- sr-users - kamailio.org

uacreg question
by Lucas Alvarez 08 Nov '11

08 Nov '11

Hi, I have properly populate the table "uacreg" as says here: http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html and I'm loading the uac module, is there anything I could be missing? Because I'm not seeing any register packet being sent to my sip provider. Thanks in advance, Lucas Alvarez

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how to give integer avp name in pseudo variable?
by Juha Heinanen 08 Nov '11

08 Nov '11

i have not been able to figure out, how to give integer avp name in a pseudo variable. test $var(name) = "i:1000"; $avp($var(name)) = "value"; xlog("L_INFO", "avp i:1000 has value <$avp(i:1000)>\n"); xlog("L_INFO", "avp s:i1000 has value <$avp(s:i:1000)>\n"); produces to syslog Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp i:1000 has value <<null>> Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp s:i1000 has value <value> that is, $var(name) value i:1000 is interpreted as string name even when its syntax conforms to integer name. is this is a bug or how should value of $var(name) look like? -- juha

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how to give integer avp name in pseudo variable?
by Juha Heinanen 08 Nov '11

08 Nov '11

i have not been able to figure out, how to give integer avp name in a pseudo variable. test $var(name) = "i:1000"; $avp($var(name)) = "value"; xlog("L_INFO", "avp i:1000 has value <$avp(i:1000)>\n"); xlog("L_INFO", "avp s:i1000 has value <$avp(s:i:1000)>\n"); produces to syslog Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp i:1000 has value <<null>> Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp s:i1000 has value <value> that is, $var(name) value i:1000 is interpreted as string name even when its syntax conforms to integer name. is this is a bug or how should value of $var(name) look like? -- juha

1 0

Conditional uacreg
by Alex Balashov 08 Nov '11

08 Nov '11

Is there some way by which credentials can be loaded into the uacreg table, but not all of them used at once? The problem I have is a need to make these registrations active and inactive periodically, whereas the registration facility that the uac module provides is fully managed/largely unattended. Using $uac_req(...) is not an option because I need to capture the failure replies (which is not supported in TM) in order to do digest challenge authentication. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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SQL Server
by Jonny Klompas 08 Nov '11

08 Nov '11

Can Kamailio send routes to and /or receive routes from SQL Server? Is there any way for Kamailio to connect to SQL Server in general? Thanks Jonny

3 2

TOS=0 in kamailio.cfg
by DELEY Marion 07 Nov '11

07 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

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Re: [SR-Users] Fwd: Kamailio as SBC / LB Media-Servers farm
by Klaus Darilion 05 Nov '11

05 Nov '11

On 04.11.2011 10:57, Sammy Govind wrote: > SORRY for PM but this is not getting through to the users List due to size. I have cc-ed the list. > > ---------- Forwarded message ---------- > From: *Sammy Govind* <govoiper(a)gmail.com <mailto:govoiper@gmail.com>> > Date: Fri, Nov 4, 2011 at 1:55 PM > Subject: Re: [SR-Users] Kamailio as SBC / LB Media-Servers farm > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > > > Klaus: mhomed did work indeed - no need to force_send_socket() now :) Thanks > > I'm progressing abit. > > RTP-Proxy was run in bridged mode like this. > > *rtpproxy -l 109.XX.XXX.212/192.168.2.29 <http://192.168.2.29> -s > udp:localhost:22222 -F -d DBUG* See description of i,e flag: http://kamailio.org/docs/modules/3.2.x/modules/rtpproxy.html#id2535618 Thus in your setup 109.... is the "internal" interface and 192.... is the "external" interface. That's mixed. Thus I advice you to switch the order of IP addresses so that internal is 192... and external is 109... This is essential to use the i,e flags correctly. > I'm getting RTPstats published on Kamailio/RTPproxy syslog. > > I am using Echo() application in asterisk. So I've observed that my > audio went to asterisk and asterisk did indeed send audio back to the > RTPproxy assigned ports BUT the RTPproxy isn't forwarding the Audio back > to the client or may-be the audio is lost somewhere in between > RTPproxy<--->Client. /[See Wirshark Flow-Diagram taken > on-Kamailio/RTPproxy Server]/ In your trace you see, that RTP does not use the internal IP address, but only the external IP address. Thus I suspect that the RTP packets from Asterisk to the public rtpproxy interface do not even arrive at rtpproxy. > > The # DTMF signal sent from Client phone reached asterisk safely and > terminated the Echo() application. > > I did the ip_forwarding thingie too.-- somebody mentioned to do this if > using RTPproxy in bridged mode. This is bullshit. People seem to not understand IP at all. rtpproxy is forwarding in user mode - it receives on a UDP socket and sends on a UDP socket. Thus, kernel based forwarding is not involved when using rtpproxy. You can disable IP forwarding again. > #echo "1" >> /proc/sys/net/ipv4/ip_forward > #sysctl -p > > > Here are the corresponding Logs from Kamailio and rtpproxy. The thing is quite easy. Swap the IP addresses when starting rtpproxy and then configure Kamailio to call rtpproxy with the proper i,e flags (as described in my previous email). For example: if (has_body("application/sdp")) { if ($si == "192.168.2.75") { # message from internal to external rtpproxy_manage("ie"); } else { # message from external to internal rtpproxy_manage("ei"); } } Put above code block in the main route block and also in a reply route which will be activated (or in the default reply route). Put some xlog statements around the rtpproxy calls to verify the proper command is called. regards Klaus

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Kamailio 3.2 Acc Runtime error
by Nathaniel L Keeling 05 Nov '11

05 Nov '11

I am getting this error when I start kamailio 3.2 on Solaris 10. Any direction on solving this would be greatly appreciated. ERROR: load_module: could not open module </opt/kamailio-3.2/lib64/kamailio/modules_k/acc.so>: ld.so.1: kamailio: fatal: relocation error: file /opt/kamailio-3.2/lib64/kamailio/modules_k/acc.so: symbol timersub: referenced symbol not found Thanks Nathaniel

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Directory missing compiling dbtext module
by Pedro Antonio Vico Solano 05 Nov '11

05 Nov '11

3 2

CANCEL is not forwarded, if callee does not respond
by Mino Haluz 04 Nov '11

04 Nov '11

Hi, the scenario is like this: - user1 makes a call to user2 - proxy sends receives INVITE and sends it to user2 - user2 does not respond at all (firewall for example) - user1 after 5 seconds hangs up the call with CANCEL - proxy sends to user1 200 canceling CANCEL is not forwarded to user2. Ok this behaviour is maybe ok, but after 30 seconds Request timed out is sent to user1 (fr_inv_timer possibly hit). So if I am thinking well, first CANCEL did not match the transaction ? Why ? It this behaviour ok or do I have to search for something wrong in the config? Thanks, Mino

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Kamailio as SBC / LB Media-Servers farm
by Sammy Govind 04 Nov '11

04 Nov '11

Hello List, I've successfully configured Kamailio with a farm of asterisk servers in L/B | F/O mode. The topology is something like below: User<-->Kamailio<-->Asterisk-Servers Both kamailio and Asterisk serevrs are on public IPs *The problem:* Kamailio and Asterisk Servers need to be on Public IPs in order to fully handle NAT/media related issues. Whereas what I require is to have two interfaces at kamailio server, one is Public other is Private and All the Asterisk Servers should be on LAN/Private IPs. I've tried making a scenario as per my requirement (attached image) but there are some serious NAT and media issues. Can anyone help me how to handle NAT'd Asterisk servers behind Kamailio? Thanks and Best Regards. Sammy

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Kamailio as SBC / LB Media-Servers farm
by Sammy Govind 04 Nov '11

04 Nov '11

Hello List, I've successfully configured Kamailio with a farm of asterisk servers in L/B | F/O mode. The topology is something like below: User<-->Kamailio<-->Asterisk-Servers Both kamailio and Asterisk serevrs are on public IPs *The problem:* Kamailio and Asterisk Servers need to be on Public IPs in order to fully handle NAT/media related issues. Whereas what I require is to have two interfaces at kamailio server, one is Public other is Private and All the Asterisk Servers should be on LAN/Private IPs. I've tried making a scenario as per my requirement (attached image) but there are some serious NAT and media issues. Can anyone help me how to handle NAT'd Asterisk servers behind Kamailio? Thanks and Best Regards. Sammy

1 0

Issue with version table
by Lucas Alvarez 03 Nov '11

03 Nov '11

Hi, I've enabled xcap server in kamailio 3.1.3 and I'm stuck with this error: 0(5073) ERROR: <core> [db.c:408]: invalid version 0 for table subscriber found, expected 6 (check table structure and table "version") I've already checked the version table and the table_name = 'subscriber' has table_version = 6. Any help will be appreciated. Regards. Lucas

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Dispatcher Confusion (v3.2.0)
by Asgaroth 03 Nov '11

03 Nov '11

Hi All, Just been doing some testing with Kamailio 3.2 and the dispatcher module, and have some strange behaviour, can we confirm if it is expected behaviour or a bug of some sort. It appears that I can only set the dspatcher state via fifo command once, and not reset it again, here is an example: version: kamailio 3.2.0 (i386/linux) 94d3b8 flags: STATS: Off, USE_IPV6, 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 94d3b8 compiled on 12:34:43 Oct 20 2011 with gcc 4.1.2 Here we start with the dispatcher list as it was pulled from the database and currently loaded in memory. # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now I try to set one gateway inactive: # kamctl fifo ds_set_state i 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= All normal so far, now if I try to set the same gateway active again, the state still stays inactive, i dont seem to be able to change the state back to active. The same thing happens if I attempt to set this state to disabled and then try to set it back to active afterwards. The state does not change from what the first attempt was. # kamctl fifo ds_set_state a 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now I can try to set the state to inactive probing, that works. # kamctl fifo ds_set_state ip 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now if I try to disable the same destination: # kamctl fifo ds_set_state d 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= State still stays in Inactive/Probing mode. Now if I try to set the same gateway to active/probing mode, I cant: # kamctl fifo ds_set_state ap 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= The only way for me to get the state back to active after the above is to do a reload on the dispatch table: # kamctl dispatcher reload # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Is the dispatcher expected to behave as above? I'm still trying to understand the different states assuming the following options are set in the configuration file: modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_probing_mode", 0) AX = Active/No Probing (*) Gateway is used in ds_select_*, ds_next_* (*) No OPTIONS/INFO messages are sent to destination to check alive state This appears to be the normal operating state with no issues detected. AP = Active/Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) OPTIONS/INFO messages ARE sent to destination to check alive state This appears to be the state taken when a gateway times out and ds_mark_dst("p") is set in failure route, the gateway is "pinged" while down and set into AX mode once a response is recieved. IX = Inactive/No Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) No OPTIONS/INFO messages sent to destination This appears to be the state taken when a gateway times out and if ds_mark_dst("i") is set in failure route, this gateway is set inactive and no "pinging" is performed. IP = Inactive/Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) OPTIONS/INFO messages ARE sent to destination I'm not sure when this state would be reached unless it is set by fifo command. The ds_mark_dst function only allows "a","i" or "p". So the probing method would only become active is ds_probing_mode = 1. In my case it is 0, so I'm not sure if I would ever reach this state unless I forced it. What would happen here is the gateway came back alive once in "IP" state, would it go back to "AX" state? DX = Disabled/No Probing (*) Gateway is administratively set to down This is where manual intervention has occured and the destination has been set to administrativly down, for example, work being carried out on the destination gateway. All these states make sense, for the most part, I'm a little unsure of how IP/IX is meant to work unless you manually set it to "inactive" in your failure route. The main concern though is the the fifo commands dont appear to be working as expected, I can only set the state for a destination manually once, and not again after that, unless I perform a dispatcher reload. Has anyone else experienced this behaviour? Any comments/suggestions explanations are most welcome. Thanks

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Kamailio dinner next week in Cape Town, South Africa
by Daniel-Constantin Mierla 03 Nov '11

03 Nov '11

Hello, it happens to come across Cape Town for few days and I was wondering if there are folks using our project or doing VoIP in general that want to meet up for a dinner. The date for dinner is planned for next Tuesday, November 8, 2011. Place will be decided once I know if there are people coming and how many. Likewise with such events done in the past, it is going to be a dinner open for everyone that wants to join, each participant paying the drinks and food for himelf/herself. The goal is to meet and discuss latest trends in IP communication and what's up lately around our project. Here are some pictures from two past similar events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ So, if there are people on these mailing lists in the area of Cape Town next Tuesday and want to meet, please reply to me till Friday evening -- I will follow up privately to each person that is going to join with details about location and start time. Looking forward to meet some of you very soon! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

TOS=0 in kamailio.cfg
by DELEY Marion 03 Nov '11

03 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

1 0

is there help on mi commands in sercmd?
by Juha Heinanen 03 Nov '11

03 Nov '11

sercmd has help that lists the commands, one of which is mi. is there a way to list all mi sub-commands? i have tried ip-proxy_ctl> help mi Execute MI command sip-proxy_ctl> mi help error: 500 - command not available -- juha

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extra _ctl in ctl file
by Juha Heinanen 03 Nov '11

03 Nov '11

i just noticed that my sip-proxy_ctl (alias kamailio_ctl) command has stopped working. it now adds an extra _ctl to the socket file name: # sip-proxy_ctl ERROR: connect_unix_sock: connect(/tmp/sip-proxy_ctl_ctl): No such file or directory [2] any idea where that comes from or it is only me, who has this problem? i have not changed anything the command used to work fine earlier. -- juha

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Dialog / BYE
by Brandon Armstead 02 Nov '11

02 Nov '11

Hello, I am attempting to tear down a call with a BYE packet generated externally (kind of similar to Kamailio fifo dlg_end_dlg). Let me describe what I am trying to do in more depth and then I will continue to tell you the problem I think I am experiencing. [PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC] So the above layout is the normal call flow / structure of calls (incoming when originating from pstn) (outgoing when originating from uac). I then have an "external" host - I am attempting to generate a BYE to [CORE SIP Proxy] and have it go both directions [PSTN] + [UAC]. So far I am able to get the call to tear down in only a single direction (only kill call with PSTN) or (only kill call with UAC). I have not been able to kill both legs of the call. I have tried sending a single BYE and using some dlg_bye("all") magic. - does not work, only kills one leg of the call. I have tried sending a BYE message to SIP Proxy for each leg (two BYE). - does not work, only kills one leg of the call. If I alternate the later method's BYE packet and only send A or B and not both Leg A or B will be killed... I think I am running into a race condition where the dialog is destroyed before the 2nd BYE packet is processed. My Question - what am I doing wrong - or what is the best method to tackle this task? Thank you for all of your help and time in advance. Sincerely, Brandon Armstead

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Kamailio3.2 debian repo fails
by Benjamin Henrion 02 Nov '11

02 Nov '11

Hi, I have the following /etc/apt/sources.list: ======================================================== $ cat /etc/apt/sources.list deb http://ftp.debian.org/debian squeeze main contrib non-free deb-src http://ftp.debian.org/debian squeeze main contrib non-free deb http://deb.kamailio.org/kamailio32 squeeze main deb-src http://deb.kamailio.org/kamailio32 squeeze main ======================================================== Which generates the following error: ======================================================== $ apt-get update Ign http://deb.kamailio.org squeeze Release.gpg Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US.UTF-8 Ign http://deb.kamailio.org squeeze Release Ign http://deb.kamailio.org squeeze/main Sources Hit http://ftp.debian.org squeeze Release.gpg Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US Ign http://deb.kamailio.org squeeze/main amd64 Packages Ign http://deb.kamailio.org squeeze/main Sources Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US.UTF-8 Ign http://ftp.debian.org/debian/ squeeze/main Translation-en Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US.UTF-8 Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US.UTF-8 Hit http://ftp.debian.org squeeze Release Ign http://deb.kamailio.org squeeze/main amd64 Packages Err http://deb.kamailio.org squeeze/main Sources 404 Not Found Err http://deb.kamailio.org squeeze/main amd64 Packages 404 Not Found Hit http://ftp.debian.org squeeze/main Sources Hit http://ftp.debian.org squeeze/contrib Sources Hit http://ftp.debian.org squeeze/non-free Sources Hit http://ftp.debian.org squeeze/main amd64 Packages Hit http://ftp.debian.org squeeze/contrib amd64 Packages Hit http://ftp.debian.org squeeze/non-free amd64 Packages W: Failed to fetch http://deb.kamailio.org/kamailio32/dists/squeeze/main/source/Sources.gz 404 Not Found W: Failed to fetch http://deb.kamailio.org/kamailio32/dists/squeeze/main/binary-amd64/Packages… 404 Not Found E: Some index files failed to download, they have been ignored, or old ones used instead. ======================================================== Who is responsible for the debian packaging? Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-4148403 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

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Re: [SR-Users] Siremis: cannot enter domain in "New Subscriber"window
by Uwe Andersen 02 Nov '11

02 Nov '11

Hi Hugh, it works now. Thank you! Uwe > -----Ursprüngliche Nachricht----- > Von: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org]Im Auftrag von > Hugh Waite > Gesendet: Mittwoch, 2. November 2011 12:52 > An: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Betreff: Re: [SR-Users] Siremis: cannot enter domain in "New > Subscriber"window > > > Hi Uwe, > > Domains are configured in Server Services->Domain List. > These are the > domains which the kamailio server is responsible for. > When they have been entered there, they will appear in the drop-down > list when adding users. > > Regards, > Hugh > > On 02/11/2011 11:33, Uwe Andersen wrote: > > Hi, > > > > I want to add a new user. So I choose "add" from the > "Subscriber Management" page. > > On the "New Subscriber" page I can enter username, > password, email-address, etc. > > But the line for "domain" seems to be an empty dropdown entry. > > I cannot choose, add or edit anything. > > As a result I cannot save the new user because Siremis > complains about the empty domain field. > > > > After adding a user by command line("kamctl add ...") the > user appears on the management page. > > If I edit that user via Siremis the line for domain is not > a dropdown but a "normal" text field. > > > > Can somebody explain this? > > > > BR, > > Uwe > > > > P.S.: I'm using Siremis 2.0 and Kamailio 3.2.0. > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > -- > Hugh Waite > Senior Design Engineer > Crocodile RCS Ltd. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

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Siremis: cannot enter domain in "New Subscriber" window
by Uwe Andersen 02 Nov '11

02 Nov '11

Hi, I want to add a new user. So I choose "add" from the "Subscriber Management" page. On the "New Subscriber" page I can enter username, password, email-address, etc. But the line for "domain" seems to be an empty dropdown entry. I cannot choose, add or edit anything. As a result I cannot save the new user because Siremis complains about the empty domain field. After adding a user by command line("kamctl add ...") the user appears on the management page. If I edit that user via Siremis the line for domain is not a dropdown but a "normal" text field. Can somebody explain this? BR, Uwe P.S.: I'm using Siremis 2.0 and Kamailio 3.2.0.

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FW: App_python Module
by Muhammad Nasir Bhutta 02 Nov '11

02 Nov '11

Hi, I am a new user of Kamailio, and I am working on enabling the App_Python module in Kamailio.. While searching on google, I came acroos your email on the forum where you people are trying to patch the App_Python module.. I am wandering whether you can please help a bit .. 1- How to do the configuration to enable the App_python module ? .. At the moment, I am just adding the following line in the configuration file, loadmodule "app_python.so" and just correcting the path to handler.py to the right directory in the code for script path. 2- What output we should expect when we enable the app_python module ? .. I have seen the handler.py script written in python which is printing some sip message information ? .. 3- How the SIP Message structure is being passed to handler.py ? ... How we are accessing the Router module in handler.py script and LM_ERR function from Kamailio ? .. Any help in this regards shall be really helpful .. In any case, whether you can solve this issue or not, please reply the email.. With many thanks.. Kind Regards, Nasir

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sdpops and dynamic codec ids
by Juha Heinanen 01 Nov '11

01 Nov '11

i made call sdp_keep_codecs_by_name("PCMU,PCMA,speex"); on an invite request and it didn't keep speex although it was in the sdp: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): sems 1 1 IN IP4 192.98.102.10 Session Name (s): sems Connection Information (c): IN IP4 192.98.102.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 40000 RTP/AVP 101 102 8 0 3 2 9 96 97 98 99 100 Media Attribute (a): rtpmap:101 iLBC/8000 Media Attribute (a): rtpmap:102 speex/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:2 G721/8000 Media Attribute (a): rtpmap:9 g722/8000 Media Attribute (a): rtpmap:96 telephone-event/8000 Media Attribute (a): rtpmap:97 G726-32/8000 Media Attribute (a): rtpmap:98 G726-24/8000 Media Attribute (a): rtpmap:99 G726-40/8000 Media Attribute (a): rtpmap:100 G726-16/8000 it is so that sdpops module does not support codecs, whose ids are dynamic, but only codecs with static ids? -- juha

2 14

enabling app_python module in Kamailio
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I am trying to enable the app_python module in Kamailio to run a simple python script externally parallel with Kamailio proxy, While I try to load the module "app_python.so" in configuration file, the Kamailio server fails to start.. I can't see what is the problem .. The steps, I am following are that: - Script written in python saved in home directory. - Loadmodule "app_python.so" statement in configuration file. - Modparam("app_python", "script_name", "location of script") - Python_exec("function", "parameters list") .. I have also tried to check the Kamailio configuration option, It gives following warning, 0(2942) WARNING: <core> [sr_module.c:584]: /usr/local/lib64/kamailio/modules/app_python.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead I am wandering if any one can please help in this matter, I shall be very thankful .. With many thanks .. Kind Regards, Nasir Muhammad Nasir Bhutta Solutions Architect Britannic Technologies Ltd t: +44 1483 544812 f: +44 845 0501001 mbhutta(a)btlnet.co.uk<mailto:mbhutta@btlnet.co.uk> www.btlnet.co.uk<http://www.btlnet.co.uk> [cid:image001.jpg@01CC881F.1EA78300]<http://www.linkedin.com/company/britannic-technologies/products>[cid:image002.jpg@01CC881F.1EA78300]<http://www.twitter.com/BritannicTech>[cid:image003.jpg@01CC881F.1EA78300]<http://www.facebook.com/pages/Britannic-Technologies/202957249732198>[cid:image004.jpg@01CC881F.1EA78300]<http://www.youtube.com/user/BritannicTech> ________________________________ [cid:image007.jpg@01CC881F.4A995BF0]<http://www.btlnet.co.uk/> Watch the Telegraph Business Club video about Britannic at our educational TV site - britannictech.tv<http://britannictech.tv> Our 8th Annual Convergence Summit is being held at Silverstone on 17th November 2011. Book your place now<http://www.btlnet.co.uk/news--events/2011/convergence-summit-2011/register> and discover the world of Hosting and Cloud Services. ________________________________ This email and any files transmitted with it are confidential and intended for the addressee only. If you are not the intended recipient please notify the sender and delete this email. The views expressed in this email are personal, and do not constitute a commitment by Britannic Technologies Ltd unless specified by a separate, written agreement. Registered Office: Britannic House, Merrow Business Park, Guildford, Surrey GU4 7WA. Registered in England no. 2090797. ________________________________

3 4

app_python module
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I have successfully installed the app_python module for the default example of handler.py .. Now, I need to see the output of handler.py,.. I have following questions: 1- What is the output of this module ? does it prints the specific message information extracted from SIP message ? 2- How to write modparam and python_exec in configuration file ? I have tried to write modparam("app_python", "mod_init") but, it gives error and says to use mod_register function instead and I don't know more about mod_register function. 3- I have also tried to use python_exec("child_init", 2) , The Kamailio starts successfully.. When, I connect a soft client to Kamailio, then nothing happens ? How to see the output of handler.py script . Any help in this regards shall be really appreciated .. With thanks. Kind Regards, Nasir

1 0

script local flags
by Uri Shacked 01 Nov '11

01 Nov '11

Hi, I am using the script local flags. i set a flag like this "setsflag(6)" in order to use it afterwards. i notice that this flag is relevent only for the request i am dealing with right now. because after the reply the flag seems to reset without me doing that doing "issflagset(6)" and getting nowhere :-) am i right or am i using it wrong?

2 1

FW: App_python Module
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I am a new user of Kamailio, and I am working on enabling the App_Python module in Kamailio.. While searching on google, I came acroos your email on the forum where you people are trying to patch the App_Python module.. I am wandering whether you can please help a bit .. 1- How to do the configuration to enable the App_python module ? .. At the moment, I am just adding the following line in the configuration file, loadmodule "app_python.so" and just correcting the path to handler.py to the right directory in the code for script path. 2- What output we should expect when we enable the app_python module ? .. I have seen the handler.py script written in python which is printing some sip message information ? .. 3- How the SIP Message structure is being passed to handler.py ? ... How we are accessing the Router module in handler.py script and LM_ERR function from Kamailio ? .. Any help in this regards shall be really helpful .. In any case, whether you can solve this issue or not, please reply the email.. With many thanks.. Kind Regards, Muhammad Nasir Bhutta Solutions Architect Britannic Technologies Ltd t: +44 1483 544812 f: +44 845 0501001 mbhutta(a)btlnet.co.uk<mailto:mbhutta@btlnet.co.uk> www.btlnet.co.uk<http://www.btlnet.co.uk> [cid:image001.jpg@01CC9895.3EC504B0]<http://www.linkedin.com/company/britannic-technologies/products>[cid:image002.jpg@01CC9895.3EC504B0]<http://www.twitter.com/BritannicTech>[cid:image003.jpg@01CC9895.3EC504B0]<http://www.facebook.com/pages/Britannic-Technologies/202957249732198>[cid:image004.jpg@01CC9895.3EC504B0]<http://www.youtube.com/user/BritannicTech> ________________________________ [cid:image005.jpg@01CC9895.3EC504B0]<http://www.btlnet.co.uk/> Watch the Telegraph Business Club video about Britannic at our educational TV site - britannictech.tv<http://britannictech.tv> Our 8th Annual Convergence Summit is being held at Silverstone on 17th November 2011. Book your place now<http://www.btlnet.co.uk/news--events/2011/convergence-summit-2011/register> and discover the world of Hosting and Cloud Services. ________________________________ This email and any files transmitted with it are confidential and intended for the addressee only. If you are not the intended recipient please notify the sender and delete this email. The views expressed in this email are personal, and do not constitute a commitment by Britannic Technologies Ltd unless specified by a separate, written agreement. Registered Office: Britannic House, Merrow Business Park, Guildford, Surrey GU4 7WA. Registered in England no. 2090797. ________________________________

1 0

flags and masks in carrierroute
by Uri Shacked 01 Nov '11

01 Nov '11

Hi, I did not understand the explanation of the use of flags and mask. I understand flags are for matching and mask is doing something. But, what exactly does the mask do? How do i see or use it? And regarding the flags... i did "setflag(16)" before the CR function. Set the DB col as needed, and no match was found... Any help here?

1 0

siremis: login error / http error 404
by Uwe Andersen 01 Nov '11

01 Nov '11

Hi Ramona, You hit it! After enabling "rewrite" it is working now. Thank you very much! Uwe > > I have installed Siremis 2.0.0 and Kamaillio 3.2.0 on a > OpenSuse 11.4 system. > > After the Siremis Web Installation Wizard finished the > setup the login failed with a http error 404. > > > > The Apache2.0 error_log shows: > > [Mon Oct 31 11:09:49 2011] [error] [client ::1] File does > not exist: /var/www/siremis-2.0.0/siremis/system, referer: > http://localhost/siremis/index.php/user/login > > > > The Apache2.0 access_log shows: > > ::1 - - [31/Oct/2011:11:09:36 +0100] "GET > /siremis/bin/antispam_image.php?form=user.form.LoginForm&name= > antispam&length=6&level=2?rand=1768032447 HTTP/1.1" 200 240 > "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 > (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > ::1 - - [31/Oct/2011:11:09:48 +0100] "POST > /siremis/bin/controller.php?F=RPCInvoke&P0=[user.form.LoginFor > m]&P1=[Login]&__this=btn_login:onclick&jsrs=1 HTTP/1.1" 200 > 214 "http://localhost/siremis/index.php/user/login" > "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > ::1 - - [31/Oct/2011:11:09:49 +0100] "GET > /siremis/system/general_default HTTP/1.1" 404 1331 > "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 > (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > > > Is there anybody how can help me out of this? > > do you have mod_rewrite enabled in your apache2 server?

1 0

Memory Leak on DB Errors?
by Klaus Darilion 31 Oct '11

31 Oct '11

Hi! I recently had a problem with Kamailio 3.1.4 (provided Debian packages): I had some DB problems (missing tables). Thus, the timer module failed to insert the statistics (for siremis): ERROR: db_mysql [km_dbase.c:120]: driver error on query: Table 'kamailio.statistics_tmx' doesn't exist ERROR: <core> [db_query.c:130]: error while submitting query ERROR: sqlops [sql_api.c:217]: cannot do the query This happened for some time (weeks?), other DB queries were unaffected. Then, suddenly Kamailio ran out of memory: ERROR: <core> [sip_msg_clone.c:506]: ERROR: sip_msg_cloner: cannot allocate memory ERROR: tm [t_lookup.c:1338]: ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It do not think it is a load problem as the server is more or less idle. May it be a memory leak due to wrong error handling? Where there any fixes recently? Thanks Klaus

3 16

current shm memory status
by Klaus Darilion 31 Oct '11

31 Oct '11

Hi! Is there a way to get the status of shm memory? (used memory vs. free memory) Thanks Klaus

2 4

siremis: login error / http error 404
by Uwe Andersen 31 Oct '11

31 Oct '11

Hi, I have installed Siremis 2.0.0 and Kamaillio 3.2.0 on a OpenSuse 11.4 system. After the Siremis Web Installation Wizard finished the setup the login failed with a http error 404. The Apache2.0 error_log shows: [Mon Oct 31 11:09:49 2011] [error] [client ::1] File does not exist: /var/www/siremis-2.0.0/siremis/system, referer: http://localhost/siremis/index.php/user/login The Apache2.0 access_log shows: ::1 - - [31/Oct/2011:11:09:36 +0100] "GET /siremis/bin/antispam_image.php?form=user.form.LoginForm&name=antispam&length=6&level=2?rand=1768032447 HTTP/1.1" 200 240 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" ::1 - - [31/Oct/2011:11:09:48 +0100] "POST /siremis/bin/controller.php?F=RPCInvoke&P0=[user.form.LoginForm]&P1=[Login]&__this=btn_login:onclick&jsrs=1 HTTP/1.1" 200 214 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" ::1 - - [31/Oct/2011:11:09:49 +0100] "GET /siremis/system/general_default HTTP/1.1" 404 1331 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" Is there anybody how can help me out of this? Regards, Uwe Andersen

2 1

Re: [SR-Users] uac module - error searching pvar "uac_req"
by Daniel-Constantin Mierla 31 Oct '11

31 Oct '11

On 10/31/11 2:09 PM, Vladimir Broz wrote: > ah, that was the problem :( ok. Don't forget to keep the mailing list always cc-ed, so people reading the archive for similar issues will find the conclusion, whether the hints were solving or not the problem. Cheers, Daniel > > Thanks Daniel! > -Vlada > > On 10/31/2011 01:37 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> be sure you load uac module from modules_k, not from modules_s. >> >> Cheers, >> Daniel >> >> >> On 10/31/11 12:36 PM, Vladimir Broz wrote: >>> ignore the "sip:b@@test.com" -> same result with the correct (only one >>> "@") value >>> >>> Thanks, >>> -Vlada >>> >>> On 10/31/2011 12:21 PM, Vladimir Broz wrote: >>>> Hi, >>>> >>>> I compiled 3.2 (master) with FLAVOUR=ser and I'm trying to use "uac" >>>> module. >>>> >>>> my config looks like: >>>> ... >>>> $uac_req(method)="NOTIFY"; >>>> $uac_req(ruri)="sip:server.test.com"; >>>> $uac_req(furi)="sip:a@test.com"; >>>> $uac_req(turi)="sip:b@@test.com"; >>>> ... >>>> >>>> and I'm getting this config file error >>>> >>>> 0(26125) ERROR: <core> [pvapi.c:706]: error searching pvar "uac_req" >>>> 0(26125) ERROR: <core> [pvapi.c:910]: wrong char [m/109] in >>>> [$uac_req(method)] at [9 (5)] >>>> 0(26125) : <core> [cfg.y:3501]: parse error in config file >>>> /root/vbroz/ser-srrs-bpi.cfg, line 388, column 2-17: unknown script >>>> pseudo variable $uac_req(method) >>>> ERROR: bad config file (1 errors) >>>> >>>> What's wrong? >>>> >>>> Any help is appreciated... >>>> >>>> Thanks in advance, >>>> -Vlada >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

uac module - error searching pvar "uac_req"
by Vladimir Broz 31 Oct '11

31 Oct '11

Hi, I compiled 3.2 (master) with FLAVOUR=ser and I'm trying to use "uac" module. my config looks like: ... $uac_req(method)="NOTIFY"; $uac_req(ruri)="sip:server.test.com"; $uac_req(furi)="sip:a@test.com"; $uac_req(turi)="sip:b@@test.com"; ... and I'm getting this config file error 0(26125) ERROR: <core> [pvapi.c:706]: error searching pvar "uac_req" 0(26125) ERROR: <core> [pvapi.c:910]: wrong char [m/109] in [$uac_req(method)] at [9 (5)] 0(26125) : <core> [cfg.y:3501]: parse error in config file /root/vbroz/ser-srrs-bpi.cfg, line 388, column 2-17: unknown script pseudo variable $uac_req(method) ERROR: bad config file (1 errors) What's wrong? Any help is appreciated... Thanks in advance, -Vlada

2 2

Re: [SR-Users] how to retrieve the contact username from 3xx reply?
by Uri Shacked 30 Oct '11

30 Oct '11

Thanks! works great.

1 0

how to retrieve the contact username from 3xx reply?
by Uri Shacked 30 Oct '11

30 Oct '11

---------- Forwarded message ---------- From: Uri Shacked <uri.shacked(a)gmail.com> Date: Sun, Oct 30, 2011 at 12:16 PM Subject: how to retrieve the contact username from 3xx reply? To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> I need to store the user i get from the 301 reply to an AVP and use it afterwards in another route. How can i retrieve the contact from the 3xx reply? uac_redirect does not have this option and t_load_contacts does not work for me as well.... Anyone has an idea?

2 1

module docs broken
by Timo Reimann 28 Oct '11

28 Oct '11

Hey, documentation on the dialog module seems to be broken: The master branch directory content is empty http://sip-router.org/docbook/sip-router/branch/master/modules_k/dialog/ 3.1 shows close to nothing http://sip-router.org/docbook/sip-router/branch/3.1/modules_k/dialog/dialog… Moreover, 3.0 gives me 404 on the entire modules_k path: http://sip-router.org/docbook/sip-router/branch/3.0/modules_k/ Help. :) Cheers, --Timo

2 2

T1 timer
by Vitaliy 28 Oct '11

28 Oct '11

Hello, How can i change T1 timeout for transaction created by the UAC module (uac_req_send). Do i have to call t_set_retr before uac_req_send and then t_reset_retr() before t_relay ? I use uac_req_send to send accounting to the remote server and i want to increase retransmit timeout for such signalling messages.

4 5

Fwd: [RAI] SIPit 29 Summary
by Klaus Darilion 28 Oct '11

28 Oct '11

FYI -------- Original Message -------- Subject: [RAI] SIPit 29 Summary Date: Fri, 28 Oct 2011 09:40:51 +0200 From: Robert Sparks <rjsparks(a)nostrum.com> To: rai(a)ietf.org (also available at https://www.sipit.net/SIPit29_summary) SIPit 29 was hosted by Etsi Plugtests in Monte Carlo, Monaco the week of October 24-27, 2011. There were 34 attendees from 17 companies visiting from 12 countries. We had 25 distinct implementations. The roles represented (some implementations act in more than one role) 20 endpoints 5 proxy/registrars The b2buas attending this event were not attempting to appear to be proxies. Implementations using each transport for SIP messages: UDP 92% TCP 96% TLS 88% (24% server-auth-only) SCTP 8% DTLS 0% 44% of the implementations present supported IPv6. There were three RFC4474 Identity implementations present. For DNS we had support for: Full RFC3263 : 76% SRV only : 16% A records only : 08% no DNS support : 0% Support for various items in the endpoints: 75% replaces 50% ice 40% 5389stun 35% turn 25% sip/stun multiplexing 30% gruu 25% history-info (there were no implementations of 4244bis) 20% outbound 20% path 20% diversion 15% 3489stun 10% join 5% service-route 0% sigcomp Support for various items in the proxy/registrars: 100% path 80% sip/stun multiplexing 60% gruu 40% outbound 40% diversion 20% service-route 0% history-info The endpoints and b2buas implemented these methods: 100% INVITE, CANCEL, ACK, BYE 95% REFER 95% NOTIFY 80% REGISTER 80% INFO 75% OPTIONS 75% UPDATE 75% PRACK 60% SUBSCRIBE 35% MESSAGE 30% PUBLISH 100% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). 30% of the implementations required the other party to use symmetric-rtp. 90% of the UAs present both sent RTCP and paid attention to RTCP they received. 80% of the endpoints present supported SRTP using sdes. There were no dtls-srtp implementations at this SIPit. 35% of the endpoints supported multipart/MIME. There were no implementations present with S/MIME support. 36% of the implementations present followed RFC6026 (corrections to the INVITE transaction) 36% followed RFC4320 (corrections to the non-INVITE transaction) Not counting implementations that support events only for REFER: There were 3 SIP Event Server implementations There were 5 SIP Event Client implementations These event packages were supported: Server Client 1 2 presence 1 1 presence.winfo 1 4 message-summary 1 1 dialog 1 1 reg 1 1 conference 1 1 xcap-diff 0 2 kpml These packages were supported with the eventlist extension: Server Client 0 1 presence Two of the proxies present still rely only on max-forwards and one that implemented 3261 loop detection. There were two implementations of fork-loop-fix, but no implementations of max-breadth. Multiparty tests (Notes provided by several test participants, including Eoin McLeod, Trond Andersen, Olle Johansson, and Adrian Georgescu) ** IPv6 Focused Tests The first multiparty session of the event focused specifically on IPv6. We used a basic forking scenario with endpoints registered with several proxies at the bottom of the tree. We had a mix of single and dual-stack endpoints, many supporting video. None of the participant's code would put both v4 and v6 candidates into ICE. Rather, it made assumptions about which candidates to use based on what was happening with the SIP signaling. We ran tests using SRV configurations stating that v6 should be tried first and v4 tried if the v6 failed (and a second configuration that tested v4 first) to ensure that implementations would fail forward and try both address families, with mixed results. We confirmed that one UA configured to use IPv6 only was able to request and use an IPv4 turn allocation and succeeded in making a call between an IPv4 UA and IPv6 ua. Most UAs that supported dual-stack had a configuration to tell the application to ignore any returned AAAAs due to issues encountered in deployments where endpoints autoconfigured IPV6 that didn't actually work. We successfully tested calls going though a mix of v4 and v6 hops (accruing Via and Route/Record-Route headers with addresses in both families. There was an instance of UA registering using sip-outbound (RFC5626) that used IPv4 and IPv6 flows simultaneously. We set up automated tests of using SRV records to indicate preference for IPv4 or IPv6. Two ipv4 only clients proved that they connect to a domain indicating preference for IPv6. ** Forking There are still UAs that do not sensibly handle multiple 200 OKs to a forked INVITE. When testing handling multiple 200 OKs on a simple two branch fork, we encountered an interesting failure. As the proxy handled the 200 OK from branch 1, it sent a CANCEL to the other branch. The UA on branch 2 had already responded with a 200 OK, so it responded to the CANCEL with a 200, but rather than ignoring it, it decided to send a BYE on the dialog. This surprised the calling UA which had chosen the 200 OK from branch 2 to keep, having already sent a BYE to the dialog from branch 1. The better thing for the UA to have done is ignore the CANCEL after replying to it. ** TLS Most of the implementations at the event supported TLS, and all the multiparty tests exercised it. One session focused on proper certificate verification. We tested a certificate at a server with several SAN URI fields, none of which matched the URI used for the server, but a CN that did. Only one UA failed to connect because of this, the rest of the UAs connected happily (or not at all, but because of other issues). Most of the implementations were "before 5922" code, which meant that there was a lot of checking of the Subject/CN. ** Identity There were several SIP Identity implementations present. We tested creating and verifying the Identity assertions at proxies. It wasn't immediately obvious to some implementers that the resource pointed to by the Identity-Info URI had to be DER encoded (a result of the requirement to use application/pkix-cert). Unfortunately, the tar encoded at the end of 4474 has .cer files in it that are PEM encoded. ** STUN/TURN/ICE and Outbound We spent a significant fraction of the multiparty test time (three sessions) focusing on STUN/TURN/ICE and Outbound. There are still clients using RFC 3489 STUN. The participants noted operator pushback against deploying ICE because when things go wrong, it's very hard to diagnose when, where and why they went wrong. If the endpoints always sent a reINVITE (or UPDATE) when concluding ICE, even if they used the default candidates, this diagnosis of failures would be much easier. Examples noted by the participants: * Without a mandatory conclusion, is not possible to know how long it took until end-points had successful media path established * Without a mandatory conclusion, is not always possible to know if different than default candidates have been chosen as lack of INVITE does not tell deterministically what happened * Without a mandatory conclusion, it is not possible for the callee to restart ICE if it does not agree with the conclusion We successfully exercised endpoints setting up and managing multiple outbound flows through separate edge proxies to a proxy/registrar (with different implementations taking turns in the proxy registrar role). We forced the network to break the flow in use in each case, verifying that subsequent dialog-forming requests would take an alternate flow. We discovered an issue with the recommendation in 5626 to have edge proxies record-route that prevents subsequent in-dialog requests from using alternate flows. ** SRTP All implementations supported SDES for SRTP key exchange. Good interop with many different combinations of endpoints. Some calls were left up for an extended period so that sequence number wrapping and roc increments were tested. Once the roc had changed the call was put on hold and resumed with success. In this situation new RTP streams with different SSRCs were created and the sequence number/roc combination re-generated thus proving that support for multiple SRTP contexts exists. Debate surrounding how to signal 'best-effort crypto' continues. Some implementations choose to advertise RTP/AVP with a=crypto attributes and then go crypto if answer also has them and non-crypto if they do not. RTP/SAVP is used to signal the desire for mandatory crypto. Another approach observed was to repeat the same media line twice, once for crypto and once for non-crypto and rely on the receiving end to only accept one of them and port reject the other. In this case both m-lines had the same port number and payload number descriptions so if a receiving end accepted both then the originator would have no clue as to whether it was receiving crypto or non-crypto traffic. ** Unusual messages We tried several kinds of valid, but unusual messages. Very few implementations did the right thing with a request-URI with an unknown scheme - some phones would answer an INVITE with that in the request-URI. ** Specification questions / comments RFC 3263 currently says "If no SRV records are found, the client SHOULD use TCP for a SIPS URI" - it should probably say TLS over TCP. In the spirit of Happy Eyeballs, there should be some guidance on what to do when the RFC 3263 process results in both A and AAAA records and the client is capable of using both. There were implementations of SIP over DTLS, but the specification of that usage seems not to have completed. There were questions on whether internationalized domain names could be used in SIP URIs. For use of SDP with ICE, there was an argument about whether a=rtcp was mandatory if rtcp was on the next port up. RFC 5245 says "If the agent is utilizing RTCP, it MUST encode the RTCP candidate using the a=rtcp attribute as defined in RFC 3605 [RFC3605].", and 3605 says "when that port is not the next higher (odd) port number following the RTP port described in the media line" to motivate why the attribute was created to begin with. Any updates to 5245 should clarify that adding the a=rtcp attribute is not optional when RTCP is being sent on the next higher port. _______________________________________________ RAI mailing list RAI(a)ietf.org https://www.ietf.org/mailman/listinfo/rai

1 0

How to limit maximum number of calls
by Austin Einter 28 Oct '11

28 Oct '11

Hi I am running kamailio proxy (1.5) as an intermediate proxy, where all my SIP signalling packets are passing through. I want to limit maximum number of calls kamailio proxy server can handle at a time. How can I do this, please give me some pointer in this regard. Regards Austin

4 7

new email contact
by Timo Reimann 28 Oct '11

28 Oct '11

Hey all, this is just to let everybody know that my email contact has changed to sr(a)foo-lounge.de . Of course, I'm still on the lists but if anyone aware of my former email address wishes to contact me directly, please refer to the new address in the future. Thanks a lot and cheers, --Timo

1 0

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 27 Oct '11

27 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 *X-Asterisk-HangupCause: Protocol error, unspecified *X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

2 1

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 26 Oct '11

26 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

1 0

Who is at Astricon?
by JR Richardson 26 Oct '11

26 Oct '11

I am. On 10/25/11, sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > 2. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2011 05:44:06 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Dispatcher Confusion (v3.2.0) > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <4EA78206.4060203(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > On 10/25/11 5:52 PM, Asgaroth wrote: >> Hi Daniel, >> >> Thanks for the clarrification, >> >> On 25/10/2011 16:30, Daniel-Constantin Mierla wrote: >>> >>> >>> 4.6. |ds_mark_dst("s")| >>> >>> Mark the last used address from destination set as inactive >>> ("i"/"I"/"0"), active ("a"/"A"/"1") or probing ("p"/"P"/"2"). With >>> this function, an automatic detection of failed gateways can be >>> implemented. When an address is marked as inactive or probing, it >>> will be ignored by 'ds_select_dst' and 'ds_select_domain'. >>> >> >> Above is the part that is a little misleading, it says that >> >> "When an address is marked as inactive or probing, it will be ignored >> by 'ds_select_dst' and 'ds_select_domain'." >> >> This, to me, means that if a gateway is in probing mode >> (Active-Probing) then it wont be selected in the destination set >> because it is in probing mode, if this is not the case then maybe the >> above line should read: >> >> "When an address is marked as inactive or inactive-probing, it will be >> ignored by 'ds_select_dst' and 'ds_select_domain'." >> >> This brings me to my next question then, how would I set the state of >> a destination to Inactive-Probing (state: IP) from the routing script. >> The ds_mark_dst() function only allows one of "a", "i", "p". >> >> If I do a ds_mark_dst("i"), then the state changes to "IX", Inactive >> with no probing to tell when gateway is back up. >> >> If I do a ds_mark_dst("i") and then right after ds_mark_dst("p"), a >> log is printed saying that you cannot put a destination into probing >> state when it is marked as inactive. > are you sure you run the devel version? There is no such case in the > sources -- send me exactly the log message you get. Only when the > destination is disabled the probing cannot be set, but not the same case > of inactive. > >> >> Is it possible to set the state of a gateway to inactive-probing from >> the routing script? > > Yes, it should be, no constraint in master branch source code. > > Daniel > >> >> Thanks >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > >

2 1

sip-router listen address
by caio 25 Oct '11

25 Oct '11

Hello, Is there a problem or anything to consider when sharing the same ip/interface between sip proxy and asterisk? I have asterisk on port 5080 and SR at 5060 in the same interface with the same IP address and have a missing BYE when call is over. However, moving asterisk service to different interface and ip in the same box, the issue is solved. I appreciate your comments. Regards, Claudio

3 4

Kamailio.cfg file woes
by Peter Schrock 24 Oct '11

24 Oct '11

I am wondering two things. 1. is there a place on the web that I may find a basic cfg file to at least get Kamailio running. 2. is there a resource that tells you what to include for what you want to use the system for. At this point, it seems like I should just include everything because I don't know what I might not need. I looked at the references on the website and found the information a little overwhelming with all the options that are available. I think smaller bit sizes would be helpful at this time for me. Peter

3 3

onsend_route and replies
by Asgaroth 24 Oct '11

24 Oct '11

Hi All, Does the onsend route get executed for all messages? I'm just performing some xlog's in the onsend_route, but I'm not seeing it being called for replies. Currently my onsend_route looks like: onsend_route { xlog("onsend_route : $rm : Recieve Details $si:$sp -> $Ri:$Rp"); xlog("onsend_route : $rm : Destination Details $snd(af) $snd(proto):$snd(ip):$snd(port)"); } I can see it being executed for initial messages coming in but I dont see it executed for any replies. Thanks

1 1

kamctl dialplan addrule
by Arthur Alkhasyan 24 Oct '11

24 Oct '11

Hi all, I'm new to the list and new to the Kamailio. We are using Kamailio with Mysql to route calls between clients asterisks and SIP PSTN gateways. We have two inbound gateways and one outbound. I need to add outbound gateway and a few new features: 1. Call routing between carriers based at prefixes added by asterisks. I would like to route all calls with prefix 99101 to gateway 1 and prefix 99102 to gateway 2. Prefix have to be striped out before sending call to carrier. 2. Remote Call Forwarding. I have to setup RCF at Kamailio. All calls to the particular DID number have to be forward to another number. I found description for dialplan module and believe the module can work for me. I recompiled Kamailio with dialplan module support and added to config # ----- dialplan params ----- modparam("dialplan", "db_url", "mysql://openser:openserrw@localhost/openser") I did search and found this thread related to my problem: http://lists.sip-router.org/pipermail/sr-users/2009-November/025416.html I try to use kamctl to add rule to the MySQL and can't find proper syntax kamctl dialplan addrule 10 100 1 ^123+ 0 ^(123.+) 0030\1 -bash: syntax error near unexpected token `(' any help is greatly appreciated. Thank you.

3 3

Re: [SR-Users] module version mismatch for carrierroute
by Uri Shacked 24 Oct '11

24 Oct '11

Hi, i found out the mistake i made... didn't add the carrierroute to the include modules when making cfg file. thanks

1 0

module version mismatch for carrierroute
by Uri Shacked 24 Oct '11

24 Oct '11

Hi, i installed 3.2 from git. when starting kamailio i get the following: "ERROR: module version mismatch for /usr/local/lib/kamailio/modules/carrierroute.so; core: kamailio 3.2.0 (i386/linux); module: kamailio 3.1.4 (i386/linux)" any ideas what to do? thanks, Uri

3 2

Daily tarballs not being created anymore?
by Timo Reimann 24 Oct '11

24 Oct '11

Hi all, something caught my attention today: Apparently, tarballs provided from http://sip-router.org/tarballs/sr/ do not seem to be updated automatically anymore (the latest files are from March) although the Download page states that this should happen: "Tarballs for master (development) and several branches are made available daily (only if there were changes in the sources from previous tarball)" Is there a reason why tarball creation has ceased? If so, I think the Download page information should be corrected. I'd be prefer to have automatic tarballs again, however, as it's easier to point people not involved into development at tarballs instead of a set of git instructions. Thanks and cheers, --Timo

3 2

Kamailio 3.2 compile error on Solaris 10
by Nathaniel L Keeling 22 Oct '11

22 Oct '11

I am trying to compile the new version of Kamailio, 3.2 on Solaris in a 64bit environment and I am getting this error. I have been searching the internet but have not been able to find any solutions. Any help would be appreciated. CC (gcc) [M xmlrpc.so] xmlrpc.o In file included from ../../parser/msg_parser.h:63:0, from ../../sr_module.h:62, from xmlrpc.c:41: ../../parser/../ip_addr.h:86:23: error: field 'sin6' has incomplete type ../../parser/../ip_addr.h: In function 'ip_addr_loopback': ../../parser/../ip_addr.h:283:3: warning: implicit declaration of function 'htonl' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h:286:3: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h: In function 'sockaddr2ip_addr': ../../parser/../ip_addr.h:353:50: error: dereferencing pointer to incomplete type ../../parser/../ip_addr.h: In function 'su_getport': ../../parser/../ip_addr.h:399:4: warning: implicit declaration of function 'ntohs' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h: In function 'su_setport': ../../parser/../ip_addr.h:418:4: warning: implicit declaration of function 'htons' [-Wimplicit-function-declaration] xmlrpc.c: In function 'rpc_scan': xmlrpc.c:1454:6: warning: variable 'fmt_len' set but not used [-Wunused-but-set-variable] gmake[1]: *** [xmlrpc.o] Error 1 gmake: *** [modules] Error 1 Thanks Nathaniel

4 3

Force signalling??
by Skyler 21 Oct '11

21 Oct '11

Hi, I'm trying to setup an inbound proxy (Proxy-1) using dispatcher to relay between 2 other proxies (proxy-2 & 3) depending on DID, but I need to force all signalling through Proxy-1. so #2 & 3 only communicate with #1 and #1 only communicates up and down stream. My trouble is, the INVITES contain the original $ru $fu $tu & $ct info when passing down to Proxy-2 or 3. The result is, only partial signalling is being sent to Proxy-1. After the ACK it seems that Proxy-1 is removed from the communication path entirely as $ru becomes $rU@proxy-2 in the ACK from upstream so Proxy-1 is out of the picture from then on. This means BYE's are not working as Proxy-2 & 3 do not have authentication upstream. I suspect that I need to re-write $ru $fu $tu & $ct info on the original INVITE to reflect Proxy-1 information before dispatching. Is that what I'm missing? If so, how would I restore the original INVITE info to send upstream again on replies etc.? Any help is greatly appreciated. Tia, Skyler

3 6

New developer: Mészáros Mihály
by Daniel-Constantin Mierla 20 Oct '11

20 Oct '11

Hello, just to let everyone know that a new person got today write access to GIT repository: Mészáros Mihály (cc-ed), having the user id misi. Welcome to development team and looking forward to your contributions! Involved in the mailing lists in the past, he submitted the first new module to be part of next major release: presence_profile, implementation of an extension about sip user agent configuration framework, specified in RFC6080: * http://tools.ietf.org/html/rfc6080 The code of the new module is now in GIT branch misi/ua-profile, to be merge soon to master after initial review are done. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Build version in custom server_header
by Asgaroth 20 Oct '11

20 Oct '11

Hi All, Is it possible to have the build/git version added in a custom server_header/useragent_header? For example, my kamailio version is: version: kamailio 3.2.0 (i386/linux) 94d3b8 if I had the following config options: server_header="Server: Proxy" useragent_header="User-Agent: Proxy" Is it possible to do something like: server_header="Server: Proxy ($buildver)" useragent_header="User-Agent: Proxy ($buildver)" Where "$buildver" is the build/git release (94d3b8)? I'm looking at the core cookbook but I dont see anything that jumps out at me, is there a variable that can do this? Thanks

2 2

ACK routing loop
by Antanas Masevicius 20 Oct '11

20 Oct '11

Hello, after troubleshooting ACK routing loops i came up with a thread: http://www.openser.org/pipermail/users/2009-March/004123.html In short, loops formed when i entered IP address to 'domain' table of a host which was remote. Call flow: UAC INVITE -> Kamailio (with remote domain entered) -> UAS (host whose IP matches that remote domain) It looks like after loose_route() check ACK to 200 OK gets routed to local host. Besides this issue, IP address of remote host in 'domain' table did no harm at all. Question is, is that normal ACK routing behaviour or loose_route() should still check if given IP in request is really bound to local interface? Domain module documentation states that "A “local” domain is one that the proxy is responsible for", but id does not tell you that you cannot set remote IP address as a local domain. Please advice. Best regards, Antanas Masevicius

2 4

Dispatcher Confusion Redeux
by Asgaroth 20 Oct '11

20 Oct '11

Hi All, I'd like to clear something up with the diapatcher module and it's probing parameters: First things first: ./kamailio -V version: kamailio 3.1.5 (i386/linux) 76fff5 flags: STATS: Off, USE_IPV6, 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 76fff5 compiled on 16:21:02 Oct 19 2011 with gcc 4.1.2 I think I am suffering from the same confusion as Klaus in the following archive message: http://web.archiveorange.com/archive/v/jZFTG61DlcEr9LJbS8Gq I have the following parameters set for the dispatcher module: modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_probing_mode", 0) When I first start kamailio all gateways are active, no probing occurs. If I then set a gateway inactive (ds_set_state i 1 sip:...), no probing occurs. If I then set same gateway active (ds_set_state a 1 sip:...), no probing occurs. If I mark same gateway as probing (ds_mark_dst("p") after 1 attempt as specified in module parameter), then probing starts. If I then set same gateway inactive (ds_set_state i 1 sip:...) probing still occurs. According to the docs if I set the state of a gateway to inactive then no probing should occur. From my understanding of the docs, if I set ds_probing_mode to 0 then only gateways marked as probing will be "pinged". Can someone with more experienced than myself please clarify how this method is supposed to work. Thanks

3 3

Re: [SR-Users] Kamailio v3.2.0 Released - new major version is
by Javier Gallart 20 Oct '11

20 Oct '11

Hello congratulations to all the people who has contributed! I'm looking forward to try it out, some of the new features look really promising... Regards Javi > ------------------------------ > > Message: 2 > Date: Tue, 18 Oct 2011 20:30:22 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: [SR-Users] Kamailio v3.2.0 Released - new major version is > out > To: kamailio users <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org > Message-ID: <4E9DC5BE.8080003(a)gmail.com> > Content-Type: text/plain; charset=windows-1252; format=flowed > > Kamailio (OpenSER) SIP Server v3.2.0 is out ? a major release with a > very large number of new features and improvements. > > v3.2.0 release is a result of more than 10 months of development and 2 > months of testing from the teams of Kamailio (OpenSER) and SIP Express > Router (SER) projects. In the year of the 10th celebration of our > project (http://sip-router.org/10-years-ser/), this version comes with > 12 brand new modules in addition to a lot of fresh features in core and > old modules. > > Full release notes, including links for download and documentation, are > available at: > > * http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/ > > Many thanks to all developers and community members that made possible > this release. > > Enjoy SIP routing in a more flexible and easier way with Kamailio v3.2.0! > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > > > > >

2 1

Kamailio v3.2.0 RPM packages
by Daniel-Constantin Mierla 20 Oct '11

20 Oct '11

Hello, forgot to announce explicitly that the RPM packages for CentOS, RedHat, Fedora and OpenSUSE were built quickly after the release. They can be downloaded from OpenSUSE build system portal: Links to download packages for each OS are available in the wiki: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

kamctl version
by caio 20 Oct '11

20 Oct '11

Hello, Is kamctl utility version independent of the kamailio binary version? # kamailio -V version: kamailio 3.2.0 (x86_64/linux) 94d3b8 # kamctl version /usr/sbin/kamctl 3.1.0 Just a typo i guess. Regards, Claudio

2 1

kamailio and high-available db backend
by Juha Heinanen 19 Oct '11

19 Oct '11

mino, instead of ndb cluster, you can use drbd together with heartbeat and pacemaker to provide high available mysql/sip server solution. -- juha

1 0

very basic handling for ua-profile event
by MÉSZÁROS Mihály 19 Oct '11

19 Oct '11

Hi, Please find attached a new _very_ basic presence modul to make available user profile provisioning as it stated in RFC6080. Best Regards, Misi

2 3

kamailio and high-available db backend
by Mino Haluz 19 Oct '11

19 Oct '11

Hi, I was using ndb cluster as the dabatase backend for my kamailio setup. The thing is, mysql ndb cluster is difficult to maintain for me and it crashed multiple times (desynchronization, datamemory limit reached, and other strange reasons) and it is not that well documented (the user-base is quite narrow). My question is, what mysql architecture would you recommend me, if it has to be high available, well documented with various examples wihitn the community, thus easy to maintain. DB has 5% writes, 100queries/sec. and its size is 100MB. Mino

1 0

ok
by abdelazize rhosni 19 Oct '11

19 Oct '11

ok

2 1

Kamailio Dispatcher fails in front of FreeSwitch, help (again).
by Henrik Aagaard Sørensen 18 Oct '11

18 Oct '11

I've asked a lot of question about Kamailio Dispatcher in front of FreeSwitch in this mailing list. So I'm sorry that I'm still "spamming", but it still fails. This is my status so far. When connecting 2 devices directly to FreeSwitch everything works. When connecting to Kamailio, some calls fails with "Timeout" and some fails with "Too many loops". This is my kamailio.cfg: http://pastebin.com/KEuZmikh When connected directly to FreeSwitch where everything works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/prkjkb7g When connected to Kamailio where nothing works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/b5x6NZyv 215.197.145.132 (sip.my-domain.com) is my Kamailio. pbx.my-domain.com is my FreeSwitch. 96.224.14.164 is my IP at home, where my 2 devices are from. Can anyone help me out? I'm kinda lost at this point.

3 11

master branch is now open for commits of new features to be in v3.3.0
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Hello, this is a short notice to let everyone know that the GIT master branch is open for commits of new features -- to be part of next major release v3.3.0, as v3.2.0 has been officially released few moments ago. From this moment on, master and 3.2 branches will diverge, 3.2 taking in only bug fixes. So if you want to use stable release of existing features, switch to branch 3.2. You can find guidelines of installing branch 3.2 in the wiki site: * http://www.kamailio.org/wiki/install/3.2.x/git I spotted already a new module by one of the community members submitted to the mailing list in the last days, related to presence extensions, therefore looks like is going to be another great development cycle until 3.3.0 will be out. Happy coding SIP routing extensions in SER and Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio v3.2.0 Released - new major version is out
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Kamailio (OpenSER) SIP Server v3.2.0 is out – a major release with a very large number of new features and improvements. v3.2.0 release is a result of more than 10 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. In the year of the 10th celebration of our project (http://sip-router.org/10-years-ser/), this version comes with 12 brand new modules in addition to a lot of fresh features in core and old modules. Full release notes, including links for download and documentation, are available at: * http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy SIP routing in a more flexible and easier way with Kamailio v3.2.0! -- Daniel-Constantin Mierla -- http://www.asipto.com

1 0

branch 3.2 was created
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Hello, short notice to let everyone know that the branch for 3.2.x release series was created with the name 3.2. For example, downloading sources from GIT and building Kamailio flavour form this branch can be done with: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.2 origin/3.2 make FLAVOUR=kamailio cfg make all make install The official release of 3.2.0 will follow just a bit later today. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

YACC error during Compile of Kamailio 3.1.5
by Nathaniel L Keeling 18 Oct '11

18 Oct '11

I am building Kamailio 3.1.5 on Solaris 10 64bit environment and I am getting this error. yacc -d -b cfg cfg.y "cfg.y", line 104: fatal: invalid escape, or illegal reserved word: expect gmake: *** [cfg.tab.h] Error 1 I tried compiling with bison, but got errors immediately after getting thru the yacc (I just issued ln -s /opt/csw/bin/bison /opt/csw/bin/yacc and recompiled) flex cfg.lex CC (gcc) [kamailio] lex.yy.o In file included from ip_addr.h:50:0, from globals.h:37, from cfg.lex:91: ut.h: In function 'str2s': ut.h:195:17: warning: variable 'init' set but not used [-Wunused-but-set-variable] cfg.lex: In function 'yylex': cfg.lex:561:11: error: 'yylval' undeclared (first use in this function) cfg.lex:561:11: note: each undeclared identifier is reported only once for each function it appears in cfg.lex:561:40: error: 'FORWARD' undeclared (first use in this function) cfg.lex:562:40: error: 'FORWARD_TCP' undeclared (first use in this function) cfg.lex:563:40: error: 'FORWARD_TLS' undeclared (first use in this function) Thanks Nathaniel L Keeling

1 0

Make all error
by Peter Schrock 17 Oct '11

17 Oct '11

I am doing a "make all" and am getting this error: error: mysql/mysql.h: No such file or directory I have mysql installed and know that kamailio is having trouble finding mysql header files. Is there a configuration I can set up to help kamailio find it? Is there a resource I can be directed to find the info? Peter

3 5

preparing to release v3.2.0
by Daniel-Constantin Mierla 17 Oct '11

17 Oct '11

Hello, tomorrow is the day to release the new major version 3.2.0. Since involves creation of a new branch on GIT repository and some adjustments to the packaging, the plan is to start in the morning with the preparations. If any developer has something left to commit, please do it before 9:00am GMT. Afterwards it is very likely that the branch 3.2 will be created and there will be need for backport (which is fine as well). Then by 14:00 GMT the 3.2 branch should be frozen completely to start packaging the release. From than moment on until the release is announced, please send an email to sr-dev in case anyone wants to commit to branch 3.2. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Organization in SIP header
by Skyler 17 Oct '11

17 Oct '11

Hello all, Is there a way to set Organization: within SIP header? I know how to change user_agent_header and server_header but I'd rather not do that. Also I know I could do append_hf, but is there an organization_header? Thanks, Skyler

3 6

Problem with REGISTER retransmissions
by Alejandro Mingo 17 Oct '11

17 Oct '11

Hi, Since we installed last version of Kamailio (3.1.4) we have been experiencing a big problem with REGISTER retransmissions. When the server receives a retransmitted REGISTER it removes the binding and the UAC remains unregistered until next refreshing period. I include a PCAP capture that shows this behaviour. I am not 100% sure that this problem never happened with older versions. Has anybody experienced the same problem? Thanks in advanced. Alejandro.

3 9

Re: [SR-Users] OPENSER MIB
by Daniel-Constantin Mierla 17 Oct '11

17 Oct '11

Hello Stefan, On 3/14/11 11:03 AM, Stefan Tiedje wrote: > Thanks for the answer. > Maybe I have some older versions of the OPENSER-MIB and the other > related MIB's since I could not find the counter you pointed at. I'm > using a MIB browser for reading the MIB's. > Is the suggested counter "expired dialogs" added in a specific release > of Kamailio? Which? We use Kamailio 3.0.2. I used Kamailio and recommend using it sine it has the latest commits for stability. However, what I wrote before is pretty much not related to the version. There is a counter that tracks the processed dialogs, but seems it is not exported by default through snmpstats module. The statistics counter is named "processed_dialogs", implemented by dialog module. You can dump all internal statistics through kamctl or via xmlrpc command, but probably to export it through snmpstats you may need to extend the mibs and the code of the module. I just grepped the sources of snmpstats module to see what dialog statistics it is exporting: $ grep -n _dialogs modules_k/snmpstats/* | grep get_statistic modules_k/snmpstats/alarm_checks.c:83: num_dialogs = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:404: int result = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:424: get_statistic("active_dialogs") - modules_k/snmpstats/snmpObjects.c:425: get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:443: int result = get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:459: int result = get_statistic("failed_dialogs"); modules_k/snmpstats/snmpObjects.c:508: int num_dialogs = get_statistic("active_dialogs"); Perhaps when the snmpstats was developed the dialog module didn't export the statistics counter of "processed_dialogs" and then it was not updated. Now, what I tried to say is that if the "processed_dialogs" counter is not available through snmpstats (and it is not now after grepping the sources) you can get its value from another application through "kamctl get_statistics all" or XMLRPC command for all of the existing kamailio releases. Upcoming one we will look to implement the export through snmpstats as well. If you have time to do it and send us a patch, we will gladly commit it to source tree in our GIT repository. Cheers, Daniel > Do you have the MIB name for the "expired dialogs" counter. I will > look for that in my version of OPENSER MIBS. > Important, do you have a link to where MIB files can be downloaded for > Kamailio 3.0.2? > Below follows an excerp from one of the MIB's. Is it old, I don't know? > > -- *********************************************************************** > > -- OPENSER-MIB: OPENSER MIB > > -- > > -- Date of Creation: Januay 2006 > > -- > > -- This MIB provides information related to the OpenSER SIP Router. > > -- > > -- Copyright (c) The Internet Society (2006) > > -- Ammendments (c) Soma Networks, Inc. (2006) > > -- > > -- All rights reserved. > > -- ***************************************************************** > > /Stefan > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* den 14 mars 2011 10:16 > *To:* Stefan Tiedje > *Cc:* sr-users(a)lists.sip-router.org > *Subject:* Re: [SR-Users] OPENSER MIB > > Hello, > > On 3/14/11 9:42 AM, Stefan Tiedje wrote: >> Hi, >> In the Kamailio OPENSER-MIB there is the counter >> "openserTotalNumFailedDialogSetups". This is a Counter32. >> The description is: >> "The total number of calls that failed with an error. The >> following codes define a failed call:" >> *Question:* >> >> * I'm looking for the corresponding counter to >> "openserTotalNumFailedDialogSetups" who counts successful >> Dialog setups of Counter32 type. Does it exist? >> * If not, does it exist a work around? >> * Where in the code can the new suggested counter be added? >> * Something else???? >> > > the dialog module counts the number of processed dialogs, see: > http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2966360 > > There is no counter currently inside dialog module exporting exactly > the number of successfully setup dialogs, it should not be hard to do > it, though. Using the above and the number of failed and expired > dialogs, you can actually get the number of successful dialogs. > > Dialog module being the one that tracks SIP dialogs, therefore being > able to count them, now I don't know if snmpstats module exports all > the counters from dialog module. I setup snmpstats just few weeks ago > and works perfect on Ubuntu/Debian servers, but I had no need to check > dialog module counters. > > Note that you can get the list of all internal statistics via kamctl: > - kamctl fifo get_statistics all > > Or via XMLRPC if you need them remotely in another application. > > Another option is to define your statistics with statistics module. > Knowing that in SIP a successful call dialog means 200ok reply to an > INVITE transaction, you can count it in the onreply_route[abc] that > you arm for relayed transactions with t_on_reply("abc"). > > Hope these help you, > Daniel > >> Suggestion for the new counter is a name like: >> "openserTotalNumSucceededDialogSetups". It has a counter32. >> Description: "The total number of calls that succeeded" >> I know that there are the counters openserCurNumDialogs, >> openserCurNumDialogsInProgress and openserCurNumDialogsInSetup but >> these are of Gauge type who only reflects the current situation. >> These Gauge counters can't be used together with a Counter32 counter. >> That don't mix. The calculation done for the counter >> "openserCurNumDialogsInProgress" should be used where every new >> dialog setup is added to the new suggested counter. A counter of 32 >> should cover a great deal of connections. These counters are usually >> read, if used, every 15 minutes or 1 hour. >> *Rationale:* >> The reason for the new counter is that a calculation between >> succeeded and failed dialog setups can be done and be used for SLA >> agreements. Without this, its hard to make any customer versus >> provider agreements. >> /Stefan >> PS. Ask if anything is unclear and I need an answer rapidly. >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

2 5

cleaning up contacts...
by Krishna Kurapati 16 Oct '11

16 Oct '11

Hi, I am seeing several Contacts for same AOR in userlocation when I do "kamctl ul show". I suspect, it's a result of UA crashing after registering and then registering with different contact after restart. Since the UA crashed, it could not send Register with expire=0 to remove the previous contact. Since the userlocation API is not exported to configuration scripting, I am wondering how to clean any spurious contacts. Thanks Krish Kura

2 1

FIFO error in cfg file
by Peter Schrock 16 Oct '11

16 Oct '11

I tried opening with "kamctl moni" and I get this error message: ERROR: Error opening Kamailio's FIFO FIFO ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "FIFO")' in your config ERROR: and also have loaded the mi_fifo module. I looked at the cfg file and it looks like what it is asking for is already there. Any suggestions would be welcomed. Peter

2 1

Force socket on reply
by Jijo 14 Oct '11

14 Oct '11

Hi All, Our system has two IP address, one is used for kamailio and 2nd one is used for data. The problem is suppose a request came in to kamailio on a TCP connection with first IP address and connection is torned down before sending the response. Later when the response is send out kamialio is using the 2nd IP to create the connection. The reason is kamailio doesn't have force socket in in reply, so it uses INADDR_ANY for soruce addres .so kernel uses source address based on the detaintion IP. I tired to force the socket using pseudo variable $fs or force_send_socket(), but neither of them worked from reply_route. So i modfied the code relay_replay to set the SND_F_FORCE_SOCKET which will use the address where request is recieved as source address to make the TCP connection The code change is shown below. Please let me know is there anyother way we could acheive it? }else{ buf = build_res_buf_from_sip_res( relayed_msg, &res_len ); /* if we build a message from shmem, we need to remove via delete lumps which are now stirred in the shmem-ed structure */ if (branch!=relay) { free_via_clen_lump(&relayed_msg->add_rm); } /* update send_flags with possible additions from the reply route */ SND_FLAGS_OR(&uas_rb->dst.send_flags, &uas_rb->dst.send_flags, &relayed_msg->rpl_send_flags); /* Make the response to use the same IP where it receives the message */ * * * uas_rb->dst.send_flags.f |= SND_F_FORCE_SOCKET; * } Thanks JIjo

2 4

Cannot concect to iptel.org
by Enda Mannion 14 Oct '11

14 Oct '11

Hi, I am trying to connect to the iptel.org SIP server using the recommended jitsi SIP UA. Is the service still up and running? I am using my newly created username and password but I get no response back from the REGISTER I send. Enda

1 0

a=nortpproxy:yes
by Austin Einter 13 Oct '11

13 Oct '11

If the SIP UA is not behind the NAT, and Kamailio insersts a=nortpproxy:yes in SDP body, can it cause one sided audio. I am using PJSIP as my UA, kamailio 3.1.5 , rtp proxy 1.2.1 as intermediate proxy for signalling and media. In kamailio config, I have enabled WITH_NAT. On call setup, only one way audio is there. When I looked at SIP messages, the only additional stuff I can see a=nortpproxy and nat=yes in recpord route. Without kamailio as intermediate proxy, audio is good both way. So I feel, when I bring up kamailio as intermediate proxy I am doing some mistake configuration wise either at PJSIP UA or in Kamailio. Has anybody faced this issue previously. If so please share how can we fix this issue. Thanks in advance Austin

2 1

Tls protocol testing with iptel account
by Lello 13 Oct '11

13 Oct '11

Hi, I'm working with a softphone project named mjSip, now I've implemented the TLS protocol in the softphone and I would try it with 2 iptel account setted on two pc, on the first I have: # MjSip UA configuration file # ___________________________________________ # host_port=5061 transport_protocols= tls outbound_proxy=sip.iptel.org from_url=<sip:alexlel@iptel.org> username=alexlel realm=iptel.org passwd=2387lelloh debug_level=8 log_path=log max_logsize=2000 do_register=yes keepalive_time=8000 _______________________________ the second is such as this. However I have no response in SSL/TLS by the iptel server. Does it work in TLS? Thanks, Regards.

2 1

Employment you've been searching!
by apache＠iptel.org 13 Oct '11

13 Oct '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I would like to take this time to welcome you to our hiring process and give you a brief synopsis of the position's benefits and requirements. If you are taking a career break, are on a maternity leave, recently retired or simply looking for some part-time job, this position is for you. [...] Content analysis details: (12.2 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [188.49.28.2 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: it-jobsearch.com] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: it-jobsearch.com] 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?188.49.28.2>] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [188.49.28.2 listed in bb.barracudacentral.org] 1.5 BAYES_60 BODY: Bayes spam probability is 60 to 80% [score: 0.6541] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Can't create a new account with kamctl
by Peter Schrock 12 Oct '11

12 Oct '11

I have finally got to the section of the install section "10. Ready to Rock". Unfortunately, I did get the error message "ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN". I did the two steps that the instructions gave and cannot find ".kamctlrc" to "add SIP_DOMAIN=mysipserver.com" Please help. I am not quite understanding what should be done to fix this Peter

2 1

Kamailio Developer Seminar, Oct 24-24, San Francisco
by Daniel-Constantin Mierla 12 Oct '11

12 Oct '11

Hello, the place and date for Kamailio/SER Developer Seminar have been set: San Mateo (just south of San Francisco, CA, in the nearby of international airport), for one day and a half, Oct 24-25, 2011. I posted more details, including about content and registration, at: * http://www.kamailio.org/w/2011/10/kamailio-developer-seminar-san-francisco-… Participating fee is 250USD. Selection of attendees is done by the rule of first come first served. The number of seats is limited, don't wait too long if you want to attend. Looking forward to meeting many of you there. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Can anyone help for a small amount of money.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

Hi everyone. I'm still facing problems with my Kamailio Dispatcher <-> FreeSWITCH setup. Everything works when connecting directly to FreeSWITCH. When connecting to Kamailio, which should work as a dispatcher (load-balancer) in front of FreeSWITCH I get "Too many hops", "Timeouts" etc. My test setup with SIP clients is an iPhone, an Android, an iPad and a laptops with Ubuntu. Can someone help me out, for a small amount of money, until I can get it to work? Hopefully it's just the kamailio.cfg which needs to do things correctly. Currently I'm using this http://pastebin.com/mahKECAw with help from Morten in this mailing list.

1 0

Creating a database error
by Peter Schrock 12 Oct '11

12 Oct '11

This is what I am getting: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating core database and grant privileges failed! I looked at the files and cannot find the problem. Please help. Peter

3 4

Kamailio Dispatcher and FreeSwitch, Too many hops.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc. After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch. I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other. Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.

6 21

Re: [SR-Users] Problem with Route containing two elements
by davy van de moere 12 Oct '11

12 Oct '11

Owki, reading RFC-3261 again ;) Imho Kamailio does what it is supposed to do, it forwards the packet to the first element in the Route header. But as I received the BYE packet from <sip:their_sbc> , that one should have stripped itself out of the Route header, making everything just work, right? Now I can hackishly fix it with a rewritehostport in the middle. But that will eventually not scale up that nicely. Ergo, does anyone have a good idea on how to e.g. have Kamailio in this situation ignore the first uri in the route header? Or am I still not getting it :) ? thx ! BYE sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060;transport=udp SIP/2.0 Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP their_sbc;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789 at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>;tag=as5716fec8 To: <sip:05xxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6 at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060;transport=udp X-Enswitch-Source: their_internals:5060

2 2

proxy BYE to itself
by caio 12 Oct '11

12 Oct '11

Hello, I have the following issue. A calls B, talk for a while, then B sends a BYE, but A never receives it. The BYE is transmited from the proxy listening interface:port to itself, but not to the asterisk port. U 2011/10/08 00:29:29.207335 Kamailio_IP:5060 -> Kamailio_IP:5060 BYE sip:Kamailio_IP;lr=on SIP/2.0. The scenario is: A (pstn subs) --> Asterisk:5080 --> Kamailio:5060 --> B (registered sip ua) Asterisk and Kamailio are in the same box, listening both at the same IP (eth iface) address, but one at port 5080, and the proxy at 5060 (udp). The configuration is based on kamailio.cfg file provided by the v3.1.5 (with a few modules added, like dialplan and cr..etc). If A is who send the BYE, everything goes well. I attach the full sip capture. Thanks for you time. Claudio

1 1

Problem with Route containing two elements
by davy van de moere 11 Oct '11

11 Oct '11

I'm trying to integrate to an integrics enswitch, almost everything works like a charm, except on BYE packets as Kamailio in my setup forwards these incorrectly. Digging somewhat I believe the culprit sits in the Route header which comes from the enswitch: Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio takes into account the first argument (the thing sip:their_sbc...) on which it cannot find the correct transaction. And forwards it back to the sender, instead of forwarding it correctly to my freeswitch. which gives a storm of traffic ;) If I compare that with an ACK packet in the same transaction , having this as a Route header : Route: <sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio sends it along correctly. IMHO if I could convince Kamailio to always take the last part of the Route header into account and ignore the first one it would correctly work. What would be a good approach? Rewriting the route header looks abit harsch. disclaimer : I'm not sure which component is disrespecting the RFC, if any, or if I just fully not get something :) Short overview + packet detail (with some obfuscation): My setup: their_sbc --- my_kamailio --- freeswitch --- my_kamailio --- enduser BYE sip:05xxxxxx@my_kamailio:5060;transport=udp SIP/2.0 Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP their_sbc;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060 BYE sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio;lr=on;ftag=as5716fec8 SIP/2.0 Record-Route: <sip:05xxxxxx@my_kamailio;lr=on;ftag=as5716fec8> Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP my_kamailio;branch=z9hG4bK1ae8.b4f95af2.0 Via: SIP/2.0/UDP my_kamailio;rport=5060;branch=z9hG4bK1ae8.a4f95af2.0 Via: SIP/2.0/UDP their_sbc;rport=5060;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Max-Forwards: 67 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060

2 1

[OT] SIP on the Web
by Iñaki Baz Castillo 11 Oct '11

11 Oct '11

Hi folks, I would like to show a project my colleage José Luis Millán and me are working on. It's about SIP protocol running in a web browser. When RTCweb (media capable web browsers) becomes a reality both technologies together will allow real SIP endpoints coded in JavaScript running in a web browser and interoperating with real SIP networks. Bye bye Flash and custom protocol gateways. http://sip-on-the-web.aliax.net/ Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 10

limiting concurrent calls with dialog module
by Uri Shacked 11 Oct '11

11 Oct '11

Hi, I am trying to create concurrent calls limitations. two questions: 1. should i use dialog module? or is there a better module? 2. i was trying to work with the dialog module, but didnt understand how and where to configure the commands and which to use.... any examples or ideas? BR, Uri

10 17

releasing v3.2.0
by Daniel-Constantin Mierla 11 Oct '11

11 Oct '11

Hello, it is time to set the time to release v3.2.0. From my point of view, testing went very well, considering also that core and main modules were not touched much during this devel cycle, the most of the efforts were on new modules and application service components such as presence server, Lua ... A convenient date for me will be Tuesday, October 18, 2011, allowing to work a bit on packaging and docs. If there are other preferences, jump in this discussion. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 8

(no subject)
by alan chiu 10 Oct '11

10 Oct '11

http://toehugger.com/CRE/info.php?html143

1 0

Audio quality issue
by Austin Einter 10 Oct '11

10 Oct '11

Hi I am using Kamailio 3.1.5 . I am using RTP proxy also. I have used default kamailio.cfg.sample fiile , and just added line #!define WITH_NAT. I have another Main proxy. I wanted all my signalling and media packets should just pass through machine where Kamailio and RTP proxy are running. With this I found, call is established, all signalling and media packets are passing through kamailio / rtp-proxy. So far so good. One way audio stream (from called party to calling party) quality is good. The other audio stream (from calling party to called party is very bad. Did anybody face this issue? Please help me to sort out this issue audio quality issue. Regards Austin

4 5

Carrierroute: Could not allocate shared memory
by Mino Haluz 10 Oct '11

10 Oct '11

Hi, is there any setting which could allow me to set maximum memory per module ? As I am testing the carrier route module, I've added for testing purposes 100 000 rules sofar. When I start the kamailio, it gives me : Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute [../../ut.h:702]: could not allocate shared memory from available pool Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute [cr_rule.c:155]: could not allocate shared memory from available pool Thanks.

3 4

Memory Allocation in Kamailio
by Austin Einter 08 Oct '11

08 Oct '11

Hi Just a quick question. How memory is allocated in Kamailio code. Few places I saw pkg_malloc function. Is it the only function used to allocate memory or there are similar functions present in kamailio. Thanks Austin

3 2

tm and msilo module
by Mark Anthony C. Delfin 08 Oct '11

08 Oct '11

Dear List, I am trying to enable msilo on Kamailio but I'm getting an error ct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Oct 8 11:30:30 localhost kamailio[13856]: INFO: sl [sl.c:163]: could not bind tm module - only stateless mode available Oct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Oct 8 11:30:30 localhost kamailio[13856]: ERROR: msilo [msilo.c:333]: can't load TM API Oct 8 11:30:30 localhost kamailio[13856]: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module msilo (/usr/local/lib64/kamailio/modules_k/msilo.so) Any ideas how can I fixed it. Thank you in Advance. Regards, Mark Anthony C. Delfin

2 1

Kamailio RTP Proxy Questions
by Austin Einter 07 Oct '11

07 Oct '11

Hi I downloaded 3.1.5 Kamalio source, did make and install. Tried to run, and tested few basic scenarios, everything working fine. I have a main proxy, want to use kamailio as intermediate proxy and my requirement is all RTP packets should pass through machine in which Kamalio proxy is running. I beleive RTP Proxy I can run in same machine where Kamailio proxy is running with proper configuration . For this I am referring, http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… page. In one of the steps it says "So, if we are using the Rtpproxy server with default configuration, we have to open /etc/default/rtpproxy file and uncomment following line regarding of udp socket, that will be sued for interconnection:". I tried to search rtpproxy configuration file, could not locate. While making and installing Kamalio proxy, the commands I used "make prefix=/usr/local/ all and make prefix=/usr/local/ install". Can somebody point me why rtpproxy config file is missing. Also the link I am refering to is correct or wrong one. Best Regards, Austin

2 6

Multiple INVITEs and discarded call_control
by Mino Haluz 07 Oct '11

07 Oct '11

Hi, my kamailio server is receiving from some customers 3 identical INVITEs when call is initiated (separated by 200ms). Those 3 INVITEs are making a big problem with call_control: WARNING: call_control [call_control.c:1156]: dialog to trace controlled call was not created. discarding callcontrol. That is why, the prepaid limit is not working at all in this case. This way the user can hack the prepaid protection of the account. Otherwise the call_control is fuilly functional. Anybody experienced the similar problem? If so, how to resolve it? Thanks, Mino Haluz

2 4

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