- sr-users - kamailio.org

How to unset_dlg_profile in request_route or react more quickly to CANCEL?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi List Testing failure situations, I discovered unset_dlg_profile can not be used in request_route: I count the channels per customer in a dlg_profile to know when they are busy. Residential POTS customer have 1 channel. Now this situation (Trying to mimik POTS behavior) Kamailio <=> CPE of 'John-Doe' => INVITE set_dlg_profile of John-Doe is 0: Not busy set_dlg_profile of John-Doe +1 <= 100 trying <= 180 ringing == LINK to CPE DISRUPTED == X<= 200 OK (not reaching kamailio) CPE is sending 200 a couple of times and fails. Caller is still hearing ringing tone as it never got 200 OK. The caller does not want to wait longer and hangs up =>X CANCEL (NOT reaching CPE) This is the moment, on which I would like to unset_dlg_profile of John-Doe to mark his channel available again. But the CPE is never going to acknowledge this CANCEL with 487. Unfortunately: unset_dlg_profile(): dialog delete profile cannot be used in request route Only when kamailio has sent CANCEL a couple of times and the failure_route for this CANCEL is triggered, I can remove the call from the dlg_profile of that customer within failure_route. If I am unlucky, in the meantime other calls get 'busy' instead of being re-routed to the configured backup number of that customer. Agreed, after a couple of seconds, the call goes to the failure route and the dialogue is ending thus calls work as expected. But wouldn't it be nice to be able to unset the dlg_profile upon receiving the CANCEL instead on when the CANCEL transaction is successful? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

DNS Host-Names in Via and Record-Route for more redundancy?
by Benoit Panizzon 20 Dec '22

20 Dec '22

Hi Folks We use two 'core' instances and two 'registrar' instances. The pairs sync their information (Dialogues + Locations) via DMQ. But I noticed that they record the Via and Record-Routes by IP addresses. So I guess if one fails, signaling would not automatically switch to the remaining host as the IP of the failed one is in the header, right? So, is there a way to put a DNS resolvable host-name in the Via and Record-Route header, so that DNS round-robin would make sure requests reach the available server if one of the pairs fails? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 6

Multiple websocket connections with the same port
by Benjamín Visón 19 Dec '22

19 Dec '22

Hello! I'm examining the WebSocket module and I am wondering how it handles multiple connections from the same port. For context, imagine the following scenario: UA1 ==> Load Balancer:41914 ==> Kamailio (WS) UA2 ==> Load Balancer:41914 ==> Kamailio (WS) Now, imagine Kamailio perceives UA1 from port 41914. What happens if that UA1 suddenly drops and another UA2 comes in from the same port? (before the connection closed event is fired for UA1) I understand by looking at the source code that `wsconn_add` is called and that this new connection is added to the list. But what happens with the previous UA entry? does it gets overwritten or does it stays there? The reason for my question is: I am perceiving the above scenario like this: Let's say UA1 is *sip:d3315cf8@9a431b1f90e5.invalid;transport=ws SIP/2.0* and UA2 is *sip:asd08a08@555343439666.invalid;transport=ws. *If UA2 comes in and takes over the port that UA1 was using, and I send an INVITE to UA1 UA2 ends up receiving the invite (obviously because it now has that port allocated). So what I am thinking is that both *d3315cf8(a)9a431b1f90e5.invalid *and *asd08a08(a)555343439666.invalid *are mapped to the same port for a short period of time. Thoughts? Saludos, [image: Twitter] <https://twitter.com/benjaminvison>[image: Instagram] <https://instagram.com/bvisonl/> Benjamín Visón / IT Engineer / Software Developer bvisonl(a)gmail.com / (829)-664-5163 https://www.linkedin.com/in/bvisonl/

3 2

auth_challenge() and async
by Alex Balashov 19 Dec '22

19 Dec '22

Hi, Is there any good solution for the scenario of an auth_challenge() with two separate transaction suspensions? To clarify (simplified): request_route { ... t_newtran(); if(!is_present_hf("Authorization") && !is_present_hf("Proxy-Authorization")) { auth_challenge("$fd", "1"); exit; } # TM suspend/continue t_newtran(); async credentials query("RESUME_AUTH"); } route[RESUME_AUTH] { # Credentials received into PVs. if(!pv_auth_check("...")) { auth_challenge("$fd", "1"); exit; } # Create transaction shell if not exists already from auth query. if(!t_lookup_request()) t_newtran(); # TM suspend/continue async route query("RESUME_ROUTING"); } route[RESUME_ROUTING] { # Unmarshal etc. t_relay(); # etc. } What happens here is that the first auth_challenge() results in a retransmission of its 407 challenge without absorbing the negative ACK. This is despite the ostensibly stateful behaviour of auth_challenge() without "force_stateless_reply" enabled. In this case, it eclipses the second, unrelated 407 challenge from the subsequent routing query (407 challenge with +1 CSeq): -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 17

Dealing with failed SRV peers
by Sebastian Damm 17 Dec '22

17 Dec '22

Hi, we have some Kamailios working as outboundproxy. So they get requests from internal systems and send them to different providers. From time to time, one provider returns a server as primary resource which is currently unavailable. I guess if the internal systems connected directly to the target, they would remember the failed server and remember to always use the server with second priority at least until DNS refresh time. In our setup, since every request is "new" for Kamailio, it doesn't remember, which host is reachable or not. Example: target: example.com _sip._udp.example.com SRV resolves to: 10 192.0.2.42 5060 20 198.51.100.42 5060 30 203.0.113.42 5060 192.0.2.42 is unavailable. Still, Kamailio uses it for every new request and failover to 198.51.100.42 occurs only after timeouts hit. Is there a best practice for solving this? I have played around with the dst_blocklist settings, but that caused even more trouble because Kamailio started blocking requests to peers that have only one server in the record having a short hickup. Thanks in advance for every input, as this is causing trouble every time we run into such a situation. Regards, Sebastian

4 7

Redundancy
by Jawaid Bazyar 16 Dec '22

16 Dec '22

Hi, I am exploring different redundancy / load-balancing models for a Kamailio cluster. When I say cluster, I mean, a number (N) of Kamailio nodes acting as stateful proxies. Each node is configured the same as the others, and all have access to the same lookup data to make routing decisions. I would appreciate any advice or experience any of you can share on these different models. Overall model: Direct to proxies Redirect servers first, which redirect to proxies Selecting the first node to talk to. Each model could use either type of selection. DNS-based (SRV or NAPTR, client makes call to dns name) Anycast with ECMP (equal-cost multi-path routing) Cluster with a mobile IP and service-down detection (this would just provide 1:1 protection) Have clients make calls through the proxy using a DNS record containing an SRV record for each node (or, alternatively, done with NAPTR). Would rely on the client to switch nodes in the event of a node failure mid-call. (Is that even possible?) Anycast would only work with UDP signaling. Use Anycast to find the first proxy, then use stateful responses to direct client back to same node for subsequent messages in a dialog. So for anyone who has tried any of these methods, I would love to hear the pros and cons.. Thanks in advance! Jawaid

2 4

Must openssl_mutex_shared.so be preloaded?
by Дилян Палаузов 16 Dec '22

16 Dec '22

Hello, The documentation on using openssl_mutex_shared.so is contradicting. https://github.com/kamailio/kamailio/blob/master/src/modules/tls/README#L293 says: > For OpenSSL (libssl) v1.1.x, it is required to preload 'openssl_mutex_shared' library shipped by Kamailio. For more details see 'src/modules/tls/openssl_mutex_shared/README.md'. https://github.com/kamailio/kamailio/blob/master/src/modules/tls/utils/open… says: > IMPORTANT: the workaround of using this preloaded shared library is no longer needed starting with Kamailio v5.3.0-pre1 (git master branch after September 14, 2019). The code of this shared library has been included in the core of Kamailio and the same behaviour is now achieved by default. It is unclear, if openssl_mutex_shared.so must be preloaded. In case the library is not needed anymore, it shall not be compiled and installed. Kind regards Дилян

1 1

Logs suddenly inundated with qm_malloc errors
by George Diamantopoulos 16 Dec '22

16 Dec '22

Hello all, I'm testing out release 5.6.2 on one production kamailio instance, and today our monitoring system triggered an alarm for lack of disk space on /. Indeed, kamailio.log started growing at an alarming rate at some point. At about the same time disk space started being depleted, the following line was frequently logged in the logs (no instances of it before that time): qm_malloc(): qm_malloc(0x7f5ade9c0010, 536) called from rtpengine: bencode.c: __bencode_piece_new(79), module: rtpengine; Free fragment not found! They don't only concern rtpengine, core has produced similar "free fragment not found" errors as well. There's additional multiple "had header field", "bad header" and "invalid header name" errors appearing after that too, but I'm guessing that's only a symptom of the qm_malloc issue? kamailio is started with -m 2048 -M 16 and there's another instance with almost identical traffic and configuration, but running on 5.6.1, which has never exhibited such failures in several months uptime. So I'm wondering, is there anything in 5.6.2 that could explain this? Thanks! Best regards, George

3 4

NAT issues with Kamailio + Asterisk with 2 NICs
by Sergio Charrua 15 Dec '22

15 Dec '22

Hi all! Got a random issue with NAT which I really have no clue how to solve. Setup is as follows: - Client A sends SIP INVITE through Public IP 1.2.3.4, where a Cisco Firewall forwards NATed data to a Kamailio instance (acting as a SIPProxy with dispatcher) listening with internal address 192.168.0.1 on port 5060. RTPEngine is in the same server and address too. - Kamailio will forward the invite to Asterisk NIC 1 (192.168.0.2) which will do some provisioning and accounting stuff and redirect the call (if authorized) to Client B's public IP address (all clients are outside the network, reachable via WWW) using NIC 2 (192.168.1.2) and public address 4.3.2.1. The issue is that from time to time, calls will not have audio. Here is what I have found: - when Client A sends INVITE, Asterisk NIC 1 will reply SIP 183 with SDP IP Address set to the internal IP of NIC 1 (192.168.0.2) - when Asterisk forwards call to Client B public IP address the SDP IP address is 192.168.1.2 (NIC 2) And I think the issue is the SDP address is wrong: when Asterisk forwards call to Client B, the SDP address should be the public IP 4.3.2.1 but it sends 192.168.1.2. And when Asterisk replies to Kamailio the SDP is 192.168.0.1. Correct me if I am wrong but i think the solution should be: - when Kamailio receives a reply from Asterisk to be forwarded to Client A, it should modify the SDP address to public IP 1.2.3.4 - when Asterisk forwards call to Client B, the sip.conf should have the parameter externip set to the public IP 4.3.2.1 Questions: - are my analysis & solution correct? - is there an example of the above setup where I could get some ideas? - how do I set Kamailio to modify the SDP address for me? Any examples? Thanks in advance. *Sérgio Charrua*

2 1

kamailio.org server maintenance - Dec 14, 2022 (tomorrow)
by Daniel-Constantin Mierla 15 Dec '22

15 Dec '22

Hello, not that tomorrow, Dec 14, 2022 starting after 09:00UTC (10:00Berlin), is planned some maintenance work on kamailio.org server. It will be unavailable from time to time, hopefully not for long time. Affected services: - web site - mailing lists - wiki site (alternative use the github repo: https://github.com/kamailio/kamailio-wiki) - html version of modules' documentation (text variant README is every module folder of git clones or on github.org repository) Short notifications will be sent when it starts and ends. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

Failover - how does it work?
by Jawaid Bazyar 15 Dec '22

15 Dec '22

I have the following basic script: request_route{ record_route(); $var(enumres) = enum_query(); switch( $var(enumres) ) { case -1: xlog("enum_query lookup failure - To URI $tU"); sl_reply("503", "Temporary lookup failure"); return(0); case 1: break; # uri replaced } xlog("INVITE ENUM query - To URI $tU"); t_relay(); I have the following ENUM records for a test setup: ;; ANSWER SECTION: x.x.x.1.5.1.8.3.0.3.1.e164.arpa. 60 IN NAPTR 100 10 "U" "E2U+sip" "!^.*$!sip:+13038151xxx@10.20.20.198!" . x.x.x.1.5.1.8.3.0.3.1.e164.arpa. 60 IN NAPTR 110 10 "U" "E2U+sip" "!^.*$!sip:+13038151xxx@10.20.20.75!" . What happens is the following: tcpdump: listening on ens3, link-type EN10MB (Ethernet), capture size 262144 bytes 17:57:19.961041 IP (tos 0x10, ttl 64, id 59648, offset 0, flags [none], proto UDP (17), length 1481) 10.20.20.40.5060 > 10.20.20.198.5060: SIP, length: 1453 INVITE sip:+13038151xxx@10.20.20.198 SIP/2.0 … 17:57:19.961058 IP (tos 0x10, ttl 64, id 24143, offset 0, flags [none], proto UDP (17), length 1480) 10.20.20.40.5060 > 10.20.20.75.5060: SIP, length: 1452 INVITE sip:+13038151xxx@10.20.20.75 SIP/2.0 … These basically go out at the same time. Eventually I get a ICMP Port Unreachable on the .198 one because that’s a dummy I just set up for testing. My question is, with this kind of setup, is there a way to tell Kamailio to respect the NAPTR order value (note, 100, 110 above) and only try the 2nd route if the first fails? Thanks, Jawaid

2 5

testing mailing list
by Daniel-Constantin Mierla 14 Dec '22

14 Dec '22

after mailing list service upgrade - ignore!! -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

None
by None 14 Dec '22

14 Dec '22

ACK messages SER keeps sending the 487 Request cancelled message as if the ACK didn't match. Any suggestion of what it can be? I'm using as the endpoint an ATA186, v2.15, and ser version 0.8.11. Best Regards, Alejandro _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ------=_NextPart_000_0009_01C322CD.41C9F360 Content-Type: text/plain; name="remote.ata.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="remote.ata.txt" interface: eth0 (200.75.73.0/255.255.255.0) filter: ip and ( port 5060 ) # U 200.75.73.22:5060 -> 200.75.91.137:5060 INVITE sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..M ax-Forwards: 10..Record-Route: <sip:2001@200.75.73.22;branch=0>..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171e2f6.0..Via: SIP/2.0/U DP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=9 28411572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 329755117 1@192.168.0.15..CSeq: 1 INVITE..Contact: <sip:1010@200.75.73.22:5061;u ser=phone;transport=udp>..User-Agent: Cisco ATA 186 v2.15 ata18x (030 122a)..Expires: 300..Content-Length: 245..Content-Type: application/sd p....v=0..o=1010 8066 8066 IN IP4 192.168.0.15..s=ATA186 Call..c=IN IP 4 200.75.73.22..t=0 0..m=audio 20002 RTP/AVP 4 8 0 101..a=rtpmap:4 G72 3/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c 171e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobr eip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=p hone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE ..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258. c171e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsob reip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user= phone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq: 1 INVIT E..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 CANCEL sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..M ax-Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171 e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip .biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phon e>..Call-ID: 3297551171@192.168.0.15..CSeq: 1 CANCEL..User-Agent: Cisc o ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171e 2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip. biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone >;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq: 1 CANCEL..Se rver: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.91.137:5060 -> 200.75.73.22:5060 SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP 200.75.73.22;branch=z9 hG4bK6258.c171e2f6.0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1 010(a)vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip .biz;user=phone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSe q: 1 INVITE..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Le ngth: 0.... # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Via: SIP/2.0/UDP 200.75.73.22;branch=z9hG4bK6258.c171e2f6.0..From: <sip:10 10(a)vozsobreip.biz;user=phone>;tag=928411572..Call-ID: 3297551171(a)192.1 68.0.15..To: <sip:2001@vozsobreip.biz;user=phone>;tag=3742565922..CSeq : 1 ACK..User-Agent: Sip EXpress router(0.8.10 (i386/linux))..Content- Length: 0.... # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . # U 200.75.73.22:5060 -> 200.75.91.137:5060 ACK sip:2001@200.75.91.137:5060;user=phone;transport=udp SIP/2.0..Max- Forwards: 10..Via: SIP/2.0/UDP 200.75.73.22;branch=0..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928 411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360 b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..U ser-Agent: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0... . exit ------=_NextPart_000_0009_01C322CD.41C9F360 Content-Type: text/plain; name="local.ata.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="local.ata.txt" interface: eth1 (192.168.0.0/255.255.255.0) filter: ip and ( port 5060 ) # U 192.168.0.15:5061 -> 200.75.73.22:5060 INVITE sip:2001@vozsobreip.biz;user=phone SIP/2.0..Via: SIP/2.0/UDP 19 2.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411 572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 3297551171@192 .168.0.15..CSeq: 1 INVITE..Contact: <sip:1010@200.75.73.22:5061;user=p hone;transport=udp>..User-Agent: Cisco ATA 186 v2.15 ata18x (030122a) ..Expires: 300..Content-Length: 245..Content-Type: application/sdp.... v=0..o=1010 8066 8066 IN IP4 192.168.0.15..s=ATA186 Call..c=IN IP4 200 .75.73.22..t=0 0..m=audio 20002 RTP/AVP 4 8 0 101..a=rtpmap:4 G723/800 0/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 tele phone-event/8000..a=fmtp:101 0-15.. # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 1 92.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=92841 1572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 3297551171@19 2.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress router (0.8.10 (i386/l inux))..Content-Length: 0..Warning: 392 200.75.73.22:5060 "Noisy feedb ack tells: pid=32747 req_src_ip=192.168.0.15 in_uri=sip:2001@vozsobrei p.biz;user=phone out_uri=sip:2001@200.75.91.137:5060;user=phone;transp ort=udp via_cnt==1".... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip:10 10(a)vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip. biz;user=phone>;tag=3742565922..Call-ID: 3297551171@192.168.0.15..CSeq : 1 INVITE..Server: Cisco ATA 186 v2.15 ata18x (030122a)..Content-Len gth: 0.... # U 192.168.0.15:5061 -> 200.75.73.22:5060 CANCEL sip:2001@vozsobreip.biz;user=phone SIP/2.0..Via: SIP/2.0/UDP 19 2.168.0.15:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411 572..To: <sip:2001@vozsobreip.biz;user=phone>..Call-ID: 3297551171@192 .168.0.15..CSeq: 1 CANCEL..User-Agent: Cisco ATA 186 v2.15 ata18x (03 0122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 200 cancelling..Via: SIP/2.0/UDP 192.168.0.15:5061..From: <sip :1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobre ip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 CANCEL..Server: Sip EXpress router ( 0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73.22:506 0 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_uri=sip: 2001(a)vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5060;use r=phone;transport=udp via_cnt==1".... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... # U 200.75.73.22:5060 -> 192.168.0.15:5061 SIP/2.0 487 Request cancelled..Via: SIP/2.0/UDP 192.168.0.15:5061..Fro m: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@v ozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e5079b-f4f8..C all-ID: 3297551171@192.168.0.15..CSeq: 1 INVITE..Server: Sip EXpress r outer (0.8.10 (i386/linux))..Content-Length: 0..Warning: 392 200.75.73 .22:5060 "Noisy feedback tells: pid=32746 req_src_ip=192.168.0.15 in_u ri=sip:2001@vozsobreip.biz;user=phone out_uri=sip:2001@200.75.91.137:5 060;user=phone;transport=udp via_cnt==1".... # U 192.168.0.15:5061 -> 200.75.73.22:5060 ACK sip:2001@vozsobreip.biz:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.1 5:5061..From: <sip:1010@vozsobreip.biz;user=phone>;tag=928411572..To: <sip:2001@vozsobreip.biz;user=phone>;tag=0a96755a1d32d0360b3e454593e50 79b-f4f8..Call-ID: 3297551171@192.168.0.15..CSeq: 1 ACK..User-Agent: C isco ATA 186 v2.15 ata18x (030122a)..Content-Length: 0.... exit ------=_NextPart_000_0009_01C322CD.41C9F360--

1 0

None
by None 14 Dec '22

14 Dec '22

ACK messages SER keeps sending the 487 Request cancelled message as if the ACK didn't match. Any suggestion of what it can be? I'm using as the endpoint an ATA186, v2.15, and ser version 0.8.11. Best Regards, Alejandro

1 0

None
by None 14 Dec '22

14 Dec '22

/bin/sh -c 'cd /usr/local/mysql ; ./bin/safe_mysqld --user=mysql &' Gary Goodloff KALG Communications, LTD T: 402-898-4152 C: 402-208-5183 F: 413-294-3404 -----Original Message----- From: serusers-admin(a)iptel.org [mailto:serusers-admin@lists.iptel.org] On Behalf Of Todd Sent: Wednesday, May 21, 2003 5:05 PM To: Sip Questions List Subject: [Serusers] Mysql Connection Errors Ok..I have setup SER according to the guide on the iptel site. Here is a copy of what's in my ser.cfg _________________________________________ # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//usr/lib/ser/modules/mysql.so" loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//usr/lib/ser/modules/auth.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present rewriteFromRoute(); # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~"computergeeks.hopto.org") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("computergeeks.hopto.org", "subscriber")) { www_challenge("computergeeks.hopto.org", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } _________________________________________________ The modules are all where they say they are but when I try to start SER I get this in my messages log file May 21 18:58:45 computergeeks ser: ser startup succeeded May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: connect_db(): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: db_init(): Error while trying to connect database May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: mod_init(): Error while connecting database May 21 18:58:45 computergeeks /usr/sbin/ser[13089]: init_modules(): Error while initializing module usrloc Can someone please tell me what I'm doing wrong? I've searched all over the net for information on this...I'm sure it's something simple..but I'm overlooking it. If I disable the Mysql integration is works perfect, so I'm thinking it has something to do with the mysql.so module?!? Thanks Todd ------=_NextPart_000_00B1_01C31FBD.C81364A0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN"> <html xmlns:o=3D"urn:schemas-microsoft-com:office:office" = xmlns:w=3D"urn:schemas-microsoft-com:office:word" = xmlns=3D"http://www.w3.org/TR/REC-html40"> <head> <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; = charset=3Dus-ascii"> <meta name=3DProgId content=3DWord.Document> <meta name=3DGenerator content=3D"Microsoft Word 10"> <meta name=3DOriginator content=3D"Microsoft Word 10"> <link rel=3DFile-List href=3D"cid:filelist.xml@01C31FBD.C730CCD0">  <style>  </style>  </head> <body lang=3DEN-US link=3Dblue vlink=3Dpurple = style=3D'tab-interval:.5in'> <div class=3DSection1> Those SQL = Errors:<o:p></o:p> <o:p> </o:p> May 21 18:58:45 computergeeks ser: ser startup succeeded May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: connect_db(): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: db_init(): Error while trying to connect = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: mod_init(): Error while connecting = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: init_modules(): Error while initializing = module usrloc<o:p></o:p> <o:p> </o:p><= /p> This means that the mysql daemon is either not running or is not running on the default = port.<o:p></o:p> <o:p> </o:p><= /p> On freebsd – this is how I start my sql daemon:<o:p></o:p> From /etc/rc.local<o:p></o:p> /bin/sh = -c 'cd /usr/local/mysql ; ./bin/safe_mysqld = --user=3Dmysql &'<o:p></o:p> <o:p> </o:p> <div> Gary Goodloff KALG Communications, LTD T: 402-898-4152 C: 402-208-5183 F: 413-294-3404 <o:p></o:p> </div> -----Original = Message----- From: = serusers-admin(a)lists.iptel.org [mailto:serusers-admin@lists.iptel.org] On = Behalf Of Todd Sent: Wednesday, May 21, = 2003 5:05 PM To: Sip Questions = List Subject: [Serusers] Mysql Connection Errors <o:p> </o:p> Ok..I have setup SER according to the guide on the iptel site. Here is a copy of what's in my ser.cfg _________________________________________ # # $Id: ser.cfg,v 1.12 = 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start = config script # # ----------- global = configuration parameters ------------------------ debug=3D3       &= nbsp;  # debug level (cmd line: -dddddddddd) fork=3Dyes = log_stderror=3Dno # (cmd = line: -E) check_via=3Dno # (cmd. = line: -v) dns=3Dno       &n= bsp;   # (cmd. line: -r) rev_dns=3Dno      # (cmd. line: -R) port=3D5060 = children=3D4 = fifo=3D"/tmp/ser_fifo" # ------------------ = module loading ---------------------------------- # Uncomment this if you = want to use SQL database loadmodule "//usr/lib/ser/modules/mysql.so" loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" # Uncomment this if you = want digest authentication # mysql.so must be loaded = ! loadmodule "//usr/lib/ser/modules/auth.so" # ----------------- = setting module-specific parameters --------------- # -- usrloc params = -- #modparam("usrloc", "db_mode",   0) # Uncomment this if you = want to use SQL database # for persistent storage = and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params = -- # Uncomment if you are = using auth module # #modparam("auth", "secret", "alsdkhglaksdhfkloiwr") modparam("auth", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), = # uncomment also the = following parameter) # modparam("auth", "password_column", "password") # = -------------------------  request routing logic ------------------- # main routing = logic route{ # initial sanity checks -- = messages with # max_forwars=3D=3D0, or = excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) = { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if = pre-loaded route headers present rewriteFromRoute(); # if the request is for = other domain use UsrLoc # (in case, it does not = work, use the following command # with proper names and = addresses in it) if (uri=3D~"computergeeks.hopto.org") { if = (method=3D=3D"REGISTER") { # Uncomment this if you = want to use digest authentication if (!www_authorize("computergeeks.hopto.org", = "subscriber")) { www_challenge("computergeeks.hopto.org", "0"); break; }; save("location"); break; }; # native SIP destinations = are handled using our USRLOC DB if = (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri = now if (!t_relay()) = { sl_reply_error(); }; } _________________________________________________ The modules are all where they say they are but when I try to start SER = I get this in my messages log file May 21 18:58:45 = computergeeks ser: ser startup succeeded May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: connect_db(): Can't connect to local MySQL server = through socket '/var/lib/mysql/mysql.sock' (2) May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: db_init(): Error while trying to connect = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: mod_init(): Error while connecting = database May 21 18:58:45 = computergeeks /usr/sbin/ser[13089]: init_modules(): Error while initializing module = usrloc Can someone please tell me what I'm doing wrong?  I've searched all = over the net for information on this...I'm sure it's something simple..but = I'm overlooking it. If I disable the Mysql integration is works perfect, so I'm thinking it = has something to do with the mysql.so module?!? Thanks Todd <o:p></o:p> </div> </body> </html> ------=_NextPart_000_00B1_01C31FBD.C81364A0--

1 0

None
by None 14 Dec '22

14 Dec '22

stream, for such cases, is it possible? Any pointers would be great. -- Michael Vasilenko

1 0

None
by None 14 Dec '22

14 Dec '22

fgets failed: Illegal seek. It happens when 2 end points try to set up a session and one of them cancels it before any end-to-end connection is established. Has anyone found this problem before? (See attached file: cancelsegdump.tar.gz) Jaime --0__=PnXC9kr0ykdwrnb6uT2PR1PNFM05c9HrUcAPjHVL7TjiDE1fTDc3v998 Content-type: application/octet-stream; name="cancelsegdump.tar.gz" Content-Disposition: attachment; filename="cancelsegdump.tar.gz" Content-transfer-encoding: base64 H4sIAAseXj4AA+0da3PaSDKfqcp/mMuHjX0HRKM32Lg2cew912adLezsfkilVIM0Aq2FRPSI7b3a /e3XM5KwhAQIG5LYqykbBOrumenp6ackTOKZ1A3p2Iqns250Ez3bfhOwIKiy/ExIWvEdCyLGyjNB U0VNxqqkSAAvYVl5hoQdjKXU4jAiAULPAt9fOfd15x9pQ8IeliRR3keuTyzHG6Opb8UuRd1XyUH4 anobfnaT127ot2pghC781YSdBY4XJa81MaIp/NWEjcPA9c30rSZOEMBfXVg6dsIoIMHdUU1MEkcT /lJ31vQm8mfz95pY9Iaa/KU2fMT+a0Lb5oz9F6Ftx7OMGQnI1KA3Mz+I+sj2Y89Ch9bIAEx6hBwv o5UsDPq4fMU+1SIdOVMaGCBDNPhC3F30AIOPg69EmYlFnm5eVko0F6iBDDpmZAR+HBU4HQR5ipmM r6VGvhSpZFJeIFa5CdaRdn3/Kp7tiHjoGiGFrwM6c2/zfcCJ9OhjWV+towpspS4p0IumeUKZaqox PD4ygwaBH9xrfGyB2aaMyAiAhX7uFIIdTAOPZKQMk7juXm5ExiwKDoUbWRAUiyjC0d2p/QN098Gx UQ5rSqOJb919Hgzujl8MT346u7g8Gb7IkUL/u8+QsNjT7PyQ2ugF7CYSOb73YslArcCf5YhFtzN6 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--0__=PnXC9kr0ykdwrnb6uT2PR1PNFM05c9HrUcAPjHVL7TjiDE1fTDc3v998-- ------<80256CDA.004DFAC0.00@ruddick> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. ******************************************************************************* ------<80256CDA.004DFAC0.00@ruddick>

1 0

None
by None 14 Dec '22

14 Dec '22

Please give me an example of their interaction. Could I use B2BUA without SER(and without any SIP proxy)? Thanks Igor Vasiliev >

1 0

Re: [SR-Users] sr-users Digest, Vol 211, Issue 8
by Pooja Yadav 13 Dec '22

13 Dec '22

Hi Daniel and Everyone, Thanks for the inputs. We understand that Kamailio is a SIP signalling server irrespective of any core. But we have seen a thread in Open5Gs community where they established the call with Kamailio IMS. Below thread is for your reference. https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ If you have any inputs on this it is most welcome. We just want to check the feasibility of establishing voice/video call with 5G core. Your inputs will be appreciated. Thanks and regards Pooja Yadav > On 09-Dec-2022, at 4:33 PM, sr-users-request(a)lists.kamailio.org wrote: >

2 1

Delays sending over WSS
by Alex Balashov 12 Dec '22

12 Dec '22

Hi, I’ve run into stubbornly persistent problems with packets not reaching WSS clients until some seconds (5-6s or more!) after they were supposedly sent. The symptom is that Kamailio logs the event as sent at 20:00:00, including in the onsend_route, but in actual reality it takes quite a bit longer to send at the OS network stack level. This happens occasionally and quite universally, with no discernible pattern tied to particular endpoints, networks, etc. For a long time, I’ve just assumed this was a receive delay, since Kamailio logged an expectedly prompt relay time. I figured it was an application or browser execution delay and had nothing to do with networking. Based on what we know about the capriciousness of browsers and their internal task scheduling, this seemed rather plausible. However, I’ve since been able to ascertain that this is not the right understanding of the problem, because a parallel and unrelated WS keepalive, going from a different backend to the same application/browser tab/event loop, is consistently replied to in a timely manner. Anyway, I’m curious what else can be done to debug and/or fix this. I did turn up the size of my write buffers — tcp_wq_conn_max and the global one — quite a lot, some time ago and for unrelated reasons. I wonder if this might actually make the problem worse, since it leads to more bytes queued to send to an endpoint on the other side that could, conceivably, not be reading them fast enough. But I also wonder if this is tied up in some low-level TLS or WS parameters unrelated to more general TCP configuration. Thanks much! — Alex — Sent from mobile, apologies for brevity and errors.

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Kamailio SIP client registration issue
by Shiva Shankar Chittallu 09 Dec '22

09 Dec '22

Hi Team, I have installed kamailio sip server 5.5.5 version in ubuntu 20.04 in AWS instance with public IP. When im trying to register an device im getting below error in the syslog. Thanks & Regards Shiva 9100663346

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Does Kamailio supports Voice and Video call support for 5G Core
by Pooja Yadav 09 Dec '22

09 Dec '22

Hi Kamailio community, We just want to check does Kamailio IMS supports the functionality of Voice and Video call support (Vo5G) with 5G core? If yes, please share the document reference. Else, please guide us when it will be available. Any plans in upcoming release to support HTTP2 based SBI interface between P- CSCF and PCF. Your inputs will be highly appreciated. Thanks & Regards, Pooja Yadav

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Next online devel meeting - Thu, Dec 8, 2022, 15:00UTC
by Daniel-Constantin Mierla 08 Dec '22

08 Dec '22

Hello, to summarize what happened at the devel meeting in Dusseldorf a few weeks ago and plan the next steps for the project, we are planning an online development meeting on Thu, Dec 8, 2022, 15:00UTC. As usual, one of the targets is to sketch the roadmap to next major release so the developers and community members can syncronize and plan what should be done till Kamailio v5.7 (or 6.0) will be out as well as discuss about current state of the project, look for new ideas to improve collaboration within the community, a.s.o. The meeting will be done on a matrix chat room. More details are in the wiki page of the event, available at: * https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings… Feel free to add there topics that you want to be discussed. The proposed date is Thursday, December 08, 2022, at 15:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev

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Outgoing SIP/TLS connection error (msg_send_buffer(): tcp_send failed)
by Denys Pozniak 08 Dec '22

08 Dec '22

Hello! We use trunk with Twilio through a SIP/TLS. They establish incoming calls from random ports, like: (Twilio)3.3.3.3:54321 --> (Kamailio)2.2.2.2:5061 In case of sending in-dialog requests to them, Kamailio should establish an outgoing connection to the dst port 5061 (as per Route), like: (Kamailio)2.2.2.2 --> (Twilio)3.3.3.3:5061 But as per tcpdump traces and logs, Kamailio does not even try to establish an outgoing connection to Twilio. There should be no problems with routes and the network on that server, as regular telnet to Twilio shows that 5061 is open. What could be the reason and how to debug? *Script:* onsend_route { xlog("L_ERR","[$cfg(route)]*RELAY ONSEND* $rm / $du / $fs / $fsn / [$ci]\n"); } *Syslog:* DEBUG <core> [core/resolve.c:1259]: srv_sip_resolvehost(): 3.3.3.3:5061 proto=3 DEBUG <core> [core/resolve.c:1371]: srv_sip_resolvehost(): returning 0x55acd63a63c0 (3.3.3.3:5061 proto=3) ERROR <script>: [DEFAULT_SEND]*RELAY ONSEND* REFER / sip:3.3.3.3:5061;transport=tls / tls:10.1.1.1:5061 / sip_wan_tls / [123] DEBUG tm [../../core/forward.h:277]: msg_send_buffer(): sending to: 3.3.3.3:5061, force_socket=4, send_sock=0x7f090049b8e8 DEBUG tm [../../core/forward.h:292]: msg_send_buffer(): tcp_send failed DEBUG tm [t_fwd.c:1568]: t_send_branch(): send to 3.3.3.3:5061 (3) failed WARNING: tm [t_fwd.c:1588]: t_send_branch(): sending request on branch 0 failed DEBUG tm [t_funcs.c:335]: t_relay_to(): t_forward_nonack returned error -1 (-477) DEBUG tm [t_funcs.c:353]: t_relay_to(): -477 error reply generation delayed DEBUG sl [sl_funcs.c:558]: sl_run_callbacks(): execute callback for event type 1 *Software:* - Kamailio is listening on (+udp, +tcp): tls: 10.1.1.1:5061 name sip_wan_tls advertise 2.2.2.2:5061 - OS is Debian 11.5 - # kamailio -v version: kamailio 5.6.2 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 - # dpkg -l | grep ssl ii libssl1.1:amd64 1.1.1n-0+deb11u3 amd64 Secure Sockets Layer toolkit - shared libraries ii openssl 1.1.1n-0+deb11u3 amd64 Secure Sockets Layer toolkit - cryptographic utility -- BR, Denys Pozniak

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KAMCTL error
by Govind Swamy 08 Dec '22

08 Dec '22

[image: Error in KAMCTL.png] Hi am unable to create using kamctl

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