 488, which makes the call fail. I'd like to hear opinions from you guys which would be the correct place to handle this? My setup has Asterisk Kamailio realtime integration, and I use dispatcher in Kamailio to route calls to Asterisk. Kamailio sounds like the logical place, but I'd rather find a way to not change the rtp profile along the way, at least until the clients can support that one. cheers, Olli 2014-07-26 12:58 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com: Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as Kamailio. The version is 11.10.2. With Kamailio I use rtpengine, which affects SDP descriptions when 488 response is received. My goal is to enable two websocket clients using Chrome to call each other, using Kamailio as outbound proxy. Kamailio routes signaling to Asterisk, and then back to clients. Currently the problem is RTP, when INVITE is received from client A to Kamailio, it is relayed to Asterisk. Asterisk responds with 488 Not Acceptable here and the cli says: NOTICE[11642][C-0006]: chan_sip.c:10124 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126 WARNING[11642][C-0006]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126 Strange thing is, I don't know why Asterisk says AVPF is not enabled. The warning about rejecting the audio stream must be behind the 488 response but I didn't find any answers that would solve my case so I must turn to you guys. In my sip.conf I have savpf=yes, but is there something else I need to enable or change in the configs or change my peer configurations? I'm not sure if this is relevant but I checked that Asterisk was successfully compiled with res_srtp module. Here's my sip.conf contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain=PU.BL.IC.IP domain=testers.com allowexternaldomains=no allowguest=no avpf=yes encryption=yes transport=ws,udp icesupport=yes srvlookup=yes And here's an example of a ws client in my realtime peer table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1406368294 defaultuser: 660 fullcontact: sip:6...@pu.bl.ic.ip:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP secret: NULL md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: NULL nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL insecure: NULL trustrpid: NULL progressinband: NULL promiscredir: NULL useclientcode: NULL accountcode: NULL setvar: NULL callerid: NULL amaflags: NULL callcounter: NULL busylevel: NULL allowoverlap: NULL allowsubscribe: NULL videosupport: NULL maxcallbitrate: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL t38pt_usertpsource: NULL regexten: NULL fromdomain: testers.com fromuser: 660 qualify: NULL defaultip: NULL rtptimeout: NULL rtpholdtimeout: NULL sendrpid: NULL
[asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as Kamailio. The version is 11.10.2. With Kamailio I use rtpengine, which affects SDP descriptions when 488 response is received. My goal is to enable two websocket clients using Chrome to call each other, using Kamailio as outbound proxy. Kamailio routes signaling to Asterisk, and then back to clients. Currently the problem is RTP, when INVITE is received from client A to Kamailio, it is relayed to Asterisk. Asterisk responds with 488 Not Acceptable here and the cli says: NOTICE[11642][C-0006]: chan_sip.c:10124 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126 WARNING[11642][C-0006]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126 Strange thing is, I don't know why Asterisk says AVPF is not enabled. The warning about rejecting the audio stream must be behind the 488 response but I didn't find any answers that would solve my case so I must turn to you guys. In my sip.conf I have savpf=yes, but is there something else I need to enable or change in the configs or change my peer configurations? I'm not sure if this is relevant but I checked that Asterisk was successfully compiled with res_srtp module. Here's my sip.conf contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain=PU.BL.IC.IP domain=testers.com allowexternaldomains=no allowguest=no avpf=yes encryption=yes transport=ws,udp icesupport=yes srvlookup=yes And here's an example of a ws client in my realtime peer table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1406368294 defaultuser: 660 fullcontact: sip:6...@pu.bl.ic.ip:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP secret: NULL md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: NULL nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL insecure: NULL trustrpid: NULL progressinband: NULL promiscredir: NULL useclientcode: NULL accountcode: NULL setvar: NULL callerid: NULL amaflags: NULL callcounter: NULL busylevel: NULL allowoverlap: NULL allowsubscribe: NULL videosupport: NULL maxcallbitrate: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL t38pt_usertpsource: NULL regexten: NULL fromdomain: testers.com fromuser: 660 qualify: NULL defaultip: NULL rtptimeout: NULL rtpholdtimeout: NULL sendrpid: NULL outboundproxy: PU.BL.IC.IP timert1: NULL timerb: NULL qualifyfreq: NULL constantssrc: NULL contactpermit: NULL contactdeny: NULL usereqphone: NULL textsupport: NULL faxdetect: NULL buggymwi: NULL auth: NULL fullname: NULL trunkname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL autoframing: NULL rtpkeepalive: NULL call-limit: NULL g726nonstandard: NULL ignoresdpversion: NULL allowtransfer: NULL dynamic: NULL path: NULL supportpath: NULL sippasswd: my-md5-pwd rpid: NULL domain: testers.com sippasswd2: NULL I'd greatly appreciate help on this! cheers, Olli -- _ -- 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:
Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper
Hello, I noticed something that might be a result from the fix suggested here, so I'll continue a bit on this thread. After removing the callbackextension field from my realtime sip peer table, the following started happening: I issued command 'sip reload' on the cli and get the following warning: WARNING[24427]: res_config_mysql.c:501 realtime_multi_mysql: MySQL RealTime: Failed to query database: Unknown column 'callbackextension' in 'where clause' This must be a result from removing that field from the db, but somewhere in the code there is a select statement where the callbackextension field is used in the where clause, resulting to the above warning. I wonder if this something to be worried about, or is going to cause problems later? My goal is of coure just to handle calls, save cdrs, do pbx features etc with this asterisk. cheers, Olli 2014-07-15 16:56 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com: Wow, thanks Joshua, it would've taken me forever to find the answer there. It did the trick and the registrations look much better. Merci beaucoup! - Olli 2014-07-15 16:26 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Thanks, there are no register lines in my sip.conf, but I have defined callbackextension fields in the realtime table, to be the same value as the extension name. In this case, extension 771 has callbackextension value 771. I tried replacing those with null values but that had no effect on the outcome. The callbackextension is the reason this is happening. From sip.conf.sample: ; A similar effect can be achieved by adding a callbackextension option in a peer section. ; this is equivalent to having the following line in the general section: ; ;register = username:secret@host/callbackextension ; ; and more readable because you don't have to write the parameters in two places ; (note that the port is ignored - this is a bug that should be fixed). Remove that column from your table, restart Asterisk, and it should go away. -- 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 -- 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] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper
Hello all, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set 'Subscribe to MWI' value to 'both', after a normal, successful registration Asterisk begins to send REGISTER messages to Kamailio every 105 seconds. Kamailio responds with 200 OK. If I set the value to 'disabled', Asterisk does not send these frequent REGISTER messages. Probably due to these REGISTERs Kamailio sees 2 AORs for the account for those clients whose 'Subscribe to MWI' setting is defined as 'both'. I know this is mostly not an Asterisk problem, but I'd like to understand better what exactly makes Asterisk to react this way? I didn't find any differences in the SIP messages during registration (I could be just blind though...), or in the way the clients are set up in the Realtime db. On the 'Subscribe to MWI' setting the Zoiper documentation states: this tag specifies when Zoiper is going to subscribe for Message Waiting Indication(MWI) for this account. In addition to values 'both' and 'disabled' there are values 'before registration (Asterisk)' and 'after registration'. To me it seems strange to use REGISTER messages for subscribing to something related to voicemail messages. So far I haven't learned about the way Asterisk handles voicemail stuff but if You guys have some clarification on why I'm getting these results I'd appreciate it! cheers, Olli -- _ -- 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] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper
Hello, Thanks for your response, I actually verified that the Zoiper setting is not the reason for Asterisk to start sending REGISTERs, it only looked like it as I checked the Kamailio output before Asterisk sent the first REGISTER to Kamailio, right after I had played with that setting. (sorry, my bad!) However, _something_ is causing these REGISTERs, here's an example of a REGISTER message sent from Asterisk to Kamailio: REGISTER sip:testers.com SIP/2.0 Via: SIP/2.0/UDP my_ip:5070;branch=z9hG4bK7477f754;rport Max-Forwards: 70 From: sip:771@my_ip;tag=as7a88c4c6 To: sip:771@my_ip Call-ID: 3e946958322b1e2d6bfa564d46bf8...@testers.com CSeq: 121 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:771@91.221.66.61:5070 Content-Length: 0 Is there any other reason - other than client settings - why this would happen? cheers, Olli 2014-07-15 15:40 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Hello all, Bonjour, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set 'Subscribe to MWI' value to 'both', after a normal, successful registration Asterisk begins to send REGISTER messages to Kamailio every 105 seconds. Kamailio responds with 200 OK. If I set the value to 'disabled', Asterisk does not send these frequent REGISTER messages. Probably due to these REGISTERs Kamailio sees 2 AORs for the account for those clients whose 'Subscribe to MWI' setting is defined as 'both'. Can you provide a link to a sip debug log of this occurring? It sounds extremely weird and I'm not really sure how chan_sip would be doing such a thing... Cheers, -- 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 -- 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] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper
Thanks, there are no register lines in my sip.conf, but I have defined callbackextension fields in the realtime table, to be the same value as the extension name. In this case, extension 771 has callbackextension value 771. I tried replacing those with null values but that had no effect on the outcome. Currently when I register clients in, after some seconds Asterisk starts sending REGISTER messages, at which point Kamailio sees 2 AORs, here's an example: (here 1.1.1.1 is the public ip of my server that houses Kamailio at 5060 and Asterisk at 5070, and 2.2.2.2 is the public ip of the network clients are in) AOR:: 7...@testers.com Contact:: sip:771@2.2.2.2:5060;rinstance=c8447637c890c010;transport=UDP Q= Expires:: 3470 Callid:: NDQ5Njk4ZmUxZGJhNzRjMzUwMTA2OThmOGFjYzc4Zjk. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35b0-608 Reg-Id:: 0 Last-Keepalive:: 1405429865 Last-Modified:: 1405429865 AOR:: 771@1.1.1.1 Contact:: sip:771@1.1.1.1:5070 Q= Expires:: 105 Callid:: 3e946958322b1e2d6bfa564d46bf8...@testers.com Cseq:: 133 User-agent:: Asterisk PBX State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35b0-708 Reg-Id:: 0 Last-Keepalive:: 1405429980 Last-Modified:: 1405429980 I guess there should be only one AOR, so Asterisk might get wrong kind of data to begin with or it's configured incorrectly. In my sip trace the REGISTER flow from client to Kamailio to Asterisk seems ok, I could be wrong though. In my setup clients authenticate with Kamailio and Kamailio sends a REGISTER to Asterisk according to guide I used: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb How would I fix this double-AOR problem, can it be fixed on Asterisk configuration? thanks, Olli 2014-07-15 16:00 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Hello, Thanks for your response, I actually verified that the Zoiper setting is not the reason for Asterisk to start sending REGISTERs, it only looked like it as I checked the Kamailio output before Asterisk sent the first REGISTER to Kamailio, right after I had played with that setting. (sorry, my bad!) However, _something_ is causing these REGISTERs, here's an example of a REGISTER message sent from Asterisk to Kamailio: REGISTER sip:testers.com http://testers.com SIP/2.0 Via: SIP/2.0/UDP my_ip:5070;branch=z9hG4bK7477f754;rport Max-Forwards: 70 From: sip:771@my_ip;tag=as7a88c4c6 To: sip:771@my_ip Call-ID: 3e946958322b1e2d6bfa564d46bf8...@testers.com mailto:3e946958322b1e2d6bfa564d46bf8...@testers.com CSeq: 121 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:771@91.221.66.61:5070 http://sip:771@91.221.66.61:5070 Content-Length: 0 Is there any other reason - other than client settings - why this would happen? If Asterisk was configured to do so, yes. Do you have any register lines in sip.conf or do you have the callbackextension option set for any peers? -- 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 -- 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] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper
Wow, thanks Joshua, it would've taken me forever to find the answer there. It did the trick and the registrations look much better. Merci beaucoup! - Olli 2014-07-15 16:26 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Thanks, there are no register lines in my sip.conf, but I have defined callbackextension fields in the realtime table, to be the same value as the extension name. In this case, extension 771 has callbackextension value 771. I tried replacing those with null values but that had no effect on the outcome. The callbackextension is the reason this is happening. From sip.conf.sample: ; A similar effect can be achieved by adding a callbackextension option in a peer section. ; this is equivalent to having the following line in the general section: ; ;register = username:secret@host/callbackextension ; ; and more readable because you don't have to write the parameters in two places ; (note that the port is ignored - this is a bug that should be fixed). Remove that column from your table, restart Asterisk, and it should go away. -- 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 -- 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] Realtime integration: Unregistered clients showing as registered?
Hello,
Thank you for your response.
Actually, I managed to solve a part of the problem; as I use Kamailio to
handle authentication, problem was that even though authentication went ok
through Kamailio, Asterisk refused to accept messages from Kamailio. That's
why Asterisk sent the 401. I think I had incorrect values in the realtime
sippeers table rows, and also I had to add values to deny and permit
fields, which in fact were in the wrong order. So no wonder I was having
problems with authentication! (and yes, I do know how digest authentication
works ;))
I fixed the deny values to 0.0.0.0/0.0.0.0 and permit value to Kamailio ip.
Even after this I had problems having Asterisk accept the authentications.
Asterisk cli was saying:
ERROR[20605]: chan_sip.c:30790 build_peer: Bad ACL entry in configuration
line 0 : kamailioip:5060
... that was because I had tried to define kamailio ip with port, as
Kamailio and Asterisk are on the same machine, but removing the port solved
that (not sure but I guess it is good I use 5060 for Kamailio and 5070 for
Asterisk instead of vice versa, perhaps this solution wouldn't work then).
Then I found that I had to add values to fields: nat (to force_rport) and
defaultip (to 0.0.0.0), and only after that I got Asterisk to see the
registered peers. So now everything looks ok from both Asterisk and
Kamailio when it comes to authentication.
I still can't get calls going though, in the asterisk cli I get 'Everyone
is busy/congested at this time', so I'm going to continue investigating
that. If you guys have good advice for me at this time I'll be most happy
to take them.
cheers,
Olli
2014-05-15 17:17 GMT+03:00 Leandro Dardini ldard...@gmail.com:
It is the way it works. First the phone sends a REGISTER without any
password. Asterisk answers with a Unauthorized and provide a nonce to be
used for the next registration attempt, using it to encrypt the password.
Leandro
2014-05-14 13:12 GMT+02:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
:
Hello,
After a small break from working on this, I got the idea of tcpdumping
the correct ports. What I see is REGISTER messages from Kamailio port to
Asterisk, which are replied with 401 Unauthorized. Why is this happening?
In my sippeers table the secret field has no value (tried both NULL and
empty string) and the added field sippasswd has the correct password for
the user.
The above might be the cause of my problem, would anyone be able to
advice me to get to correct behaviour? Now Kamailio sees the clients as
registered, which would be wrong if Asterisk doesn't.
cheers,
Olli
2014-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
:
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime
load sipusers name 660 shows the user data. Field regseconds has a value
and fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as
they are on the same machine).
I have a very simple dialplan:
[general]
[default]
exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
same = n,Dial(SIP/${EXTEN},3600,rt)
same = n,Hangup
Here's more on my problem and background to it, guys on the Kamailio
list helped out but looks like I need to check my Asterisk configuration.
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html
My goal is to have all clients in the asterisk database, asterisk (one
at this point, several later) handling the calls and Kamailio as proxy. In
Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
domain 'testers.com'.
I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
the same rental virtual server. Clients are in my home network behind nat.
In MySQL I have database asterisk with table sippeers, where I have
clients added like this:
INSERT INTO sippeers
(name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
','660','friend');
In this message there are some outputs and a sip trace of a register:
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html
What I don't know is how to configure sip.conf, so far I've just been
making guesses based on online examples and documentation.
My current sip.conf looks like this:
[general]
bindport = 5070
bindaddr = 127.0.0.1
tcpbindaddr = 127.0.0.1:5070
tcpenable = no
limitonpeers = yes
;rtcachefriends = yes
tos_sip=cs3
tos_audio=ef
realm = testers.com
I've tried defining realm and domain values, but I lack proper
understanding of those. Can you guys help me out? Are there any other
configurations I need to check?
Respectfully
Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?
Hello,
After a small break from working on this, I got the idea of tcpdumping the
correct ports. What I see is REGISTER messages from Kamailio port to
Asterisk, which are replied with 401 Unauthorized. Why is this happening?
In my sippeers table the secret field has no value (tried both NULL and
empty string) and the added field sippasswd has the correct password for
the user.
The above might be the cause of my problem, would anyone be able to advice
me to get to correct behaviour? Now Kamailio sees the clients as
registered, which would be wrong if Asterisk doesn't.
cheers,
Olli
2014-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the user data. Field regseconds has a value and
fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as they
are on the same machine).
I have a very simple dialplan:
[general]
[default]
exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
same = n,Dial(SIP/${EXTEN},3600,rt)
same = n,Hangup
Here's more on my problem and background to it, guys on the Kamailio list
helped out but looks like I need to check my Asterisk configuration.
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html
My goal is to have all clients in the asterisk database, asterisk (one at
this point, several later) handling the calls and Kamailio as proxy. In
Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
domain 'testers.com'.
I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
the same rental virtual server. Clients are in my home network behind nat.
In MySQL I have database asterisk with table sippeers, where I have
clients added like this:
INSERT INTO sippeers
(name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
','660','friend');
In this message there are some outputs and a sip trace of a register:
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html
What I don't know is how to configure sip.conf, so far I've just been
making guesses based on online examples and documentation.
My current sip.conf looks like this:
[general]
bindport = 5070
bindaddr = 127.0.0.1
tcpbindaddr = 127.0.0.1:5070
tcpenable = no
limitonpeers = yes
;rtcachefriends = yes
tos_sip=cs3
tos_audio=ef
realm = testers.com
I've tried defining realm and domain values, but I lack proper
understanding of those. Can you guys help me out? Are there any other
configurations I need to check?
Respectfully,
Olli
--
_
-- 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] Asterisk 11.9 with webRTC demo integration
Hello, I'm far from being an expert, but as far as I know when you use https in your website the browser will ask to use the audio devices only once and then remembers your decision. When using http it will ask every time. Sorry I can't be of more help but hope this helps. cheers, Olli 2014-05-10 10:27 GMT+03:00 bhavik patel bhavikpatel14...@gmail.com: Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support. I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions. I have tried both browser Google Chrome and Firefox with their latest versions. For asterisk, I made some configuration like below. Please check : http://pastebin.com/7KCvtcNf For Outbound calls : when i am dialling 8002 - 8001 every time Chrome Browser asking for allow microphone. Is there any way to disable asking permission and allowing it by default ? when i allow microphone then SIpml5 phone showing like Not Allow. Here is the asterisk logs : http://pastebin.com/JZeDjyay For Incoming calls : When call come to browser,And allow microphone then Call rejected and asterisk showing like Got SIP response 603 Failed to get local SDP in asterisk CLI. But After some google i found new link https://code.google.com/p/sipml5/wiki/Downloads for SIPml-api.js and after replacing that JS File Calls are comming in browser even i am able to answer that calls,Also in browser it says In call but in asterisk CLI it keep showing ringing and other end showing like remote ringing . Here is the asterisk logs : http://pastebin.com/e8Ap3bhq Can anyone please let me know what am i doing wrong? -- Thanks, Bhavik Patel -- _ -- 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] Realtime integration: Unregistered clients showing as registered?
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the user data. Field regseconds has a value and
fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as they
are on the same machine).
I have a very simple dialplan:
[general]
[default]
exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
same = n,Dial(SIP/${EXTEN},3600,rt)
same = n,Hangup
Here's more on my problem and background to it, guys on the Kamailio list
helped out but looks like I need to check my Asterisk configuration.
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html
My goal is to have all clients in the asterisk database, asterisk (one at
this point, several later) handling the calls and Kamailio as proxy. In
Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
domain 'testers.com'.
I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
the same rental virtual server. Clients are in my home network behind nat.
In MySQL I have database asterisk with table sippeers, where I have clients
added like this:
INSERT INTO sippeers
(name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
','660','friend');
In this message there are some outputs and a sip trace of a register:
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html
What I don't know is how to configure sip.conf, so far I've just been
making guesses based on online examples and documentation.
My current sip.conf looks like this:
[general]
bindport = 5070
bindaddr = 127.0.0.1
tcpbindaddr = 127.0.0.1:5070
tcpenable = no
limitonpeers = yes
;rtcachefriends = yes
tos_sip=cs3
tos_audio=ef
realm = testers.com
I've tried defining realm and domain values, but I lack proper
understanding of those. Can you guys help me out? Are there any other
configurations I need to check?
Respectfully,
Olli
--
_
-- 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
