sr-users December 2005

sr-users@lists.kamailio.org

158 participants
313 discussions

Re;Re [Serusers] QoS and call quality
by Ashutosh kumar 08 Dec '05

08 Dec '05

Hi, Has anyone tried the QoS tricks given at http://www.voip-info.org/wiki/view/QoS+Linux+with+HFSC, <http://www.voip-info.org/wiki/view/QoS+Linux+with+HFSC,%0bhttp:/www.voip-in fo.org/wiki/view/QoS+Linux,http:/www.voip-info.org/wiki/view/QoS+with+Linux+ using+PRIO+and+HTB> http://www.voip-info.org/wiki/view/QoS+Linux, http://www.voip-info.org/wiki/view/QoS+with+Linux+using+PRIO+and+HTB <http://www.voip-info.org/wiki/view/QoS+Linux+with+HFSC,%0bhttp:/www.voip-in fo.org/wiki/view/QoS+Linux,http:/www.voip-info.org/wiki/view/QoS+with+Linux+ using+PRIO+and+HTB> Do they work? Or has anyone has experience using GlobalIPSound's products to enhance sound quality? Regards, Ashutosh -----Original Message----- From: sip [mailto:sip@arcdiv.com] Sent: Wednesday, December 07, 2005 7:11 PM To: Ashutosh kumar; serusers(a)lists.iptel.org Subject: Re: [Serusers] QoS and call quality SER has no control over the RTP voice stream in ANY way. That's handled by the UAs and your network equipment. N. On Wed, 7 Dec 2005 19:07:37 +0530, Ashutosh kumar wrote > Hi, > Does SER have any control or any can it be configured (if yes, please guide) to implement various QoS mechanisms like Diffserv,RSVP, Admission control etc.., to improve the voice quality and delays and/or jitter? Because lately, we are facing a decrease in call quality in our ser calls. > Or do we have to do this only at the network level, eg. Routers etc..? > > Any poiners/guides will be most welcome. > Regards, > Ashutosh Kumar >

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[Serusers] aliases table
by Xavier TRENTIN 08 Dec '05

08 Dec '05

Hi all, I'm running ser with dbtext, and I can't install mysql on my system. Can anybody tell me if the aliases table is loaded by ser when it starts? I also need to know what exactly each field must contain, so if somebody can give me an example of a line of the aliases table, it will be great. Thanks in advance. B.R. Xavier

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[Users] What's causing this error...
by Wilko van der Ploeg 08 Dec '05

08 Dec '05

After a clean installation of openser (so no 'old tables') a openser.cfg script is causing the following error messages : Dec 8 14:09:17 sip1 /sbin/openser[2419]: register_udomain(): Invalid table version (use ser_mysql.sh reinstall) Dec 8 14:09:17 sip1 /sbin/openser[2419]: domain_fixup(): Error while registering domain Dec 8 14:09:17 sip1 /sbin/openser[2419]: ERROR: fix_expr : fix_actions error It looks like there is some mismatch in table version in the installtion tar.... Does anyone know what should be altered to make it work ? Wilko

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[Serusers] Call forwarding to Cisco PSTN GW and correct charging
by Kyriakos Mavromichalis 08 Dec '05

08 Dec '05

Has anyone implemented call forwarding from SER to a PSTN CISCO gw? Lets say we have the situation below: A is caller from PSTN network B is a sip user that forwarded his number to C C is a pstn number If A calls B then he should be redirected to C , but A should be charged for calling to B , and B should be charged for calling to C. After some digging in the list and Cisco documents found out about CC-Diversion header and remote party id. What i want to do is to have a cdr created by the cisco machine that fills up the calling number field with A number and the redirecting number field with B number. Which header is the one to use ? thanks -- Kyriakos Mavromichalis Otenet Telecom

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[Serusers] Parse Vie Error : REGISTER
by Ricardo Martinez 08 Dec '05

08 Dec '05

Hello list. I updated my SER version to the last Stable version from CVS. I used almost the same ser.cfg file, now, for certain users trying to register in my SER box i'm getting this : REGISTER sip:sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;. From: 5565625856537 <sip:5565625856537@sip.mydomain.com>;tag=5369ee326df7cf6fo1. To: 5565625856537 <sip:5565625856537@sip.mydomain.com>. Call-ID: b3ea4367-c371f5b2(a)10.0.5.120. CSeq: 62661 REGISTER. Max-Forwards: 70. Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c92a5c19d60a1bd1a 9d35e91f35dd97f493",uri="sip:sip.mydomain.com",algorithm=MD5,response="6e894 b948364091bbc2e6bfeeb4e412c",qop=auth,nc=0000107c,cnonce="28e5d319". Contact: 5565625856537 <sip:5565625856537@200.6.65.82:30221>;expires=600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Linksys/RT31P2-3.1.3(LI). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. . Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via: invalid via - end of header in state 6 Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via on: <SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;^M From: 5565625856537 <sip:5565625856537@sip.mydomain.com>;tag=5369ee326df7cf6fo1^M To: 5565625856537 <sip:5565625856537@sip.mydomain.com>^M Call-ID: b3ea4367-c371f5b2(a)10.0.5.120^M CSeq: 62661 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c92a5c19d60a1bd1a 9d35e91f35dd97f493",uri="sip:sip.mydomain.com",algorithm=MD5,response="6e894 b948364091bbc2e6bfeeb4e412c",qop=auth,nc=0000107c,cnonce="28e5d319"^M Contact: 5565625856537 <sip:5565625856537@200.6.65.82:30221>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: get_hdr_field: bad via Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_msg: message=<REGISTER sip:sip.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;^M From: 5565625856537 <sip:5565625856537@sip.mydomain.com>;tag=5369ee326df7cf6fo1^M To: 5565625856537 <sip:5565625856537@sip.mydomain.com>^M Call-ID: b3ea4367-c371f5b2(a)10.0.5.120^M CSeq: 62661 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c92a5c19d60a1bd1a 9d35e91f35dd97f493",uri="sip:sip.mydomain.com",algorithm=MD5,response="6e894 b948364091bbc2e6bfeeb4e412c",qop=auth,nc=0000107c,cnonce="28e5d319"^M Contact: 5565625856537 <sip:5565625856537@200.6.65.82:30221>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: receive_msg: parse_msg failed Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via: invalid via - end of header in state 6 Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via on: <SIP/2.0/UDP 200.6.65.82:30219;branch=z9hG4bK-803aca07;^M From: 5565625848753 <sip:5565625848753@sip.mydomain.com>;tag=96e51eca631897b7o0^M To: 5565625848753 <sip:5565625848753@sip.mydomain.com>^M Call-ID: 79a49a8f-c7348aa(a)10.0.5.120^M CSeq: 62704 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c968045009a67b029 9bbd27e7a11961b95d",uri="sip:sip.mydomain.com",algorithm=MD5,response="05ba5 577a2171f34a306a21cdb253ac7",qop=auth,nc=0000107a,cnonce="2faea8f6"^M Contact: 5565625848753 <sip:5565625848753@200.6.65.82:30220>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP 200.6.65.82:30219;branch=z9hG4bK-803aca07;^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: get_hdr_field: bad via Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_msg: message=<REGISTER sip:sip.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 200.6.65.82:30219;branch=z9hG4bK-803aca07;^M From: 5565625848753 <sip:5565625848753@sip.mydomain.com>;tag=96e51eca631897b7o0^M To: 5565625848753 <sip:5565625848753@sip.mydomain.com>^M Call-ID: 79a49a8f-c7348aa(a)10.0.5.120^M CSeq: 62704 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c968045009a67b029 9bbd27e7a11961b95d",uri="sip:sip.mydomain.com",algorithm=MD5,response="05ba5 577a2171f34a306a21cdb253ac7",qop=auth,nc=0000107a,cnonce="2faea8f6"^M Contact: 5565625848753 <sip:5565625848753@200.6.65.82:30220>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: receive_msg: parse_msg failed Can someone help me here? What does it means? Thanks in advance Regards Ricardo Martinez

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[Serusers] Contract work needed for custom SER configuation
by Andy Pyles 07 Dec '05

07 Dec '05

Please contact me privately if you are very familiar with NATHELPER and AVPOPS modules. I'm looking for a custom configuration solution, that needs to be delivered very quickly. Thanks, Andy

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Re: [Serusers] set timer for "Request Timeout"
by Matteo Piazza 07 Dec '05

07 Dec '05

Hi Jens Can you send me your complete ser.cfg file. I'am interested to see you complete configuration Regards Matteo Hello all, I have following records in the table 'usr_preferences': username attribute value 1000 fwdbusy sip:1001 at example.de 1000 fwdnoanswer sip:1001 at example.de I use these records to get the functionallity of "Call Forwarding - Busy" and "Call Forwarding - No Answer" into the OpenSer - server. I use a failure_route to act on the SIP - messages "Request Timeout (408)" and "Busy(486)" and forward the INVITE - message to the URI saved in the table 'usr_preferences'. See my snippet of the openser.cfg: ----snip --- failure_route[1] { if (isflagset(27) && t_check_error(408)) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); route(6); break; }; }; if (isflagset(26) && t_check_error(486)) { if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); resetflag(26); route(6); break; }; }; }; ----snap --- These configuration works fine. Is there a possibillity to set a timer for the phone after the server receives the message "Request Timeout" from the phone? Or can I let the server ring for example five times and generate the message? Greetings Jens ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

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[Serusers] How to Minimize rtp proxy usage for all cases including when behind the same NAT
by S G 07 Dec '05

07 Dec '05

Hello, I have SER setup to only use the rtpproxy if the client is behind a symmetric NAT. This is accomplished by using STUN since STUN will not modify the contact headers if it detects a symmetric NAT. So SER see's the local address in the contact header and only forces rtpproxy for those types of call. When users are behind the same NAT then ser detects this using AVP's and force_rtpproxy is not used for the call. This only works when STUN is not enabled. Since when STUN succeeds the orignal local address of the phone is changed to the public address in the sip message. So when two STUN enabled clients try to call each other from behind the same nat the call fails. The call fails because the NAT does not support harpin of media as most don't. The only way this call would work is if the contacts could be changed back to use their local addresses pre-STUN. How can SER be instructed to use the local address for calls behind the same NAT when STUN is enabled? One way I can think of doing this is to change nathelper to extract the original IP from the VIA header and rewrite the SDP. In the SIP trace i see only IP's in contact headers and SDP are changed when STUN succeeds. Is there a simpler way to accomplish this? Thanks, Sumeet _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Users] CALL FORWARD / PSTN GW
by Kjell Arild Frigstad 07 Dec '05

07 Dec '05

Hello! I am trying to implement Call forwarding. Server runs openser 1.0.0 and I have a Cisco PSTN GW where only defined calling numbers are permitted (because of accounting on the "upstream" provider). My problem is when a call from PSTN is beeing forwarded and I have to "turn it back" to the PSTN gatway. Calls from : 1) Sip server ---> SIP-CALL-FW ---> PSTN GW OK, the calling number is also changed to the number which has turned call forwarding on. The display on the phone also showing the number which has turned Call forwarding on.. 2) PSTN ---> SIP-CALL-FW ---> SIP SERVER Works OK 3) PSTN ----> SIP-CALL-FW ---> PSTN is not working. Dialpeer matching in Cisco not OK because calling number not OK. I can define a way so that all calls can go through the GW. Then I sometimes get a "482 loop detected". The disply on the phone showing the orginal calling number (not the number wich hat turnd Call forwarding on). Is this the right way to "turn" a call back to where it came from? Have a missed a "rebuild" of the message? if (avp_db_load("$ruri","$fw_to") & method=="INVITE") { # CALL Forward log(1, "INFO: AVP_db_load fw_to true \n"); #save called number avp_write("$ruri/username","s:org_to"); #Insert call forward number avp_pushto("$ruri/username","$fw_to"); #is call forward number local or to PSTN if (does_uri_exist()) { log(1, "INFO:Call FW to Local users\n"); if (!lookup("location")) { log(1, "INFO: call FW to local user, not online\n"); route(4); return; }; log(1, "INFO:Call FW to local users online\n"); uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060"); if (!t_relay()) { sl_reply_error(); return; }; return; } else { log(1, "INFO: Call FW to PSTN\n"); #here is somting wrong. What??? #move Called number as From number and send to PSTN GW uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060"); rewritehostport("80.239.101.34:5060"); append_hf("P-hint: call fwd to PSTN\r\n"); append_branch(); record_route(); if (!t_relay()) { sl_reply_error(); return; }; }; return; } Cisco trace : calling from 38092382 (PSTN) to 38707130 (on the SIP server) witch is forwarded to 95066833 (PSTN) Call comming in from PSTN to SIP-server : Sent: (from Cisco) INVITE sip:38707130@212.125.213.151:5060 SIP/2.0 Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38092382@80.239.101.34>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Date: Fri, 02 Dec 2005 18:41:38 GMT Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 457212883-1654264282-2767847443-3293903248 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: <sip:38092382@80.239.101.34>;party=calling;screen=yes;privacy=off Timestamp: 1133548898 Contact: <sip:38092382@80.239.101.34:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 253 v=0 o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34 s=SIP Call c=IN IP4 80.239.101.34 t=0 0 m=audio 16788 RTP/AVP 18 8 0 c=IN IP4 80.239.101.34 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 *Dec 2 18:41:38.278: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38092382@80.239.101.34>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 CSeq: 101 INVITE Server: OpenSer (1.0.0 (i386/linux)) Content-Length: 0 Warning: 392 212.125.213.151:5060 "Noisy feedback tells: pid=29232 req_src_ip=80.239.101.34 req_src_port=55717 in_uri=sip:38707130@212.125.213.151:5060 out_uri=sip:95066833@80.239.101.34:5060 via_cnt==1" *Dec 2 18:41:38.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:95066833@80.239.101.34:5060 SIP/2.0 Record-Route: <sip:212.125.213.151;ftag=2A699224-CC6;lr=on> Via: SIP/2.0/UDP 212.125.213.151;branch=z9hG4bK1e4.11a1a9d3.0 Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38707130@212.125.213.151:5060>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Date: Fri, 02 Dec 2005 18:41:38 GMT Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 457212883-1654264282-2767847443-3293903248 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 69 Remote-Party-ID: <sip:38092382@80.239.101.34>;party=calling;screen=yes;privacy=off Timestamp: 1133548898 Contact: <sip:38092382@80.239.101.34:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 253 P-hint: call fwd to PSTN v=0 o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34 s=SIP Call c=IN IP4 80.239.101.34 t=0 0 m=audio 16788 RTP/AVP 18 8 0 c=IN IP4 80.239.101.34 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 Dial peer maching (call forward from SIP server). Calling number is wrong, should be 38707130 (number who has put forwarding on) Calling Number=38092382, Called Number=95066833, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Dec 2 19:44:08.330: //-1/D6732AD9A532/DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore: Calling Number=, Called Number=95066833, Peer Info Type=DIALPEER_INFO_SPEECH *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=95066833 Best regards Kjell Arild

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[Users] Limit simultaneous calls
by Christophe Irles 07 Dec '05

07 Dec '05

Hi, Is there a way to limit the number of simultaneous calls ? According the provider, the bandwith will not permit a lot of simultaneous call so in order to prevent very bad feeling I prefer to limit the use. In a second step, I wondering if this type of restriction can be extended according the phone domain. For example, simultaneous call betwwen two sub-sites of the same enterprise is limited to 10 but all others a limited to 5 for example. I know the ACC module but it's not the one fot this even if the information stored could resolve this point. Perhaps another module already exist ? Thanks, Christophe

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sr-users December 2005