- sr-users - kamailio.org

How to change To header URI from Kamailio?
by Arun NV 05 Dec '17

05 Dec '17

Hi, I am using *Kamailio 4.4*. I would like to forward the request to a different port number of my endpoint. I have changed the destination *URI *and the *INVITE *correctly reached to the new port. But the *To header *in the *INVITE *request has the old port. so the endpoint is not responding to the request. Then I tried to remove and replace *To header* using *remove_hf("To")* and* insert_hf("To: $var(modified_to_header) \r\n");* functions. But the to header not changed. So, is there any way to change the *To header URI*.

2 1

Dialog Module Hash Size vs Number of Dialogs
by Mark Blackford 04 Dec '17

04 Dec '17

Hello, I am trying to properly size the use of the Dialog Module hash for our implementation using: modparam("dialog", "hash_size", <number that is power of two>) However, in my testing, I have been unable to figure out the relationship between the hash size and a number of dialogs I need to support. I think the hash size is specifying a memory block in kB, but even setting the hash size to a very tiny number does stop me from creating hundreds of dialogs. Is there a way to determine a relationship between the hash size and a rough number of dialogs that would be expected? An example of a a dialog looks like this from kamctl: [root@kamailio01 ~]# kamctl dialog show dialog memory records dialog:: hash=22:70 state:: 4 ref_count:: 2 timestart:: 1512151205 timeout:: 36083666 callid:: 0gQAAC8WAAACBAAALxYAAClws2wyL8GE+CSgRY7HIhmg9ZUIISZad46ntOPng3iPIcLaxzLFaytRTI7M0Bzz0g--(a)10.155.8.40 from_uri:: sip:b53667d44239457fbc94fc2f4c4e25a6@sip.dcs-staging.net from_tag:: 10.155.8.40+1+689d7e5e+8fcf481a caller_contact:: sip:43f0ae1480846185e8803f21e9f2b721@10.155.8.40:5060 ;transport=udp caller_cseq:: 24115 caller_route_set:: caller_bind_addr:: udp:10.155.8.11:5060 callee_bind_addr:: udp:10.155.8.11:5060 to_uri:: sip:2052773090@sip.dcs-staging.net to_tag:: sip+1+bdcd0004+2038f37c callee_contact:: sip:ca2013e84f10348a1cc825c12562bde7@10.155.8.40:5060 ;transport=udp callee_cseq:: 0 callee_route_set:: Thanks! -- Mark Blackford Digium Cloud Services 678.230.8769

3 5

How to use ruri-pattern with allow_trusted?
by voipspace voipspace 04 Dec '17

04 Dec '17

Hi Have used allow_trused to pass the allowed ip. Now I want to check the request has a correct tech prefix. With an entry in ruri-patten column in the trusted table for example 75622#. So if calls are to olny be allowed when the "to" destination starts with 75622#. Regards

1 0

max_contacts does not work as expected
by Gholamreza Sabery 04 Dec '17

04 Dec '17

I have a Kamailio server that has: modparam("registrar", "max_contacts", 2) But recently I noticed some users have more than two records in the location table. For example one of the users has three different records with nearly the same information. The records are as follows. Is this behavior normal? (I am using Kamailio 4XX). *************************** 1. row *************************** id: 171296745 ruid: uloc-2c69-59db6c73-7ca-132d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: XpCnmaAhRF cseq: 43 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 14 *************************** 2. row *************************** id: 171296746 ruid: uloc-2c69-59db6c73-7f2-eb5d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: BzMKuyJN8C cseq: 50 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 23 *************************** 3. row *************************** id: 171296747 ruid: uloc-2c69-59db6c73-80f-541d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: 3p9P4PgZSv cseq: 51 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 15

4 12

Dialog Module Hash Size vs Number of Dialogs
by Mark Blackford 04 Dec '17

04 Dec '17

Hello, I am trying to properly size the use of the Dialog Module hash table for Kamailio 4.1.3 implementation using: modparam("dialog", "hash_size", <number that is power of two>) However, in my testing, I have been unable to figure out the relationship between the hash size and a number of dialogs I need to support. I think the hash size is specifying a memory block in kB, but even setting the hash size to a very tiny number does stop me from creating hundreds of dialogs. Is there a way to determine a relationship between the hash size and a rough number of dialogs that would be expected? Thanks, -- Mark Blackford Digium Cloud Services 678.230.8769

2 1

How to get call on an extension, which is registered when a call towards it reach kamailio. How to modify pending session parameters in kamailio?
by Arun NV 04 Dec '17

04 Dec '17

Hi, I am using *Kamailio 4.4* as the proxy with my *Asteri**sk *server. I am trying to develop a scenario where an extension gets registered on asterisk via *Kamailio *when it receives a push notification. This push notification is sent to the sip extension when a call towards this extension reaches to the *Kamailio*. For example, suppose there is two SIP extension(* extension 1 and extension 2*) registered on *Asterisk *via Kamailio. When a call from *extension 1* reaches the asterisk, it forwards the *INVITE *request towards *extension 2* via Kamailio.Kamailio will try to forward it to extension 2. suppose the *extension 2* is not able to receive the *INVITE *request from *Kamailio*. When extension two receive a push notification, it will register on asterisk. So I need to get the call on *extension 2 *through the new registration. We are trying to simulate registration of extension to the asterisk when receiving the push notification. First, we registered *extension 2 *and disconnected the network. Then we tried to register the same extension when a call from *extension 1 *reaches to Kamailio. This is a simulation of push-based registration since an extension that receives the push will attempt to register when an incoming call is received. When asterisk sends *INVITE *request to Kamailio, it immediately responded with *100 trying* provisional response. This *100* response by Kamailio towards asterisk prevents asterisk from re-transmitting the INVITE. Then Kamailio tried to send and retransmit the packet to *extension 2*, which does not have network access. This* extension 2 *was on port number *24071*. Even after successful registration(in port *59995*) of the *extension 2*, Kamailio continued to transmit the packets to the old port. After that, we have configured Kamailio in a way that it won't send an immeadeate provisional reply(*100 trying* ) for *INVITE *request. Here Kamailio is not immediately sending *100 trying* message to *Asterisk*. This forces Asterisk to re-transmit. Asterisk was found to retransmit the same packets. However, even after the successful registration of* extension 2*, asterisk continued to send the old invite to Kamailio not the new one to the latest port. This is the problem for me since push relies on the *INVITE *reaching the phone at the correct port number. So, is there other good approaches to solve this issue? One thing I would like to try is modifying the pending *INVITE *request towards old registered port with the new port details when new registration reaches to Kamailio. Can I get the ongoing requests from Kamailio? Please suggest a viable solution. Thank you, Arunbalan

2 1

Using dispatcher's algorithm 12
by George Diamantopoulos 04 Dec '17

04 Dec '17

Hello all, I've been trying to use dispatcher's algorithm "12" to fork REGISTER messages constructed with uac module to all destinations within a set. However, the request is routed to only one destination. Is anything else needed in addition to calling ds_select_dst("0", "12") in the configuration script? Thanks. BR, George

3 5

bash script to install siremis&kamailio
by Atux Atux 04 Dec '17

04 Dec '17

Hi. i would like to ask if someone has found any bash script that could install all at once for Kamailio and Siremis in ubuntu/debian or even in centos.

2 1

dmq issues
by Aidar Kamalov 03 Dec '17

03 Dec '17

Hello. I have 4 kamailio nodes with same config: modparam("dmq", "server_address", "sip:LOCALIP:5050") modparam("dmq", "notification_address", "sip:192.168.8.213:5050") modparam("dmq", "notification_address", "sip:192.168.10.54:5050") modparam("dmq", "notification_address", "sip:192.168.107.195:5050") modparam("dmq", "notification_address", "sip:192.168.150.225:5050") modparam("dmq", "num_workers", 6) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "batch_size", 2000) modparam("dmq_usrloc", "batch_usleep", 1000) regularly (don't know why) I've got issues - replication is broken, for example now at all nodes I have: # kamcmd dmq.list_nodes { host: 192.168.8.213 port: 5050 resolved_ip: 192.168.8.213 status: 2 last_notification: 0 local: 0 } { host: 192.168.10.54 port: 5050 resolved_ip: 192.168.10.54 status: 2 last_notification: 0 local: 0 } { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 8 last_notification: 0 local: 0 } { host: 192.168.107.195 port: 5050 resolved_ip: 192.168.107.195 status: 2 last_notification: 0 local: 1 } but at node 192.168.150.225: # kamcmd dmq.list_nodes { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 2 last_notification: 0 local: 0 } { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 2 last_notification: 0 local: 1 } nodes restart doesn't help. but if i exec systemctl stop kamailio && sleep 15 && systemctl start kamailio it is partially solve my issue - dmq.list_nodes show all nodes, but at 192.168.150.225 i see not all user locations: they are added gradually There is no errors in logs Dec 1 08:28:37 sipufa /usr/sbin/kamailio[18252]: NOTICE: <script>: ROOT: KDMQ notification_peer [15f3edf9258014d1-7114(a)192.168.8.213] Dec 1 08:28:38 sipufa /usr/sbin/kamailio[18271]: NOTICE: <script>: ROOT: KDMQ notification_peer [7acb9eae627482f8-87828(a)192.168.10.54] Dec 1 08:28:39 sipufa /usr/sbin/kamailio[18250]: NOTICE: <script>: ROOT: KDMQ notification_peer [621ad9f474b335e2-18272(a)192.168.150.225] Dec 1 08:28:40 sipufa /usr/sbin/kamailio[18252]: NOTICE: <script>: ROOT: KDMQ notification_peer [7b817f6a6244f923-8872(a)192.168.107.195] Dec 1 08:28:47 sipufa /usr/sbin/kamailio[18271]: NOTICE: <script>: ROOT: KDMQ usrloc [7acb9eae627482f2-87794(a)192.168.10.54] Dec 1 08:29:20 sipufa /usr/sbin/kamailio[18251]: NOTICE: <script>: ROOT: KDMQ usrloc [7acb9eae627482ef-87790(a)192.168.10.54] -- Aydar A. Kamalov

3 8

kamailio as application server
by eyas barhouk 01 Dec '17

01 Dec '17

Hello if i need to use kamailio as an application server like mmtel for example how could i do that and what is the features that kamailio have if i want to use it as application server in another word what is the type of application server it could be act like (mmtel - presence) ? and thanks for help

2 1

In need of a simple, stateless 302 redirect for INVITE
by Robert 01 Dec '17

01 Dec '17

Hello again, I am not a SIP expert, and I’m new to Kamailio, really enjoying learning it, but I’ve hit a blocker and I’m pretty sure I’ve missed something really simple. I’m looking to implement the Redirect Server from here: https://tools.ietf.org/html/rfc3665#section-3.6 <https://tools.ietf.org/html/rfc3665#section-3.6> (the Avaya Session Manager being Alice), Kamailio as the Redirect Server. My configuration is working fine, and I can see the 302 going back (to Avaya SM), I then get the ACK I’m expecting. The incoming SIP request contains a number of fields, including a Contact header, which I’m told SM needs on the return. But the send_reply() or sl_send_reply() doesn’t return the Contact header, and I am unable to manually set it, or any other header, using any of the append_hf(), append_to_reply() etc. mechanisms. I’m executing a msg_apply_changes() which succeeds, but no Contact header is returned. I’ve tried an sl_send_reply(“100”, “Trying”) before any other processing, and Wireshark confirms that the SIP response is basically identical to the 302 response sent after processing… The goal is to run this stateless, I’m assuming this is possible - it certainly feels like it should be. I’ve re-read the intros & tutorials, and other reading material such as this http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf <http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf> but, this, even my best Google-fu has left me utterly stumped… The sl_send_reply documentation https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply <https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply> states “If the code is in the range 300-399 (redirect reply), the current destination set is appended to the reply as Contact headers” but the Contact header is definitely not being set when I use this. Apologies in advance if eyes are rolling and this is really obvious, or (and?) that my lack of SIP knowledge is shining through - any tips or guidance would be appreciated. I certainly feel overwhelmed by the number of modules and the interplay between them. If the answers to this, and other things along these lines are contained in the admin book, that’s fab, I’ve just filled the form out :) Cheers - Robert...

3 3

DBURL password in clear
by Robert 01 Dec '17

01 Dec '17

Hello, I’m working for a UK high street bank and our Kamailio implementation has been challenged because we’ve got database passwords held in clear in the configuration file. I am unable to find any examples of where this has been worked around, there doesn’t seem to be any module or configuration means of supplying a variable in the modparam() entry that is expanded a startup. The security tutorials only seem to relate to the SIP level of security, not Kamailio as a platform. My requirement is simple, I need to be able to supply a password via means such as loading a variable from a run-once script at start up, or a module. The ideal would be to be able to read in a Docker secret :) I am by no means a Kamailio expert, so apologies in advance if this is a mindblowingly basic thing to achieve, but I do feel I’ve exhausted the Kamailio documentation, wiki etc. and all the goodness Google usually has to offer and drawn a blank. Sincere thanks in advance for any assistance. Cheers - Robert...

5 14

Dispatcher latency stats monitoring with statsd
by Julien Chavanton 01 Dec '17

01 Dec '17

Not sure if anyone is using "latency measurement" feature in Kamailio dispatcher module, I thought I could share a very simple Python script that I am using to monitor remote gateway using this. Sample Script is attached, it will post data to statsd and is working in cron The feature can be enabled : 3.27. ds_ping_latency_stats (int) Enable latency measurement when pinging nodes If set to 0, disable latency measurement. If set to 1, enable latency measurement https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc… This is one use case of this feature, another one will be to load balance based on congestion detected on latency, I am planning to share a lab experiment in a little while. Regards

2 1

[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 30 Nov '17

30 Nov '17

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2018 takes place 3-4 February 2018 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 4th of February 2018. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 30th of November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM18/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2018/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 3rd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on the 2nd of February 2018 http://wiki.xmpp.org/web/Summit_22 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 2nd of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

3 2

blf in kamailio dmq cluster
by Aidar Kamalov 30 Nov '17

30 Nov '17

Hello, I have two(or more) kamailio servers with dmq module and dmq_usrloc - sip1.example.com and sip2.example.com publish and subcribe replication works too. BLF works fine if clients connected to the same kamailio server, because in active_watchers table stored with same domain sip1.example.com But what to do when call going from one server to second? Is there any configuration to ignore domain in pusblish, subscribe, notify, etc? -- Aydar A. Kamalov

3 7

OpenSIPS-devel not starting (TCP Error)
by SamyGo 29 Nov '17

29 Nov '17

Hi, I just installed the head version from git. version: opensips 2.4.0-dev (x86_64/linux) flags: STATS: On, SHM_EXTRA_STATS, EXTRA_DEBUG, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: d1a4419 main.c compiled on 19:53:20 Nov 28 2017 with gcc 4.8 It was compiled with EXTRA_DEBUG and DBG_TCPCON flags; While starting opensips with the residential vanilla script it gives following error: Nov 28 20:05:52 [30899] ERROR:core:tcp_init: oom con hist Nov 28 20:05:52 [30899] CRITICAL:core:main: could not initialize tcp Nov 28 20:05:52 [30899] INFO:core:cleanup: cleanup Nov 28 20:05:52 [30899] DBG:core:shm_mem_destroy: destroying the shared memory lock Nov 28 20:05:52 [30899] NOTICE:core:main: Exiting.... already running Even if I remove the proto_tcp, and tcp related things this still gives error and doesn't start. Kindly help. Regards, Sammy

2 3

parse_from_header(): bad From header
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. We experience the problem with malformed messages. First of all, I want to say, that most of time kamailio works well and nothing happens that can drop sessions. But from time-to-time something changes (may be in the network) and kamailio receives requests with malformed headers (To or From hfs). The schema: uplinks -> kamailio <-> routing server (asterisk) Malformed messages were received only from asterisk server. Sip debug on asterisk showed that messages were transmitted correctly and headers were nice. But kamailio obtains changed (broken) packets or perhaps can't read them properly. How does it look like: Nov 10 12:37:06 kamailio-name kamailio[965]: INFO: <script>: Going to NATMANAGE for BYE from sip:useragent7122@kamailio_address:5068 (IP=wss:client_address:62881) - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: NOTICE: <script>: Relaying request to <null> - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: uac [replace.c:783]: restore_uris_reply(): failed to find/parse FROM hdr Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: Skip manage DEVICE_STATE for BYE from <null> (IP=udp:asterisk_address:50600) with (200 - OK) - R=<null> ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: ------------------------------------MANAGE BYE by DEVICE_STATE_BYE_MANAGE route - 200 - OK M=BYE IP=udp:asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine_funcs.c:331]: get_from_tag(): failed to parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine.c:2252]: rtpp_function_call(): can't get From tag Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header How the packet looks like: ACK sip:tt7etmau@bb0pd5t63b26.invalid;alias=client_address~57573~6;transport=ws SIP/2.0 Via: SIP/2.0/UDP asterisk_address:50600;branch=z9hG4bK117106ca;rport Route: <sip:kam2.domain.com:5068 ;nat=yes;transport=udp;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331>,<sip:kam2.callision.com:5068 ;nat=yes;transport=ws;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331> Max-Forwards: 70 From: "19172423539" <sip:19172423539@asterisk_address:50600>;tag=as06869a5d To: <sip:useragent7122@10.0.1.18:5068>;tag=nmv7kmpau3 Contact: <sip:19172423539@asterisk_address:50600> Call-ID: 629d5a132c194c536b5f5c1a2a3c6e32@asterisk_address:50600 CSeq: 102 ACK User-Agent: asterisk PBX Content-Length: 0 The transport is used to send messages between asterisk and kamailio is TCP. TCP configurations: tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 I found the idea, that we need to change following parameters: modparam("uac","restore_mode","auto") modparam("uac","restore_dlg",1) to: modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) I did that, now it looks like all is fine, but I think it can get back. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua>

3 5

Source socket/port for in-dialog requests
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. My apologies for so frequent appealing to you. I'm trying to solve problem with ending of sessions. The problem consists of no 200 OK coming from uplink on our BYE requests. Topology. First leg: Webrtc client <-wss-> kamailio <-sip tcp-> asterisk routing server Second leg: Uplink <-sip tcp-> kamailio <-sip tcp-> asterisk routing server The problem appears only in case when dialog was ended by webrtc client. The first leg (dialog) of the call ends nice, without any hints on problem. But the second leg (with uplink) has problem with no 200 OK (coming from uplink) on BYE request coming from asterisk. Handshake between asterisk server and uplink establishes properly. It looks like: uplink.domain.com:5060 -> INVITE tcp -> our.kamailio.server:5060 -> INVITE tcp -> our.asterisk.server:5060 . uplink.domain.com:5060 <- 100 trying tcp <- our.kamailio.server:5060 . uplink.domain.com:5060 <- 180 ringing tcp <- our.kamailio.server:5060 <- 180 ringing tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 <- 200 OK tcp <- our.kamailio.server:5060 <- 200 OK tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 -> ACK tcp -> our.kamailio.server:5060 -> ACK tcp -> our.asterisk.server:5060 . uplink.domain.com <-media stream-> rtpengine <-media stream-> our.asterisk.server . Here kamailio starts to use random source port for relaying in-dialog BYE uplink.domain.com:5060 <- BYE tcp <- our.kamailio.server:45355 <- BYE tcp <- our.asterisk.server:5060 . And here the leg with uplink is expected to end with 200 OK (coming from uplink proxy). But uplink doesn't answer at all. We requested a tcpdump from uplink to see how packets are forwared from their side. And I saw that that 200 OK tries to be sent within first tcp session - to 5060 port of kamailio server. But sngrep on our side shows nothing, nothing appears in kamailio log, so 200 OK can't reach the 5060 socket because of some transport problem I think. Top via hf of our BYE request, contains kamailio record = uplink.domain.com:5060 My main thought on this count is to suppress using of random source ports for in-dialog requests, this behaviour looks irrelevant. We have turned on mhomed parameter (mhomed=1). And cookbook says: "Set the server to try to locate outbound interface on multihomed host. This parameter affects the selection of the outgoing socket for forwarding requests." But, there is a big but - we can not turn of this parameter for this moment, because routing script made with using of this one. Kamilio works on AWS EC2 platform. It has following IP address schema: private address atouched to container -> public address assinged to this private address (so this is standard ip address schema for AWS containers). For this example private address will be: 10.0.0.1 and public address will be: 100.0.0.1 Configuration - main parameters (all values are changed for an example): advertised_address=our.kamailio.server advertised_port=5060 alias=our.kamailio.server alias=our.kamailio.server:5060 alias=100.0.0.1 alias=100.0.0.1:5060 alias=10.0.0.1 alias=10.0.0.1:5060 auto_aliases=no port=5060 listen=100.0.0.1 listen=10.0.0.1 listen=tcp:100.0.0.1:5060 listen=tcp:10.0.0.1:5060 mhomed=1 fork=yes fork_delay=5000 children=6 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 One of the solutions to this problem was to add following inside the relaying route block: $fs = "tcp:" + "10.0.0.1" + ":5060"; if (!t_relay()) { sl_reply_error(); } But it seems to me it's not a smart solution. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 1

Best way to do routing against a list of gateways
by voipspace voipspace 29 Nov '17

29 Nov '17

3 6

allow_trusted not passing
by voipspace voipspace 29 Nov '17

29 Nov '17

4 5

URL problems connecting to external (Azure) MySQL database
by Floimair Florian 29 Nov '17

29 Nov '17

Hi! I'm having troubles connecting Kamailio to an external MySQL database (on Microsoft Azure Cloud). Initially creating the database with kamctl worked fine, but accessing it at runtime via db_url fails. A somewhat unusual thing about the connection to a MySQL database on Azure is that the username contains an '@' character. This leads me to believe that this case is somewhat unhandled when parsing the database URI. Trying to connect to something like modparam("sipcapture", "db_url", "mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db") Results in errors starting Kamailio ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'clouddb' (-2) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool ERROR: sipcapture [sipcapture.c:1146]: child_init(): unable to connect to database [mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db] from capture_mode param. Is there a trick on how to achieve a connection in this case like some sort of escape character probably? Mit freundlichen Grüßen Florian Floimair (IMS) 1568

2 4

lcr_id vs lcr_count
by Patrick Wakano 29 Nov '17

29 Nov '17

Hello Kamailio list! I've been working with the LCR module and I wanted to use the lcr_id to match some ID's I have somewhere else. However, due to the lcr_count my lcr_ids must be less than the lcr_count. I understand the reason behind the lcr_count, what I think is a mistake is to have the lcr_id dependent of the lcr_count. Ideally we should be able to use any lcr_id value, just observing that the total amount of lcr groups should not exceed the lcr_count. The problem is this if that appears in the load_gws, to_gw and from_gw functions in lcr_mod.c: if ((lcr_id < 1) || (lcr_id > lcr_count_param)) { LM_ERR("invalid lcr_id parameter value %d\n", lcr_id); return -1; } I ended up using another table to make the conversion of ids, but this is horrible.... Any other reason to force the lcr_id to be less than the lcr_count? Best Regards, Patrick Wakano

3 3

how to add new media server in dispatcher table through "ANSIBLE"
by Narayan P 29 Nov '17

29 Nov '17

Hello Experts, I am new to ansible and deploying kamailio through ansible.Sucessfully deployed kamailio but my requirement is to add media server through ansible.As I am using dispatcher module in kamailio,not able to find any script that directly insert one media server IP and port into dispatcher table from ansible.(I am using ansible in one system and deploying kamailio in other). Any suggestion will be highly appreciated as always. Thanks, Narayan +91-9035438928

3 4

Re: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue.
by otron2016＠gmail.com 29 Nov '17

29 Nov '17

Through Firefox likely means via websockets (wss) rather than via the sip port. Many things could interfere with the latter. Take heart, though: websockets is harder to set up so you're likely through the worst. Look at logs and see what iftop shows you about traffic on the server as you try to make sip connections. Also try your sip client with a known-good sip server (e.g. Iptel) just to make sure the problem isn't with the client. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/28/2017 6:39 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue. Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won’t work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don’t know 100% for sure. Thank you

1 0

Using SIP tools through Kamailio and Asterisk - chrome issue.
by Wilkins, Steve 28 Nov '17

28 Nov '17

Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won't work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don't know 100% for sure. Thank you

1 0

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