sr-users May 2018

sr-users@lists.kamailio.org

92 participants
139 discussions

by Alex Balashov

Recent repeated questions on this topic have inspired a blog post: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… This is a rough cut, and may be edited or expanded later. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

6 years, 6 months

Route header in local reply

by Евгений Голей

Hi. Colleagues write how to configure the local responses to requests so that the response added the header of the Route in accordance with the record-route headers that were in the request ? Example SBC --> Kamailio REGISTER sip:domain.com SIP/2.0 Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com> From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Max-Forwards: 70 Content-Length: 0 User-Agent: SBC Expires: 0 Kamailio --> SBC SIP/2.0 200 OK Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com>;tag=550bc1454fc5070bdadb084c686d99a2.ac18 From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Server: kamailio (5.1.0 (x86_64/linux)) Content-Length: 0 -- Evgeniy

6 years, 6 months

How long time does a push take?

by Kjeld Flarup

Hello I have a question to those of You who uses push via Google or Apple to initiate calls. How long time does it usually take, before the phone gets the push and wakes up? Our experience is, that it can take from 2 to 10 seconds, and that not impressing when we are talking about telephony. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

6 years, 6 months

Pleasing all kinds of UACs for RTP/SRTP with rtpengine

by Sergiu Pojoga

In continuation of previous <https://marc.info/?l=sr-users&m=150730146302404&w=2>thread about none/optional/compulsory SRTP handling... I used the suggested approach of *t_reuse_branch()* in a *tm:branch-failure* detected by t_check_status("(415)|(488)") replies, with the only difference of me doing it vice-versa from the suggested mantra of offer SRTP first then fallback to RTP. It works... well, for some UACs, not so with others. The 'good' UACs happily accept an updated branch, like for e.g. Yealink, will reply with "*488 Not Acceptable Here*" at first followed by a *200 OK*. The 'bad' UACs, like for e.g. Bria || Zoiper, will issue a "*415 Unsupported Media Type*" at first, and then a "*482 Merged Request*" after receiving and updated branch with a different RTP profile. Rightfully so, they decided that the updated branch is a retransmission, since both branches have the same CSeq, Call-ID and From tag. The branch ID however differs, so does the RTP profile obviously. *Any suggestion on how to convince all UACs that the 2nd attempt is different than the 1st one?* According to RFC 3261, a request is a merged request if ... "8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction."

6 years, 6 months

TOPOS (5.1.[23]) breaks PRACK

by Daniel Tryba

Enabling topos breaks PRACK in my simple (near default) proxy setup (attached kamailio.conf). asterisk 13 (10.0.3.147) <-> kamailio (10.0.3.87) <-> asterisk 13 (10.0.3.154) With the event route topos:msg-outgoing dropping all. The flow of messages is as expected. INVITE -> <- Ringing PRACK -> <- 200 OK With topos enabled, the PRACK doesn't get past route[WITHINGDLG] and is answered with the sl_send_reply("404","Not here"); INVITE -> <- Ringing PRACK -> <- 404 Not here These specific tests were done on kamailio 5.1.2, but with 5.1.3 I see the exact same messages.

6 years, 6 months

Need help with NAT Traversal - Changing INVITE OK contact to IP port to Contact in Location

by KamDev Essa

The stock cfg states definng WITH_NAT and alls good. I did exactly that. However boing with stock cfg and WITH_NAT, the Registration info in location table is perfect with the UAs external IP address and port. This is not the case with the OK that in response to an INVITE. In the Contact of the OK the private IP shows up and Kamailio just sends it off to the Carrier who obviously cannot send a ack to a private IP. Is there a step by step guide for NAT Traversal? H

6 years, 6 months

log CDRs to text file

by eyas barhouk

Hello Dears I'm trying to get Kamailio to log its CDRs (collected By ACC module ) to text file instead of a database so i used modparam("acc", "db_url", "flatstore:/var/log/acc") and in each time kamailio trying to write on the text file i get the error : " db_flatstore flat_new_connection() : not a directory" so could any one guide me about how i have to create the database text file "acc"? and what is the cause of the error i got ?

6 years, 6 months

Carrier IP on softphone call history !!!

by KamDev Essa

I have a simple call flow for Inbound call going to an extendion carrier call ===> WITH_ALIASDB ===> extension (on Xlite) Works perfectly getting the call. Issue : - On Xlite I see the call as <TN>@<carrierIP>. I dont want my customers to know what carrier I use and also I want to be in the loop for CDR. - Hanging on either phone will not send a BYE to my server. How do I get Kamailio to be in the SIP ladder for this call? KD

6 years, 6 months

Private IP in Contact Header of 200 OK back to Carrier, so no ACK - Call drop

by KamDev Essa

Scenario Carrier ===> Kamailio ===> Ext on domainINVITE ===> ====> 200 OK <== <=== Repeated because carrier pcap shows private 192.168 IP in Contact header. cfg has WITH_NAT setup. I assumed that all I need to do. What more is required to get Kamilio IP and port in 200K to Carrier? K

6 years, 6 months

Where do I get rpc documentation for Kamailio

by KamDev Essa

I would like to perform subscriber and domain adds and reloads programmatically. Where can I find info on kamailio APIs I could use? KD

6 years, 6 months

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sr-users May 2018