sr-users October 2008

sr-users@lists.kamailio.org

93 participants
170 discussions

Re: [Kamailio-Users] Users Digest, Vol 41, Issue 24
by luzango mfupe 09 Oct '08

09 Oct '08

Ok David, will give feedback soon.rgds, Lu. On Thu, Oct 9, 2008 at 11:53 AM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person … [View More]managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: No audio both ways. (David Villasmil) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 9 Oct 2008 11:07:12 +0200 > From: "David Villasmil" <david.villasmil.work(a)gmail.com> > Subject: Re: [Kamailio-Users] No audio both ways. > Cc: users(a)lists.kamailio.org > Message-ID: > <9853f4ff0810090207p11d15bddt8dca8f57370c83c9(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Try adding a: > > nat=yes > > to the kamailio/openser peer definition and test > > dvg > > On Thu, Oct 9, 2008 at 10:58 AM, luzango mfupe <luzango.mfupe(a)gmail.com > >wrote: > > > > > Hi mates,I have this setup: > > Xlite---->Openser---->Asterisk------>VoIP to PTSN Provider > > > > I use kamailio 1.3.3(port5060), MediaProxy and Asterisk 1.4 (port 5065) > on > > the same Debian Box in Realtime with no NAT. Asterisk connects calls to > the > > VoIP to PSTN provider. I am able to establish calls towards the PSTN > > side(Landline & Mobiles) but with no audio. I can hear the ringing tone > but > > when the call connects and the conversation begin i hear nothing so as > the > > Callee side. > > > > Below are my configs,the ngrep captured packets and codecs. > > > > > #################################################################################### > > route[4] { > > # routing to the public network > > rewritehostport("xx.xxx.xxx.xx:5065"); > > t_on_failure("2"); > > if (!t_relay()) { > > sl_reply_error(); > > }; > > exit; > > } > > > > route[6] { > > # > > # -- NAT handling -- > > # > > if (isbflagset(6) || isbflagset(7)) { > > append_hf("P-hint: Route[6]: mediaproxy \r\n"); > > use_media_proxy(); > > }; > > } > > > > route[10] { > > #from an internal domain -> inbound > > #Native SIP destinations are handled using the location table > > #Gateway destinations are handled by regular expressions > > append_hf("P-hint: inbound->inbound \r\n"); > > > > if (uri=~"^sip:0[1-9][0-9]+@.*") { > > if (is_user_in("credentials","local")) { > > strip(1); > > prefix("27"); > > route(6); > > route(4); > > exit; > > } else { > > sl_send_reply("403", "No permissions for local calls"); > > exit; > > }; > > }; > > > > if (uri=~"^sip:00[1-9][0-9]+@.*") { > > if (is_user_in("credentials","int")) { > > strip(2); > > route(6); > > route(4); > > exit; > > } else { > > sl_send_reply("403", "No permissions for international > calls"); > > }; > > }; > > > > > > > ################################################################################### > > This call was from the xlite softphone 1974 towards the Landline > > 0123825710. > > > > > ################################################################################### > > U 2008/12/06 03:38:43.896057 196.212.209.18:46738 -> kamailio IP:5060 > > INVITE sip:0123825710@KAMAILIO ip SIP/2.0..Via: SIP/2.0/UDP > 192.168.0.55 > > : > > > > > 46738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..Max-Forward > > s: 70..Contact: <sip:1974@196.212.209.18:46738>..To: > > "0123825710"<sip:01238 > > 25710@kamailio IP>..From: <sip:1974@kamailioIP>;tag=353dd217..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Allow: > > INVIT > > E, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, > > INFO..Cont > > ent-Type: application/sdp..User-Agent: X-Lite release 1014k stamp > > 47051..Co > > ntent-Length: 237....v=0..o=- 1 2 IN IP4 192.168.0.55..s=CounterPath > > X-Lite > > 3.0..c=IN IP4 192.168.0.55..t=0 0..m=audio 60782 RTP/AVP 0 8 3 > > 101..a=fmtp > > :101 0-15..a=rtpmap:101 telephone-event/8000..a=alt:1 1 : ljiQYRpD > > NiMZsfdZ > > 192.168.0.55 60782..a=sendrecv.. > > > > U 2008/12/06 03:38:43.896350 Kamailio IP:5060 -> 196.212.209.18:46738 > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP > > 192.168.0.55:46 > > > > > 738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport=46738;received > > =196.212.209.18..To: "0123825710"<sip:0123825710@kamailioIP > >;tag=329cfea > > a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@kamailioIP > >;tag=353dd217 > > ..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 > > INVITE..Pr > > oxy-Authenticate: Digest realm="41.208.212.97", > > nonce="4939d8cfeb060ab14354 > > 85eee811cdf644f759a2"..Content-Length: 0.... > > > > U 2008/12/06 03:38:44.086313 196.212.209.18:46738 -> kamailio IP:5060 > > ACK sip:0123825710@kamailio IP SIP/2.0..Via: SIP/2.0/UDP > > 192.168.0.55:467 > > 38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To: > > "012382571 > > 0"<sip:0123825710@kamailioIP > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe. > > .From: <sip:1974@41.208.212.97 <sip%3A1974(a)41.208.212.97> < > sip%3A1974(a)41.208.212.97 <sip%253A1974(a)41.208.212.97> > >>;tag=353dd217..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3 > > ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 ACK..Content-Length: 0.... > > > > > > > > U 2008/12/06 03:38:44.582208 asterisk IP:5065 -> kamailio IP:5060 > > SIP/2.0 100 Trying..Via: SIP/2.0/UDP kamailio > IP;branch=z9hG4bKd78.7bd576 > > 24.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738 > > ;received=1 > > 96.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=4673 > > 8..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: > <si > > p:1974@kamailio IP>;tag=353dd217..To: > > "0123825710"<sip:0123825710@41.208. > > 212.97>..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 2 > > INV > > ITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, > BYE, > > RE > > FER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: > > <sip:27123825710@41.2 > > 08.212.97:5065>..Content-Length: 0.... > > > > U 2008/12/06 03:38:44.711225 70.42.72.49:5060 -> Asterisk IP:5065 > > SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk > IP:5065;branch=z9hG4b > > K22abec12;rport=5065..From: "1974" <sip:1974@AsteriskIP > :5065>;tag=as6a7c > > b89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(a)70.42.72.49> > >>..Call-ID: > > 1934f5d443abffe07 > > c59d0a42215b49c@41.208.212.97..CSeq: 102 INVITE..Server: OpenSER > > (1.3.2-not > > ls (i386/solaris))..Content-Length: 0.... > > > > U 2008/12/06 03:38:47.206445 70.42.72.49:5060 -> Asterisk IP:5065 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk > IP:5065;branch=z9 > > hG4bK22abec12;rport=5065..From: "1974" <sip:1974@AsteriskIP > :5065>;tag=as > > 6a7cb89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(a)70.42.72.49> > > > > >;tag=cba-1a6e-48ecbab6.. > > Call-ID: 1934f5d443abffe07c59d0a42215b49c@Asterisk IP..CSeq: 102 > > INVITE.. > > Contact: <sip:1214650027123825710@70.42.72.138<sip%3A1214650027123825710(a)70.42.72.138> > <sip%3A1214650027123825710(a)70.42.72.138<sip%253A1214650027123825710(a)70.42.72.138> > >>..Date: > > Wed, 08 Oct 2008 13: > > 50:49 GMT..Server: BRSIP v2.0.1.2..Record-Route: <sip:70.42.72.49 > > ;lr=on;fta > > g=as6a7cb89f>..Content-Type: application/sdp..Content-Length: > > 212....v=0..o > > =BRSDP 792898 792898 IN IP4 216.49.201.22..s=BRSDP Session..c=IN IP4 > > 216.49 > > .201.22..t=0 0..m=audio 27852 RTP/AVP 0 101..a=ptime:20..a=rtpmap:0 > > PCMU/80 > > 00..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. > > > > U 2008/12/06 03:38:47.206891 Asterisk IP:5065 -> Kamailio IP:5060 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP kamailio > IP;branch=z9hG4bK > > d78.7bd57624.0;received=kamailio IP..Via: SIP/2.0/UDP > 192.168.0.55:46738 > > ; > > received=196.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-; > > rport=46738..Record-Route: <sip:41.208.212.97 > > ;lr=on;ftag=353dd217;nat=yes>. > > .From: <sip:1974@kamilio IP>;tag=353dd217..To: > "0123825710"<sip:01238257 > > 10(a)41.208.212.97>;tag=as4377a96d..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVl > > NTQ0NTUwZjg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, > > ACK, > > CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: > > replaces..Conta > > ct: <sip:27123825710@Asterisk IP:5065>..Content-Type: > > application/sdp..Co > > ntent-Length: 287....v=0..o=root 2664 2664 IN IP4 > > 41.208.212.97..s=session. > > .c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 > > 101..a=rtpmap:8 > > PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 > > telepho > > ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - > > -..a=ptime:20..a=se > > ndrecv.. > > > > U 2008/12/06 03:38:47.207106 Kamailio IP:5060 -> 196.212.209.18:46738 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.55:46738 > > ;received= > > 196.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=467 > > 38..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: > <s > > ip:1974@Kamailio IP>;tag=353dd217..To: > > "0123825710"<sip:0123825710@kamilio IP>;tag=as4377a96d..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZ > > jg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, > > CANCEL, > > OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: > > <sip: > > 27123825710@41.208.212.97:5065>..Content-Type: > > application/sdp..Content-Len > > gth: 287....v=0..o=root 2664 2664 IN IP4 Asterisk IP..s=session..c=IN > IP4 > > 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8 > > PCMA/800 > > 0..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 > > telephone-event/ > > 8000..a=fmtp:101 0-16..a=silenceSupp:off - - - > > -..a=ptime:20..a=sendrecv.. > > > > > > > ######################################################################################## > > Sip.conf > > [general] > > context=from-trunk > > bindport=5065 > > autocreatepeer=yes > > bindaddr=xx.xxx.xxx.xx > > > > > > disallow=all > > ;allow=gsm > > ;allow=amr > > allow=alaw > > allow=ulaw > > allow=gsm > > ;allow=ilbc > > ;disallow=all > > ; > > ;useragent=Asterisk PBX > > dtmfmode = rfc2833 > > > > > > domain=xx.xxx.xxx.xx ; Add IP address as local domain > > > > [Provider] > > disallow=all > > canreinvite=no > > context=from-trunk > > allow=all > > ;allow=ulaw > > ;allow=gsm > > host=xx.xx.xx.xx > > insecure=port,invite > > type=peer ; we only want to call out, not > be > > call$ > > dtmfmode=rfc2833 > > > > > ######################################################################################### > > Here is my codecs > > > > 41*CLI> core show translation > > Translation times between formats (in milliseconds) for one > second > > of data > > Source Format (Rows) Destination Format (Columns) > > > > g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc > g726 > > g722 amr > > g723 - - - - - - - - - - - > - > > - - > > gsm - - 2 2 2 2 1 5 - 19 - > 2 > > - 14 > > ulaw - 5 - 1 2 2 1 5 - 19 - > 2 > > - 14 > > alaw - 5 1 - 2 2 1 5 - 19 - > 2 > > - 14 > > g726aal2 - 5 2 2 - 2 1 5 - 19 - > 1 > > - 14 > > adpcm - 5 2 2 2 - 1 5 - 19 - > 2 > > - 14 > > slin - 4 1 1 1 1 - 4 - 18 - > 1 > > - 13 > > lpc10 - 6 3 3 3 3 2 - - 20 - > 3 > > - 15 > > g729 - - - - - - - - - - - > - > > - - > > speex - 6 3 3 3 3 2 6 - - - > 3 > > - 15 > > ilbc - - - - - - - - - - - > - > > - - > > g726 - 5 2 2 1 2 1 5 - 19 - > - > > - 14 > > g722 - - - - - - - - - - - > - > > - - > > amr - 6 3 3 3 3 2 6 - 20 - > 3 > > - - > > > > > > > > With best regards, > > Lu. > > > > -- > > Luzango Mfupe > > TUUNE MOBILE > > Tel:0128440528/0123825710 > > Tshwane-RSA > > > > "...Ships are safe in harbor, but they were never meant to stay > > there......." > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > [View Less]

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[Serusers] SIP MESSAGE BODY ENCRYPTION
by Adeyeye Oluwasegun Michael 09 Oct '08

09 Oct '08

Hi, I was wondering if the content or body of a SIP MESSAGE from a SIP client is always encrypted when a TLS-supported SIP proxy for the client registration and the likes. I know that S-MIME supported SIP stack could do that for me, but I was wondering if a TLS-supported proxy could do as well. Thanks, Michael.

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[Kamailio-Users] SST module: Min-SE optional in request
by Schumann Sebastian 09 Oct '08

09 Oct '08

Dear all I might have noticed a bug in SST module: If I send an INVITE with Session-Expires: 300 Supported: timer I get rejected with 422 Session-Timer too small. If a Min-SE: x header with x > set min_se parameter included I do not get the message. This is IMHO wrong behavior, as Min-SE is optional in initial INVITE-requests sent by the phone acc. RFC 4028, table 1. Another thing that I noticed is the response to this message includes the Min-SE header (as it should), but has … [View More]

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[Kamailio-Users] !!Urgent Need to setup SIP SERVER like SIP Account Provider as my own Server.Thanks for your very great Help.
by sam 09 Oct '08

09 Oct '08

HI , I want to be a SIP providers like broadvoice gimzo ,where can i get DID numbers ,do we need to have contact with telecommunication Department. Please give me a full details.When we use *kamailio *,then there is no need to get sip account and virtual numbers. -- Cheers K.S.Sampath Kumar

3 2

[Kamailio-Users] Kamailio and Asterisk trunking?
by John D 09 Oct '08

09 Oct '08

Not sure if I'm properly wording this, but I used to have two asterisk servers speaking to each other over SIP, configured via sip.conf. I now need a way to forward calls from my Asterisk instance to my Kamailio server so it can properly distribute the call. At this point however, I cannot find any documentation in order to properly have the two speak to each other. Would anyone be kind enough to point me in the right direction to establish that connection? Thanks.

3 2

[Kamailio-Users] Attended Transfer with asterisk
by ALAEDDINE abbech 08 Oct '08

08 Oct '08

Hello, Does anybody got a working configuration to make an "attended call transfer" with a call through an Asterisk gateway? Example: PSTN --> Asterisk --> SER --+-- A |<---transfer call pstn to B +-- B The call will come from the PSTN Network and will go through "A". A sets the call on "Hold" and calls "B". After A is connected with B, A hangup and B got the call from PSTN. This is not working at the moment. Attended … [View More]

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[Kamailio-Users] ACK without matching transaction discarded
by Alex R.S.M 08 Oct '08

08 Oct '08

Hi, I am using openSER as a proxy for a simple SIP call (UAC <--> openSER <--> UAS). openSER receives ACK from callee but doesn't send it to caller. It seems openSER can not find a matching transaction, so it ignore and discard the message. if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... … [View More]

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[Kamailio-Users] Ping both UACs when a call starts
by Cesar Troncoso 08 Oct '08

08 Oct '08

Hello! When a call starts I would like OpenSER to ping (by sending OPTIONS or INFO request) both UAC´s just to know if they are still there or for some reason they crashed or lost connectivity. In case they don't response, the call should be finished. What is the best way to do this? Thanks in advance! Cesar

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[Kamailio-Users] openser 1.3.3 domain module
by toly 07 Oct '08

07 Oct '08

Greetings, In domain module, command is_domain_local - using parameter_fixup where there is a call pkg_malloc, but nowhere function free_fixup is used. This will be memory leak, or I'm missing something. Thanks for the prompt reply, Toly -- View this message in context: http://www.nabble.com/openser-1.3.3-domain-module-tp19863638p19863638.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

[Kamailio-Users] LCR to Carrierroute with DNS/SRV Support
by Krom 07 Oct '08

07 Oct '08

Hello, I am/will be a newbie to 1.3.x and beyond and have a question regarding a new requirement of two of my upstream providers. We have been running 1.1.1-notls since it was released. We have been in the mindset of, "if it's not broken, don't fix it". But our upstream providers are now telling us to move to DNS/SRV for signaling outbound - for good reason. We have been using LCR for carrier/gateway selection and for URI manipulation prior to relaying to a cluster of session boarder … [View More]

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sr-users October 2008