- sr-users - kamailio.org

about tls client certificates
by Juha Heinanen 11 Apr '14

11 Apr '14

while doing some tls tests, i noticed that if tls.cfg has a section like this [server:default] verify_certificate = no require_certificate = no tls_method = SSLv23 private_key = /etc/sip-proxy/certs/sip-proxy/key.pem certificate = /etc/sip-proxy/certs/sip-proxy/cert.pem ca_list = /etc/ssl/certs/cacert.org.pem then client does not give its certificate to kamailio server during tls connection setup even if it had one. if i specify: require_certificate = no then client sends its certificate to kamailio server, but if another client does not have a client certificate, then it cannot connect at all. one way to solve this would be making kamailio listen on two tls ports, one for clients that are required to present a a certificate and another port for clients that do not have a certificate. unfortunately, it is not possible to add a mask to ip address in tls.cfg section like this: [server:0.0.0.0/0:5062] does anyone have a solution to this problem (other that running two kamailio instances)? -- juha

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SDP c= IP doesn't get rewritten
by Daniel Ciprus 11 Apr '14

11 Apr '14

All, I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration: #!subst "/RTP_INT_ADDR/10.96.173.15/" #!subst "/RTP_EXT_ADDR/168.7.24.17/" #!subst "/BOT_RTP_ADDR/10.96.173.52/" subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g'); subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g'); and this is what I ended up with: c=IN IP4 10.96.173.15168.7.24.17 Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if. I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ? Thanks Dan Server:: kamailio (4.2.0-dev0 (x86_64/linux)) Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7 Flags:: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT:: unknown -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

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Disjoint REGISTER and INVITE username/uri sets
by James Cloos 10 Apr '14

10 Apr '14

Given a proxy where the only registrations will be next-hop pbxes, where those names will not be INVITEable, what is the ideal way to tell kama the list of accepted usernames and authentication info for REGISTER? It is important to ensure that any INVITE to any of the REGISTERable usernames 404s. Thanks, -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 1024D/ED7DAEA6

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carrierroute prefix regex?
by Igor Potjevlesch 10 Apr '14

10 Apr '14

Hello, I'm wondering if we can set into < scan_prefix > an entrier with regex like : 3[0-9]{3}? If not, what would be the best module to manage in a DB the prefix that can be composed by SIP clients before call cr_route? Something equivalent to a dialplan I guess. Thanks for your feedback. Regards, Igor.

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Re: [SR-Users] How Do you Stack up Against Cisco's Call Manager
by Marjorie Lawrence 10 Apr '14

10 Apr '14

Hi, Not sure if this is a fair question. But how do you compare your call processing engine against Cisco's Call Manager? Appreciate your time. Marjorie Lawrence Global Wireless Networks, Inc. Senior Voice Engineer/Unified Communications Email: <mailto:Marjorie@gwnwireless.com> Marjorie(a)gwnwireless.com Website: <http://www.gwnwireless.com/> http://www.gwnwireless.com Office (301)674-5933

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how to track down issue with shared lines
by mark li 10 Apr '14

10 Apr '14

I have a scenario where calls that are transferred to a shared line doesn't trigger the "in use" light to show up on both phones that have the shared line. but when i make an outgoing call, let's say, using the shared line, the "in use" light lights up on both phones that have the shared line. Phone - polycom 650s. Just a noob with SIP, kamailio and telecom, so not sure where to start debugging. Any suggestions would be appreciated. thanks in advance for your time.

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Load fr_inv_timer from DB
by Igor Potjevlesch 10 Apr '14

10 Apr '14

Hello, I try to set the "fr_inv_timer" dynamically from DB. Under 60s it works fine. But I can't set more than 60s. Here are the TM parameters: # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 5000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 60000) Here are the instructions to set dynamically: avp_db_load("$ruri/username", "$avp(s:inv_timeout)/usr_timeout"); t_set_fr($avp(s:inv_timeout)*1000); Is there anything wrong? Regards, Igor.

1 0

topoh and onhold reinvites
by Daniel Tryba 10 Apr '14

10 Apr '14

I'm having some trouble with kamailio that uses the topology hider module and an incoming call being put onhold by the caller. The INVITE containing the sdp a=sendonly is to the 172.19.162.1 (the topoh mask_ip) contact header from the 200 OK to the initial INVITE. U 109.235.36.178:5060 -> 109.235.32.40:5060 INVITE sip:172.19.162.1:5060;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone SIP/2.0. This is handled within dialog (has_totag() && !loose_route()), which in the default config ends in a sl_send_reply("404","Not here") Explicitly handing INVITE (just relaying), results in the INVITE being sent to 172.19.162.1 (which doesn't exist) U 109.235.32.40:5060 -> 172.19.162.1:5060 INVITE sip:172.19.162.1:5060;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone SIP/2.0. How to get this INVITE to the actual destination? To get the same result for the INVITE as for the ACK to the 200 OK? U 109.235.32.40:5060 -> 109.235.36.178:5060 SIP/2.0 200 OK. ... Contact: <sip:172.19.162.1;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*>. U 109.235.36.178:5060 -> 109.235.32.40:5060 ACK sip:172.19.162.1;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy* SIP/2.0. U 109.235.32.40:5060 -> 109.235.34.107:5060 ACK sip:+31756418030@109.235.34.107:5060;transport=udp SIP/2.0. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

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ldap SSL
by Slava Bendersky 10 Apr '14

10 Apr '14

Hello Everyone, How I can specify ldap CA cert in ldap configuration ? The directive ca_list bellow is ignored. [07/Apr/2014:13:40:37 -0400] conn=20 op=-1 fd=70 closed - Peer does not recognize and trust the CA that issued your certificate. [07/Apr/2014:13:40:37 -0400] conn=22 op=-1 fd=64 closed - Encountered end of file. Current config ldap_server_url = "ldaps://ds389.network.com" ldap_bind_dn = "uid=adm,ou=People,dc=network,dc=com" ldap_bind_password = "password" authtype = simple tls = on ca_list = /etc/kamailio/cert/ca-cert_ldap.pem

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$sel(tls.peer.subject.cn) error
by Juha Heinanen 10 Apr '14

10 Apr '14

i have these two lines in config: $var(common_name) = @tls.peer.subject.cn; $var(common_name) = $sel(tls.peer.subject.cn); the latter gives error at startup: 0(24214) ERROR: <core> [select.c:316]: resolve_select(): Unable to resolve select 'tls' at level 0 0(24214) ERROR: <core> [select.c:177]: w_parse_select(): parse_select: error while resolve_select 'tls.peer.subject.cn' 0(24214) ERROR: pv [pv_select.c:45]: pv_parse_select_name(): invalid select name [tls.peer.subject.cn] 0(24214) ERROR: <core> [pvapi.c:839]: pv_parse_spec2(): pvar "sel" has an invalid name param [tls.peer.subject.cn] 0(24214) ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char [)/41] in [$sel(tls.peer.subject.cn)] at [24 (5)] 0(24214) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 874, column 29-53: Can't get from cache: $sel(tls.peer.subject.cn) ERROR: bad config file (1 errors) based on pseudo variable wiki page, the syntax should be correct. what is wrong with $sel(tls.peer.subject.cn)? -- juha

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Re: [SR-Users] LCR
by Keith 10 Apr '14

10 Apr '14

Hi, I am having issues with trying to use multiple lcr_ids so I can route to different groups. When I reload lcr it always only shows one group. I have copied the first group exactly and just changed the number but can't work out why it's not working. Any ideas? Thanks Keith

2 1

Exported params of USRLOC module
by aawaise 10 Apr '14

10 Apr '14

Hello, I am using kamailio-3.1 with siremis. I want to ask some questions about the exported params of usrloc module defined in the link given below. http://kamailio.org/docs/modules/3.1.x/modules_k/usrloc.html#id2997066 1. These exported params export value to whom ?? Other params or siremis location table ?? 2. If any export param is not defined in kamailio.cfg. Does its default value considered automatically or it is not used at all ?? 3. In siremis location table, I am not able to show my domain name of selected online user when I click on user. I want to do it. User's contact, received and other columns are filled but not domain name :( Shall I dig deep into usrloc module or I need to follow some other path to resolve this issue ?? Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Exported-params-of-USRLOC-module-tp… Sent from the Users mailing list archive at Nabble.com.

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Show Domain-Name in Siremis
by aawaise 10 Apr '14

10 Apr '14

Hello, I am running kamailio server in my set up. And have got siremis working for monitoring purposes. Problem is that in web interface, domain name of users is not shown in siremis. If a user is online, I get following credentials infront of "Client" column, Client.Name@Client's.IP How can I show domain name in siremis ?? Any help will be highly appreciated. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp12660… Sent from the Users mailing list archive at Nabble.com.

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Segfault when trying to send MESSAGE through websocket with jsSIP
by Alex Villacís Lasso 09 Apr '14

09 Apr '14

I am currently trying to replace a pure-Asterisk implementation of SIP messaging through Websockets, with a Kamailio-4.1.2-based implementation. However, when I try to send a message with jsSIP, Kamailio crashes: Program terminated with signal 11, Segmentation fault. #0 0x00007f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 350 slot = reg_get_entry(hash, _reg_htable->htsize); Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.132.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.10.3-10.el6_4.6.x86_64 libcom_err-1.41.12-18.el6.x86_64 libgcc-4.4.7-4.el6.x86_64 libselinux-2.0.94-5.3.el6_4.1.x86_64 libtool-ltdl-2.2.6-15.5.el6.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-14.el6.x86_64 mysql-connector-odbc-5.1.5r1144-7.el6.x86_64 mysql-libs-5.1.73-3.el6_5.x86_64 nss-softokn-freebl-3.14.3-9.el6.x86_64 openssl-1.0.1e-16.el6_5.4.x86_64 unixODBC-2.2.14-12.el6_3.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x00007f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 #1 0x00007f0e5cf36c71 in uac_reg_lookup (msg=0x7f0e6271e790, src=0x7fff59734b00, dst=0x7f0e6267c950, mode=0) at uac_reg.c:924 #2 0x00007f0e5cf2f991 in w_uac_reg_lookup (msg=0x7f0e6271e790, src=0x7f0e6267e0d0 "\a", dst=0x7f0e6267c950 "\006") at uac.c:560 #3 0x0000000000419bf6 in do_action (h=0x7fff59735690, a=0x7f0e6267f1e0, msg=0x7f0e6271e790) at action.c:1111 #4 0x0000000000422878 in run_actions (h=0x7fff59735690, a=0x7f0e6267e510, msg=0x7f0e6271e790) at action.c:1599 #5 0x0000000000417900 in do_action (h=0x7fff59735690, a=0x7f0e62664aa0, msg=0x7f0e6271e790) at action.c:715 #6 0x0000000000422878 in run_actions (h=0x7fff59735690, a=0x7f0e6265d3b8, msg=0x7f0e6271e790) at action.c:1599 #7 0x0000000000423017 in run_top_route (a=0x7f0e6265d3b8, msg=0x7f0e6271e790, c=0x0) at action.c:1685 #8 0x00000000004a5153 in receive_msg ( buf=0x7f0e570d0168 "MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@gatitos."..., len=585, rcv_info=0x7f0e570cfe90) at receive.c:212 #9 0x00007f0e5c8e5802 in ws_frame_receive (data=0x7fff59735a40) at ws_frame.c:652 #10 0x000000000045531c in sr_event_exec (type=10, data=0x7fff59735a40) at events.c:254 #11 0x000000000052e04e in ws_process_msg ( tcpbuf=0x7f0e570d0160 "\201\376\002I\032\327\302\344MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@"..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1146 #12 0x000000000052e21f in receive_tcp_msg ( tcpbuf=0x7f0e570d0160 "\201\376\002I\032\327\302\344MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@"..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1182 #13 0x000000000052f2f3 in tcp_read_req (con=0x7f0e570cfe78, bytes_read=0x7fff59735cac, read_flags=0x7fff59735ca4) at tcp_read.c:1383 #14 0x0000000000530d27 in handle_io (fm=0x7f0e62724d30, events=1, idx=-1) at tcp_read.c:1617 #15 0x00000000005296c0 in io_wait_loop_epoll (h=0x8df220, t=2, repeat=0) at io_wait.h:1092 #16 0x0000000000531650 in tcp_receive_loop (unix_sock=53) at tcp_read.c:1728 #17 0x0000000000523c21 in tcp_init_children () at tcp_main.c:4959 #18 0x000000000046d6a3 in main_loop () at main.c:1702 #19 0x000000000047030b in main (argc=13, argv=0x7fff59736178) at main.c:2533 Is this a known bug?

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message 484
by Slava Bendersky 09 Apr '14

09 Apr '14

Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ? <--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" <sip:1300@networklab.loc>;tag=as0a530a8d To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367(a)networklab.loc CSeq: 102 OPTIONS Contact: <sip:1300@192.168.100.145:5062>;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 Slava.

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cnxcc
by Kelvin Chua 09 Apr '14

09 Apr '14

i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.000000,consumed_amount:0.000000; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua

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rfc: evapi architecture and message format
by Daniel-Constantin Mierla 09 Apr '14

09 Apr '14

Hello, (cross-posting because the results are impacting users) recently I added a new module named evapi: - http://www.kamailio.org/docs/modules/devel/modules/evapi.html The purpose is to able to send custom messages to applications that connect to kamailio via tcp on a special port (different than sip port) as well as receive messages and react on them. One of the first use case we look at is integration with cgrates for real time billing. For example: - send notification when a new call comes in and cgrates will return allow/reject - send notification when the call starts so cgrates can start billing - send notification when the call ends so cgrates will close bllling The module is now in quite good shape, sending and receiving messages. Sending is done from config file, in the future some modules might do it from inside. Received message is passed to config file via an event_route and variables. Now, the point of this discussion is to debate following points: 1) should there be a specific format used for this messages? The examples in the README use json. The module can encapsulate them in netstring format when sending over tcp to be easy to parse and read the message at once. This is optional, the message can be sent as it is 2) now messages are sent to all connected applications, should be there some subscription mechanism per event? If yes, then there has to be a way to specify the event, so point 1) might need a specific format 3) handle response - should be there a defined api that will say allow, reject, drop? I was thinking that the return could be some script for an embedded interpreter (e.g., Lua) that can be executed from config file. Or, alternative, manually scripted in the event route, so each user defines its own api 3) other suggestions for what should be discussed? Looking forward to opinions regarding these matters. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Kamailio IMS dnszone
by Andrea Meroni 09 Apr '14

09 Apr '14

Issue fixed: i just had to add the following line dns_try_naptr=yes to all the kamailio.cfg files of each of my IMS entity. I suggest to modify accordingly the config files in each of the examples (pcscf, icscf, scscf) in the master tree in order to make them properly working with the kamailio_ims_dnszone example Regards, Andrea

1 0

Dialog: Keep-alive issue:
by Vassilis Radis 08 Apr '14

08 Apr '14

Hello, I am trying to use the dialog module for dealing with some cases where clients do not send BYE messages correctly and just disappear. For this reason i use the ka_timer and ka_interval along with ka-dst and ka-src attributes. I am using the master branch. I have setup the following test: I have setup a sipp instance as a callee uas configured to discard OPTIONS messages. Kamailio serves as a proxy between this sipp instance and a VoIP device registered to kamailio. Kamailio is also configured to send every call to the sipp uas with stateful proxing and dialog support: CALLER ------> KAMAILIO -----> CALLEE (sipp / ignores OPTIONS msg) I have the following issue: 1. Dialog does send the OPTIONS messages to caller and callee, and as expected, only the caller responds (with a 200 OK response). But the dialog module never terminates the dialog as it should (because the callee never responds to the OPTIONS msg). Instead it keeps sending those OPTIONS for ever. It is like there is no timer entry in the timer list for those OPTIONS. I looked a bit in the source code and I am trying to find where a timeout for those keep alives is a)set and b)handled . I see that the dlg_timer_routine is called every second but after turning debugging on, i see that the only timer reported is the default timeout for the call which i ve set to 3600 secs in the module parameters. Any insight or advice? Thanks a lot.

2 3

uac_replace_from
by Kelvin Chua 08 Apr '14

08 Apr '14

Has anybody experienced having uac_replace_from() concatenate the old from: value after the new domain? for example: From: "user1" <user1(a)domain.com> becomes From: "user2" <user2(a)domain.comuser1> Kelvin Chua

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Kamailio IMS dnszone
by Andrea Meroni 08 Apr '14

08 Apr '14

Hi all, i have a strange problem: i configured a basic IMS infrastructure Kamailio-based, with a DNS server (bind9) running on a virtual machine with IP address 192.168.100.55. Whenever i try to register a user, the first registration almost always fails, while the second is successful. I attached my dnszone config used with bind9 and the pcaps of the 2 regs; my questions are - is the IMS dnszone configuration correct? - the problems i got are just due to latency issues or have i misconfigured something? Regards, Andrea

1 0

RTP not coming via RTPProxy
by varun pratapsingh 08 Apr '14

08 Apr '14

HI All, I am using the Kamailio 4.1.2 with RTPProxy 1.2.1 as a SIP Proxy. I want to ensure that all my sip and media (RTP) passes through the SIP Proxy (Kamailio + RTP Proxy) while we my SIP Client sends any SIP and RTP to SIP Server (asterisk). So it is like below: SIP Agent <--------SIP+RTP---------->SIP Proxy(Kamailio+RTPProxy)<------SIP+RTP-------->SIP Server(Asterisk). Now it is working fine with my current configuration on Kamailio with RTPProxy with SIP and RTP both if my SIP Agent is on the 802.11 LAN that is my PC have local IP address like 192.168.1.2. Now I am facing the problem in the case where my PC is accessing the internet through the USB data card (or Internet USB dongle). Here my PC IP is like 116.203.51.209. In this case my SIP is successfully passes through the SIP Proxy and works good. But in this case my RTP is does not passes through the SIP Proxy (RTPProxy). The SIP Proxy(Kamailio + RTPPRoxy) and SIP Server is running on the public IP. Please find attached my kamailio.cfg. Please let me know what is the issue. Thanks and Regards Varun

1 0

No audio issue
by jaflong jaflong 08 Apr '14

08 Apr '14

Hi, I am at the point where connection is established and no apparent errors are reported. However audio is not output. The rtp traffic seems to be transfering between the points as conclueded because Asterisk debug log shows Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021868, ts 221760, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001383, ts 1917269534, len 000160) Sent RTP packet to 10.1.xxx.xxx41143 (via ICE) (type 08, seq 021869, ts 221920, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001384, ts 1917269694, len 000160) Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021870, ts 222080, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001385, ts 1917269854, len 000160) And the browser machine on the other endpoint on a tcpdump does shows traffic on the port (41143) What could be causing there to be no audio? This is the connected sdp =0 o=root 350315728 350315728 IN IP4 10.31.xxx.xxx s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.31.xxx.xxx t=0 0 m=audio 24316 UDP/TLS/RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:1c5c5d52130f06fd70e1e23f0d6323f2 a=ice-pwd:12611b8146599a9019d59b4b649a7970 a=candidate:Ha1f026f 1 UDP 2130706431 10.31.xxx.xxx 24316 typ host a=candidate:Ha1f026f 2 UDP 2130706430 10.31.xxx.xxx 24317 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv

2 2

P-CSCF using MYSQL database
by Andrea Meroni 08 Apr '14

08 Apr '14

Hi all, i am trying to configure an IMS Core Kamailio-based: while configuring the P-CSCF, i set up the parameters 'db_url' and 'db_mode' of the module ims_usrloc_pcscf respectively to 'mysql://kamailio:kamailiorw@localhost/kamailio' and 1 (because i want a DB for persistence). I had created successfully the DB using kamdbctl, and loaded the module 'db_mysql' in kamailio.cfg. Then i tried to run Kamailio, and i got this error: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query: driver error on query: Unknown column 'aor' in 'field list' What am i missing? Consider then when i created the DB i answer yes to all the questions about creating additional tables ... Andrea

2 24

sql_xquery() and xavp checks
by Alex Balashov 08 Apr '14

08 Apr '14

Hi, When using sql_xquery() like this: sql_xquery("ca", "SELECT * FROM gateways", "gateways"); ... what's a good way to check if any rows were returned? Since one does not have a $dbr(gateways=>rows) value in this scenario, what should one do? - is_avp_set("$xavp(gateways=>id)")) does not appear to operate on XAVPs, or at least, the fixup functions reject them: ERROR: avpops [avpops.c:935]: fixup_is_avp_set(): bad attribute name <$xavp(gateways=>id)> - the 'defined' operator does not appear to return a negative condition here: if(!defined $xavp(gateways=>id)) This condition evaluates to true. Much appreciated! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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