sr-users September 2013

sr-users@lists.kamailio.org

89 participants
127 discussions

Re: [SR-Users] Getting Started with SIREMIS

by David Wilson

Hi Daniel, That's the path I'm on, just wanted to make sure I hadn't overlooked something obvious. In a test environment, I have a SIP client registered to kamailio. I also have kamailio registered with an ITSP using uac module, based on a UACReg record configured through Siremis. Inbound calls are routing nicely, but I'm having trouble with outbound calls. My ITSP account is challenging the outbound INVITE. Configuring peering is an option, but I'm trying to understand the authentication case for my own satisfaction. If I understand correctly, there are a number of approaches for outbound: 1. Use another service as a gateway (asterisk, etc) 2. Use lcr module 3. Use carrierroute module 4. Use manual routing in the config file Some comments on these: 1. I understand that kamailio is not designed to be a B2BUA, I'm trying to use it as a proxy. I prefer not to introduce another element. 2. lcr module looks promising. I got as far as loading a gateway from the database, but then hit the authentication problem described in 4. below. 3. carrierroute module has been troublesome. I'm using the APT package and found that carrierroute was not included. After installing kamailio-carrierroute-modules, kamailio fails to start with a load_module error: undefined symbol cfg_set_error_function. Similar trouble has been reported in relation to libconfuse. This is installed and 'ldd carrierroute.so' does not report any missing dependencies. 4. Since I'm using only a single outbound route (for now), I'm working on manual routing in the config file while I get the authentication working. I have got as far as catching the 401 Unauth response from the ITSP to my INVITE, calling uac_auth() in the failure_route (see fragment below) and relaying. The ITSP then returns 482 Merged Request in response to the INVITE that includes the auth header. This occurs regardless of whether I call append_branch before relaying. I have read a number of historical threads about the CSeq not being incremented and it makes sense to me in this context not to do so, but a different branch tag doesn't appear to be enough in this case for the server to treat it as a different request. failure_route[1] { if (t_check_status("401")) { append_branch(); uac_auth(); } t_relay(); } Incidentally, in order to create the Auth response, I had to hard code credentials into the config file with modparam("uac","credential","user:domain:password") Can these credentials be configured from Siremis? I thought it might use the existing UACReg record, but that doesn't seem to be the case. I'm happy to break out any of these into individual threads, but I've put them all together here to give you an idea of where I'm at. Cheers, Dave. On 20 Sep 2013, at 4:37 am, sr-users-request(a)lists.sip-router.org wrote: > > Hello, > > > On 9/19/13 9:59 AM, David Wilson wrote: >> Hi All, >> >> I have Kamailio 4.0.3 installed via APT onto Ubuntu 12.04 and have Siremis 4.0 installed and running nicely. >> >> Is there an example config file that demonstrates how to use the features configurable through Siremis? >> >> For example, when I populate a UACReg record via Siremis, I had to adapt the kamailio.cfg to use that information. Similar adaptations are required for LCR or CR records and these are where I'm struggling; there is some routing code that is implied by the data structures, but the documentation for the corresponding modules doesn't provide an overall example. > siremis is the toolkit to manage the database records for most of the > records. Building the logic to use the records from kamailio.cfg is a > matter of requirements to implement. > > The starting point is the readme for the module, there you can find > details about parameters and function. The way to go is trying something > by yourself based on readme and if you get stuck, then ask help here. > Otherwise no one would be able to give hints or config parts to plug in > an unknown environment. > > Cheers, > Daniel

10 years, 7 months

Getting Started with SIREMIS

by David Wilson

Hi All, I have Kamailio 4.0.3 installed via APT onto Ubuntu 12.04 and have Siremis 4.0 installed and running nicely. Is there an example config file that demonstrates how to use the features configurable through Siremis? For example, when I populate a UACReg record via Siremis, I had to adapt the kamailio.cfg to use that information. Similar adaptations are required for LCR or CR records and these are where I'm struggling; there is some routing code that is implied by the data structures, but the documentation for the corresponding modules doesn't provide an overall example. Regards, Dave.

10 years, 7 months

Modifying path to kamailio.cfg

by Grant Bagdasarian

Hello, I compiled and installed Kamailio 4.0.3 like usual on a new test machine, but changed the CFGFILE variable in /etc/init.d/kamailio to /usr/local/etc/kamailio/kamailio.cfg. Kamailio started normally. When I reset CFGFILE to CFGFILE=/etc/kamailio/kamailio.cfg and start Kamailio again, I'm getting an error saying it can't find the file at /usr/local/etc/kamailio/kamailio.cfg. I renamed the config file at /usr/local/etc/kamailio/ to make sure it's not using that one, but it still is. Why is the change not being accepted? Regards, Grant

10 years, 7 months

siremis 4.0 problem

by Marek Cervenka

hi, i'm trying siremis 4.0 with kamailio 4.0.3. installation with web wizard was successfull i have strange problem with login. login succed but then i'm immediately logged out. i see MSG_LOGIN_SUCCESSFUL in log/DEBUG.log can you point me where i can enable additional DEBUG in siremis? or any other ideas what's the problem? thanks centos6, mysql 5.1, php 5.3 -- --------------------------------------- Marek Cervenka =======================================

10 years, 7 months

SCA Module Drops Shared Line if Barge-In Aborted on Polycom

by Robert Boisvert

SCA Developers, I ran into a problem using the SCA module and was looking for some assistance for the following situation. Can you help? Here is the scenario: 1. When a conversation is occurring on a shared line with Polycom phones; 2. Another caller on the same shared line picks up the phone and uses the barge-in feature; 3. The caller aborts the barge-in before being connected; 4. SCA or Polycom, I don't know which at this point, turns off the LEDs on the shared line and then the shared line no longer behaves like a shared line. 5. Even though the existing shared call continues, if the new caller picks up the shared line he gets a green LED, hears a dial tone and is not prompted to barge-in by the soft keys. 6. When all parties end the conversation the shared line returns to its previous correct behavior. Have you seen something like this? Do you have any suggestions on how to troubleshoot or workaround the problem? Thanks again for your help. The SCA module remains a very useful product for us, Bob

10 years, 7 months

Store dispatcher weight attribute in dispatcher table

by Grant Bagdasarian

Hello, Is it possible to store the weight attribute in the dispatcher table? If so, which column must I use? I see it is possible when using the list file. The attrs_avp is the closest thing, but I think this value has to be loaded from the database also? Regards, Grant

10 years, 7 months

Authentication and outbound proxy

by Isaac McDonald

Hello, Is there any way to authenticate requests when using Kamailio as an outbound proxy? For example, if all the phones are configured to use Kamailio as an outbound proxy for NAT traversal, the credentials on the phone authenticate against the destination SIP proxy and not the outbound SIP proxy (Kamailio). Is there a way to have the credentials on the phone authenticate to BOTH the outbound proxy and the destination SIP proxy? Thanks, Isaac

10 years, 7 months

Regd. Presence_xml module

by Shankar

Hi, While reading the module documentation of 'PRESENCE_XML' module, I came across 'intergrated_xcap_server'. If it is set to value '0', then it has to use xcap_client module to fetch documents from XCAP server. But as per documentation, it says this mode is not supported. So can I assume that only intergrated xcap server is supported in kamailio and not standalone. If YES, module xcap_client cannot be used at all? Can someone please confirm the same. Regards, Shankar

10 years, 7 months

Re: [SR-Users] (no subject)

by Keith

Hi, I am trying to use SIPP to load test Kamailio. I am using a combination with Freeswitch and Kamailio as an SBC with the calls being delivered to Asterisk. The problem seems to be with the ACK sent back from SIPP on response to the 200 OK. This ACK is not being forwarded by Kamailio and therefore FS drops the call. I have scoured every page and tried everything so any help would be appreciated. Thanks, Keith

10 years, 7 months

TEXTOPS parser problem with binary data in MIME body

by Klaus Feichtinger

Hello, I have a problem with Kamailio version 3.2.4 (tested also with 3.3.5) regarding "binary data" in message bodies. According RFC3261 it is explicitly allowed using binary data within MIME bodies: RFC 3261 section 7.4.1: SIP messages MAY contain binary bodies or body parts. When no explicit charset parameter is provided by the sender, media subtypes of the "text" type are defined to have a default charset value of "UTF-8". However, the Kamailio function "Regular Expression Transformation" (re.subst), which is based on the transformation class (exported by the textops module) is causing problems in our customer system. In regular scenarios, Kamailio is receiving SIP INVITE messages with a MIME body, which is containing following parts: - application/sdp (standard conform) - application/x-q931 (Cisco proprietary with BINARY data!) - application/gtd (Cisco proprietary with ASCII strings) The content of the "x-q931" and "gtd" body parts is depending on (UUS1) data that were received on the ISDN line. Kamailio has to forward User-User specific information from the ISDN line to SIP user agents (in form of the "User-to-User" header field). The content of these UUS1 data may contain also byte values "00", which is legitimate. In general, Kamailio is in every INVITE message searching specific content in the last body (application/gtd) and copying this content to a config variable. As soon as the x-q931 body contains nul values (0x00 in binary format), the parser stops at this position and cannot parse the rest of the message. Therefore, I am missing the information that should be copied to the SIP header field, as the parser stops before the end of the message body! As long as the message body does not contain 0x00, it is working fine! My question is: - is this a bug in Kamailio parsing functions? - is this a design issue of Kamailio text parsers (as binary data are allowed acc. RFC3261) - does anybody know a solution for this problem? This "bug" is causing big troubles.... Thanks in advance, Klaus Feichtinger P.S. trace data (1)...(3) of my problem (1) exemplary content of the x-q931 body ======================Sending: Binary Message Body Sep 13 10:50:19: Content-Type: application/x-q931 08 01 B4 05 04 03 80 90 A2 18 01 89 1E 02 82 88 6C 05 B1 35 30 30 34 70 05 B1 35 30 30 31 7D 02 91 81 7E 18 04 1D 42 75 25 92 43 31 62 94 00 00 2C 68 20 64 00 62 F2 10 41 B9 D7 BD 0D 0A (2) SIP INVITE message (debugger view) =============================INVITE sip:115101@<ipv4>:5060 SIP/2.0 Via: SIP/2.0/UDP <ipv4>:5060;branch=z9hG4bK171164E From: <sip:1101015004@<ipv4>>;tag=29E16410-2541 To: <sip:115101@<ipv4>> Call-ID: 1DBCDACB-1B9911E3-8992FF70-D2BED293@<ipv4> Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 90 CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:1101015004@<ipv4>:5060> Expires: 1800 P-Asserted-Identity: <sip:1101015004@<ipv4>> Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 820 --uniqueBoundary Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=CiscoSystemsSIP-GW-UserAgent 1770 5294 IN IP4 <ipv4> s=SIP Call c=IN IP4 <ipv4> t=0 0 m=audio 16384 RTP/AVP 8 0 c=IN IP4 <ipv4> a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 --uniqueBoundary Content-Type: application/x-q931 Content-Disposition: signal;handling=optional Content-Length: 62 4" l15004p15001}~Bu%C1b..,h d.brA9W --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,08,,1,5001 CGN,08,,1,,,5004 UUS,3,1d427525924331629400002c6820640062f21041b9d7bd CPC,09 FCI,,,,,,,y, GCI,185ef1291b9911e381d500270dff3fa0 --uniqueBoundary-- (3) config excerpt ============if (has_body("multipart/mixed")) { if (filter_body("application/sdp")) { remove_hf("Mime-Version"); remove_hf("Content-Type"); append_hf("Content-Type: application/sdp\r\n"); } else { xlog("L_WARN", " <route> message body part 'application/sdp' not found\n"); } $var(UUS)=$(rb{re.subst,/^(.*)UUS,.,([a-z0-9,]*)..[A-Z][A-Z][A-Z],(.*)/\2/s}); append_hf("User-to-User: $var(UUS)\r\n", "CSeq"); if !(subst_body('/Content-Disposition..session.handling=required....//s')) { xlog("L_WARN", " <route> substituting Content-Disposition FAILED!!! \n"); } msg_apply_changes(); if (search_body("a=[a-z]+:.+[\r\n]{4}$")) { #!ifdef WITH_XLOGDEBUG xlog("L_INFO", " <route> double CRLF at the end of the message body detected - is reduced to one now. \n"); #!endif $var(sdp) = $(rb{s.striptail,2}); set_body("$var(sdp)", "application/sdp"); } }

10 years, 7 months

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sr-users September 2013