Hello,
last edition of Kamailio Development Workshop happened about three years
ago. As I got requests about it from time to time, I thought of trying
to organize a new edition. Rather short term by now due to various
constraints, so the plan is to do it again in Alicante, Spain, during
February 15-16, 2016.
I put together more details at:
* http://www.kamailio.org/w/development-workshop/
Besides spending 2 days digging into Kamailio C code, such event is also
good for discovering and sharing development resources among
participants, as well as meeting people from community -- at the
previous editions I met for first time with Victor Seva, Vicente
Hernando and Seudin Kasumovic, which in turn contributed a lot of code
afterwards. It will be also testing phase for upcoming major release,
maybe we can plan some stress testing session on site.
Right now I am looking to see if there is any interest in such event in
order to nail down organizing it or not. Should anyone consider to
participate, write me.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
Hi all,
We are investigating an issue with BLF whereby the BLF key on the subscribed phone does not always "turn off" after the call has ended. We have enabled force_single_dialog to keep things simple for debugging. The attached pcap is the result. The relevant stuff as I see it is:
* Packet #60 - UA sends a BYE
* Packet #62 - PUA sends a PUBLISH over loopback with a state of "terminated"
* Packet #65 - The PUBLISH is sent out to the subscribed UA as a NOTIFY with a state of "early"
This causes the BLF light on the phone to continue flashing.
We are running a version of Kamailio compiled from the HEAD of the current 4.3 branch.
Any ideas?
Phil Lavin
Telecoms Systems Manager
CloudCall by SYNETY
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I am stumped as my DID provider can't route via SIP URI to PBXes but they can to IPTel. I can call my own PBX, and from my own PBX to the PBXes URI without issue. When I route the DID via URI to IPTel and set voicebox it works and routes to the voicebox. When I add a forward for the PBXes URI it then breaks and gets a denial of 404 not in service.
Ideas anyone? I am scratching my head here. Appreciate any help!
Regards,
Alex McConnachie
Hi,
We're running a system with Kamailio running in front of Asterisk just
handling registrations and forwarding everything else to Asterisk. But
we're having an issue during hangup on incoming calls. If the initiator
hangs up, the call completes successfully. But if one of our phones hangs
up, the BYE message comes back with a 404 "Not Found" and the call doesn't
hang up on the carrier side.
According to the carrier, it's because the IP in the contact on our ACK
message goes to their audio IP while the header of our BYE points to their
signaling IP.
ACK sip:[Kamailio Pub
IP]:5060;line=sr--rkpsDAp6YAp6DIpZDZmZeI2ZYI26YIRVDcpsDIpsem* SIP/2.0
Via: SIP/2.0/UDP *[Carrier Signaling IP]*;branch=z9hG4bK2236.1402e7b4.2
Via: SIP/2.0/UDP *[Carrier Audio IP]*;received=*[Carrier Audio IP]*
;branch=z9hG4bK07a8bccb;rport=5060
Route: <sip:[Kamailio Pub
IP];r2=on;lr=on;ftag=as67cef00d;nat=yes>,<sip:10.120.0.1;line=sr--rkpsDthVDIhVDNh6Ogo6eKh6eAQs4LRflC2srQRflC2srGqAl-CAP6rZrZkGDmpGed2APtCvlx1>
From: "+16014477389" <sip:6014477389@*[Carrier Audio IP]*>;tag=as67cef00d
To: <sip:6016025063@*[Carrier Signaling IP]*>;tag=as643b40ca
Contact: <sip:6014477389@*[Carrier Audio IP]*>
Call-ID: 4aaefec90826a2a221f0af9500ad211b@*[Carrier Audio IP]*
CSeq: 102 ACK
User-Agent: packetrino
Max-Forwards: 69
Content-Length: 0
BYE sip:6014477389@*[Carrier Signaling IP] *SIP/2.0
Via: SIP/2.0/UDP [Kamailio Pub
IP];branch=z9hG4bK2236.fca983a45913fb510f97e781a85c7392.0
Via: SIP/2.0/UDP
10.120.0.1;branch=z9hG4bKsr-IqktV1L26BCx0jmwZeI2ZYI26YIRVDcpsDIpsem.-EF8-EtCZYmg6edIMehhA4A.AEzyuiZKfPKo7N-qAcWq6D-rsYc4Zp**
Route: <sip:*[Carrier Signaling IP]*;lr=on>
Max-Forwards: 69
From: <sip:6016025063@[Kamailio Pub IP]>;tag=as643b40ca
To: "+16014477389" <sip:6014477389@*[Carrier Audio IP]*>;tag=as67cef00d
Call-ID: 4aaefec90826a2a221f0af9500ad211b@*[Carrier Audio IP]*
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
I'm thinking it's happening because their side isn't configured correctly
to handle traffic coming back from a proxy, but in the meantime is there a
way to rewrite the top of the BYE header to match the "audio IP" they're
requesting it be sent to?
Thanks!
--
Ryan Mottley, Developer
VOXO, LLC
voxo.co - (601)602-5063
Hi there,
I'm trying to install this rpm for my RHEL 5 box, but it asks for
mysql-libs; but RHEL 5 isn't shipped with mysql-lib,
only mysql55-mysql-libs, which kamailio-mysql doesn't recognize as a valid
dependency.
Any workaround here? Thanks!
--
Zuxy
Beauty is truth,
While truth is beauty.
PGP KeyID: E8555ED6
Hi, all
We're trying to build a system that consists of pbx, kamailio and asterisk
in the following configuration.
pbx (sip trunk) --- kamailio --- asterisk
The kamailio and asterisk are integrated with same database. The outgoing
calls to pbx works. But there is a problem with incoming calls from pbx.
If we make a consecutive calls from the same pbx user to the same user
registered with kamailio. The first would go through, but the second call
would be rejected by asterisk. We have insecure=invite set on the
trunk/peer, so asterisk is not supposed to auth the invite from kamailio.
But the pbx user (from in this case) is not in the database.
The asterisk log says:
[Jan 21 23:13:19] VERBOSE[20785] chan_sip.c: --- (16 headers 13 lines) ---
[Jan 21 23:13:19] VERBOSE[20785] chan_sip.c: Sending to 10.0.1.30:5061 (no
NAT)
[Jan 21 23:13:19] VERBOSE[20785][C-00000001] chan_sip.c: Sending to
10.0.1.30:5061 (no NAT)
[Jan 21 23:13:19] VERBOSE[20785][C-00000001] chan_sip.c: Using INVITE
request as basis request - 4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061
[Jan 21 23:13:20] NOTICE[20785][C-00000001] acl.c: SIP Peer ACL: Rejecting
'10.0.1.30' due to a failure to pass ACL '(BASELINE)'
[Jan 21 23:13:20] NOTICE[20785][C-00000001] chan_sip.c: Failed to
authenticate device <sip:95678@10.0.1.35>;tag=as4028dabf
[Jan 21 23:13:20] VERBOSE[20785][C-00000001] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 10.0.1.30:5061 --->
SIP/2.0 403 Forbidden^M
Via: SIP/2.0/TLS 10.0.1.30:5061
;branch=z9hG4bK9c8e.5cd2c05f6a572312c7793abf5fe1183c.0;i=2;received=10.0.1.30^M
Via: SIP/2.0/TLS 10.0.1.35:5061
;received=10.0.1.35;branch=z9hG4bK249855c1;rport=59929^M
From: <sip:95678@10.0.1.35>;tag=as4028dabf^M
To: <sip:16317@10.0.1.30>;tag=as35f47241^M
Call-ID: 4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 13.6.0^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE^M
Supported: replaces, timer^M
Content-Length: 0^M
Comparing the two invites from kamailio to asterisk, it seems the only
difference is that the second invite has an "i=2" in the Via header while
the first one has "i=1". Not sure what the "i=1" is for. Would appreciate
some insights on how kamailio is adding/handling the "i=#" in Via header.
Thanks.
Ding Ma
SPG, Motorola Solutions
Hello,
Hi team I have a kamailio SIP server for call Routing,my call flow is like
VoipDialer--->Kamailio-Routing Server-->Softswitch-->Telco/PSTN Mobile
caller----------------------------------------------------------------------->calle
Now Am facing an issue for a call from voip to telco after answering the
call
i tried to disconnect.but call is not disconneting..what will i do in
kamailio.cfg
pls help me to re-write the script am using tls also
Regards
AmanIdo
