sr-users May 2014

sr-users@lists.kamailio.org

96 participants
156 discussions

BLF issue with Kamailio behind NAT
by Errol Samuels 02 May '14

02 May '14

Hello All, I have an integration with Kamailio and FusionPBX (Freeswitch) working great but I been trying to troubleshoot a BLF issue with Kamailio where the BLF seems to only work when Kamailio is on a Public IP rather than behind NAT. The BLF works fine when we register directly with Freeswitch either on a Public IP or behind NAT. Both Kamailio and FusionPBX (Freeswitch) are behind pfSense firewall on private IPs and registrations, Inbound and Outbound Calls etc are working fine. Just this BLF issue giving me some grief. The phone in question is a Yealink T20. I made some SIP Traces and I noticed the following: In the FS Message Header I see we have the SUBSCRIBE: ***** Begin Freeswitch trace**** Frame 13: 619 bytes on wire (4952 bits), 619 bytes captured (4952 bits) Internet Protocol Version 4, Src: 192.168.1.138 (192.168.1.138), Dst: 207.xx.xx.109 (207.xx.xx.109) User Datagram Protocol, Src Port: x11 (6062), Dst Port: ca-2 (5065) Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:testfs.mydomain.com:5065 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.138:6062;branch=z9hG4bK748715490 From: "302" <sip:302@testfs.mydomain.com:5065>;tag=530545144 To: "302" <sip:302@testfs.mydomain.com:5065> Call-ID: 2118669441(a)192.168.1.138 CSeq: 8 REGISTER Contact: <sip:302@192.168.1.138:6062> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T20P 9.72.0.30 Expires: 300 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 Frame 14: 698 bytes on wire (5584 bits), 698 bytes captured (5584 bits) Internet Protocol Version 4, Src: 207.xx.xx.109 (207.xx.xx.109), Dst: 192.168.1.138 (192.168.1.138) User Datagram Protocol, Src Port: ca-2 (5065), Dst Port: x11 (6062) Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.138:6062 ;branch=z9hG4bK748715490;received=2.xx.xx.187;rport=6062 From: "302" <sip:302@testfs.mydomain.com:5065>;tag=530545144 To: "302" <sip:302@testfs.mydomain.com:5065>;tag=j71FeN5gcBjDe Call-ID: 2118669441(a)192.168.1.138 CSeq: 8 REGISTER Contact: <sip:302@192.168.1.138:6062 ;received=2.xx.xx.187:6062>;expires=300 Date: Thu, 24 Apr 2014 17:03:18 GMT User-Agent: FreeSWITCH-mod_sofia/1.2.23+git~20140419T040926Z~8a20da75ef~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 Frame 15: 509 bytes on wire (4072 bits), 509 bytes captured (4072 bits) Internet Protocol Version 4, Src: 192.168.1.138 (192.168.1.138), Dst: 10.22.110.9 (10.22.110.9) User Datagram Protocol, Src Port: x11 (6062), Dst Port: ca-2 (5065) Session Initiation Protocol (SUBSCRIBE) Request-Line: SUBSCRIBE sip:301@10.22.110.9:5065 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.138:6062;branch=z9hG4bK148600064 From: "302" <sip:302@testfs.mydomain.com:5065>;tag=2146254460 To: <sip:301@testfs.mydomain.com:5065>;tag=70uoBUVRvJyr Call-ID: 168472766(a)192.168.1.138 CSeq: 2 SUBSCRIBE Contact: <sip:302@192.168.1.138:6062> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: Yealink SIP-T20P 9.72.0.30 Expires: 1800 Event: dialog Content-Length: 0 ****** End Freeswitch trace ******** In the Kamailio Header there is no SUBSCRIBE for the BLF. ***** Begin Kamailio trace**** Frame 17: 809 bytes on wire (6472 bits), 809 bytes captured (6472 bits) Internet Protocol Version 4, Src: 192.168.1.138 (192.168.1.138), Dst: 207.xx.xx.108 (207.xx.xx.108) User Datagram Protocol, Src Port: x11 (6062), Dst Port: sip (5060) Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:test.mydomain.com:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.138:6062;branch=z9hG4bK1572502835 From: "302" <sip:302@test.mydomain.com>;tag=1877226231 To: "302" <sip:302@test.mydomain.com> Call-ID: 887242692(a)192.168.1.138 CSeq: 3 REGISTER Contact: <sip:302@192.168.1.138:6062> Authorization: Digest username="302", realm="test.mydomain.com", nonce="U1lHHFNZRfB7/5Lyb/Rz3p/xu26B1lpE", uri="sip:test.mydomain.com:5060", response="7687c7d3141d1898e63a1cc9538f05d7", algorithm=MD5 Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T20P 9.72.0.30 Expires: 300 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 Frame 18: 496 bytes on wire (3968 bits), 496 bytes captured (3968 bits) Internet Protocol Version 4, Src: 207.xx.xx.108 (207.xx.xx.108), Dst: 192.168.1.138 (192.168.1.138) User Datagram Protocol, Src Port: sip (5060), Dst Port: x11 (6062) Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.138:6062 ;branch=z9hG4bK1572502835;rport=6062;received=2.xx.xx.187 From: "302" <sip:302@test.mydomain.com>;tag=1877226231 To: "302" <sip:302@test.mydomain.com >;tag=ff6b32c6b28b8954915484dd79f46b1c.fcdc Call-ID: 887242692(a)192.168.1.138 CSeq: 3 REGISTER Contact: <sip:302@192.168.1.138:6062 >;expires=300;received="sip:2.xx.xx.187:6062" Server: kamailio (4.0.6 (x86_64/linux)) Content-Length: 0 Frame 19: 316 bytes on wire (2528 bits), 316 bytes captured (2528 bits) Internet Protocol Version 4, Src: 207.xx.xx.108 (207.xx.xx.108), Dst: 192.168.1.138 (192.168.1.138) User Datagram Protocol, Src Port: sip (5060), Dst Port: x11 (6062) Session Initiation Protocol (OPTIONS) Request-Line: OPTIONS sip:2.xx.xx.187:6062 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.22.110.11:5060;branch=0 From: sip:pinger@kamailio.org ;tag=uloc-53579917-4e9-89-0b632cbb-a8aa99c To: sip:2.xx.xx.187:6062 Call-ID: f9749d7-2d0b0be7-c678c54(a)10.22.110.11 CSeq: 1 OPTIONS Content-Length: 0 ***** End Kamailio trace**** Anyone faced this issue or have any ideas? regards, Errol

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Voice is breaking when using Kamailio 4.1.3
by varun pratapsingh 02 May '14

02 May '14

HI All/ Daniel, We are using the Kamailio+ RTPProxy as a SIP and RTP Proxy to call the Phones. Our setup is like below: SIP UAC--------->Kamailio+RTPProxy------>SIP Server------->Phone Now when we call from UAC to Phone then voice is coming good at the UAC side. But we receive the bad voice at the Phone side the Voice is breaking and hardly audible. Can anybody faced this problem. Please suggest me any solution if any. I can mail any settings if anybody asks. Thanks in advance. Regards Varun

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Adding dialogs to a profile in a failure_route
by Alex Balashov 02 May '14

02 May '14

Hi, Am I not allowed to call dlg_manage() and add a dialog to a profile inside a request_route that is called out of a failure_route? The sequence of events is like this: 1. INVITE ----------------> 2. <---------- 302 redirect 3. ACK -------------------> 4. INVITE (new branch) ---> It's in #4 that I add the dialog to a profile and call dlg_manage(). There's a slight ambiguity because I am calling in a request_route, but the request_route is really called from a failure_route that catches the 302. It contains common logic that is used in both #1 and #4. So, the message being processed is 302, but the execution context is request_route. The docs say that dlg_manage() can be used in a request_route, and that set_dlg_profile() can be called from request/branch/reply/failure routes. But, I suppose it's logical that dlg_manage() ought to operate on the initial dialog-opening request. However, since the 302 is part of the same transaction as the initial INVITE, I would think that's okay. Is it? Anyway, I get these messages a lot: Apr 28 17:41:04 p01 /usr/local/sbin/kamailio[8581]: CRITICAL: dialog [dlg_profile.c:444]: set_dlg_profile(): BUG - dialog not found in a non REQUEST route (1) Apr 28 17:41:04 p02 /usr/local/sbin/kamailio[8581]: ERROR: dialog [dialog.c:800]: w_set_dlg_profile(): failed to set profile At no point am I actually calling set_dlg_profile() outside of a request_route, but the request_route might be called from a failure_route. So, the question is, am I doing something wrong? What's the best way to accommodate this scenario? I don't know if I want to track the dialog or add it to a profile until after I get the 302. Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Need an explanation of the contents of the LCR gw_uri_avp for replacement I am making to some of the functionality of the LCR module
by Will Ferrer 01 May '14

01 May '14

Hi I am working on replacing the functionality of the load_gws LCR function call in our application. I have found you can overwrite the contents of the gw_uri_avp (default: $avp(i:709)) substitute this function call. However I am a bit unclear about what goes in this avp. I was wondering if any one knew of a place I could find this information. The contents of the avp is an list whose entries look something like this :"2|1|0|||258722358||5060||1|1". The values separated by the | are likely values from the database, and some of them are obvious (like the port and the prefix), but others are not. In particular the longer number baffles me, it's not the ip of the gateway from the lcr_gw table, unless it is but has been altered in some way. Any advice any one has for me would be greatly appreciated. All the best Will Ferrer Switchsoft LLC

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S-CSCF reg_fetch_contacts
by Andrea Meroni 01 May '14

01 May '14

Hi all, does anybody know the correct sintax to get data from the variable $imssulc? Andrea

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Kamailio in TLS and Non-TLS mode
by aawaise 01 May '14

01 May '14

Hello all, I want to know that is it possible to connect clients to kamailio server in TLS and Non-TLS mode simultaneously ?? I believe I will have to add both ports (for TLS and Non-TLS ) in kamailio.cfg. Besides that what will be required if its possible ? Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-in-TLS-and-Non-TLS-mode-tp… Sent from the Users mailing list archive at Nabble.com.

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sr-users May 2014