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500 Server Error Occurred 1/sl

sorted by: [ date ] [ thread ] [ subject ] [ author ] yes, ngrep. Check the ngrep manual: "-P char Change the non-printable character from the default ``.'' to the character specified." In this case the '.' is '\r' character. Regards, Ovidiu Sas On 3/16/07, CARTWRIGHT, CORY C (ATTSNET) wrote: > Are the "." after each line in the capture below caused by ngrep or just > in the email? > > > > -----Original Message----- > From: devel-bounces at openser.org [mailto:devel-bounces at openser.org] On > Behalf Of Bogdan-Andrei Iancu > Sent: Tuesday, February 13, 2007 1:09 PM > To: Ovidiu Sas > Cc: devel at openser.org > Subject: Re: [Devel] 500 Server error occurred (1/SL) > > > Hi Ovidiu, > > second INVITE looks to be a retransmission. Are you using t_newtran() + > t_relay() ?? what is the script sequence for relaying? > > regards, > bogdan > > Ovidiu Sas wrote: > > Hi, > > > > > > I got this bizarre 500 error (openser 1.2.0-pre4-tls (arm/linux)). > > > > 192.168.2.102 - CISCO IP PHONE > > 192.168.2.22 - openser > > 192.168.2.20 - asterisk > > > > Cisco --> openser --> asterisk > > > > > > Here's the the log: > > ERROR:tm:t_forward_nonack: no branch for forwarding > > ERROR:tm:w_t_relay: t_forward_nonack failed > > ERR: t_relay() in route[1] > > > > > > and here's the ngrep: > > # > > U 2007/02/12 21:03:59.644707 192.168.2.102:50611 -> 192.168.2.22:5060 > > INVITE sip:4169771414 at 192.168.2.22;user=phone SIP/2.0. > > Via: SIP/2.0/UDP 192.168.2.102:5060;branch=z9hG4bK6fc15dd5. > > From: "Line 6" > > ;tag=00036be7b12459bc6a35fdb5-427c1545. > > To: . > > Call-ID: 00036be7-b1240073-27eb575a-3c266a6b at 192.168.2.102. > > Max-Forwards: 70. > > Date: Tue, 13 Feb 2007 02:03:38 GMT. > > CSeq: 102 INVITE. > > User-Agent: Cisco-CP7960G/7.5. > > Contact: . > > Proxy-Authorization: Digest > > > username="line-6",realm="192.168.2.22",uri="sip:4169771414 at 192.168.2.22; > user=phone",response="16f0ea81a76e28996eefcf4d9b5cfb38",nonce="45d0d76be > d759d9c74a1f2c55e052931bda8c01a",algorithm=md5. > > > > Expires: 180. > > Remote-Party-ID: "Line 6" > > > ;party=calling;id-type=subscriber;privacy=off; > screen=yes. > > > > Content-Length: 256. > > Content-Type: application/sdp. > > Content-Disposition: session;handling=optional. > > . > > v=0. > > o=Cisco-SIPUA 9502 0 IN IP4 192.168.2.102. > > s=SIP Call. > > t=0 0. > > m=audio 29660 RTP/AVP 0 8 18 101. > > c=IN IP4 192.168.2.102. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:1

Join INTELLIGENT WORK FORUMSFOR COMPUTER PROFESSIONALS Log In Come Join Us! Are you aComputer / IT professional?Join Tek-Tips Forums! Talk With Other Members Be Notified Of ResponsesTo Your Posts Keyword Search One-Click Access To YourFavorite Forums Automated SignaturesOn Your Posts Best Of All, It's Free! Join Us! *Tek-Tips's functionality depends on members receiving e-mail. By joining you http://opensips.org/pipermail/devel/2007-March/006591.html are opting in to receive e-mail. Posting Guidelines Promoting, selling, recruiting, coursework and thesis posting is forbidden.Tek-Tips Posting Policies Jobs Jobs from Indeed What: Where: jobs by Link To This Forum! Add Stickiness To Your Site By Linking To This Professionally Managed Technical Forum.Just copy and http://www.tek-tips.com/viewthread.cfm?qid=1672813 paste the BBCode HTML Markdown MediaWiki reStructuredText code below into your site. Avaya: IP Office Forum at Tek-Tips HomeForumsWiring ClosetPhone SystemsAvaya: IP Office Forum SIP Trunk Not Working thread940-1672813 Forum Search FAQs Links MVPs SIP Trunk Not Working SIP Trunk Not Working jcallen03 (Programmer) (OP) 24 Jan 12 11:14 I am trying to set up a Nexvortex SIP trunk.Here are the messages I am seeing with the monitor.695464mS SIP Tx: UDP 192.168.42.2:5060 -> 66.23.129.253:5060INVITE sip:@192.168.42.1 SIP/2.0Via: SIP/2.0/UDP 192.168.42.2:5060;rport;branch=z9hG4bK66febafa5b39e8e2f4bad834d3167158From: "Membership" ;tag=ac90fe895a3fe640To: Call-ID: ad01a39c824cab8e851e4dd45a4d110a@192.168.42.2CSeq: 151821899 INVITEContact: "Membership" Max-Forwards: 70Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATEContent-Type: application/sdpSupported: timerContent-Length: 218v=0o=UserA 3991637103 830104317 IN IP4 192.168.42.2s=Session SDPc=IN IP4 192.168.42.2t=0 0m=audio 49154 RTP/AVP 0 8 18a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=no695535mS SIP Rx: UDP 66.23.129.253:5060 -> 192.168.42.2:5060SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)Via: SIP/2.0/UDP 192.168.42.2:5060;rport=5060;branch=z9hG4bK66febafa5b39e8e2f4bad834d3167158;received=67.53.0.87From: "Membership"

by: [ date ] [ thread ] [ subject ] [ author ] Hello, On 07/12/06 14:02, zhan http://lists.kamailio.org/pipermail/users/2006-July/005589.html li wrote: > RTP proxy is enabled, happened for http://www.ip-phone-forum.de/showthread.php?t=262614 PSTN termination. > also happpened when 2 SIP client behind the different NAT, or when the > 2 SIP client behind same NAT, the BYE message will cause the same > error message. > > The configuration file is just 500 server the same as the file from startup > package for RTP Proxy. you should get some error messages in syslog /var/log/syslog or /var/log/messages. Something happened and openser couldn't process the message. If you do not get something satisfactory, run openser in debug mode: debug=7 log_stderror=yes Do you get this situation 500 server error often or it happens once in a while. Cheers, Daniel > > > > > U 69.88.77.66:5120 -> 201.22.33.178:5060 > SIP/2.0 200 OK..Via: SIP/2.0/UDP 201.22.33.178:5060;branch=0..Call-ID: > c87 > e3594-32a56723-65e5 [!at] 201.22.33.178..CSeq (replace the [!at] with > a @): 1 OPTIONS..From: sip:pinger at opens > er.org;tag=ad27d066..To: sip:69.88.77.66:5120;tag=qjp2KfWkSXDEoML2..Cont > act: ..Allow: INVITE, ACK, OPTIONS, BY > E, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE..Accept: > application > /sdp, message/sipfrag, application/dtmf-relay..Supported: > replaces..Content > -Length: 0.. > > U 69.88.77.66:5120 -> 201.22.33.178:5060 > INVITE sip:12127977000 at 201.22.33.178 SIP/2.0..Via: SIP/2.0/UDP 192.1 > 68.1.10:5120;branch=z9hG4bK2xcF52ZeIG9RyjPc..Max-Forwards: > 70..User-Agent: > COSUN V1.49.002 CFG0 ..From: "1001" 178>;tag=CXTKU08d8synkO1U..To: "12127977000" 03.98.176.178>..Call-ID: ziblKnSznNbjxCjv [!at] 192.168.1.10..Contact > (replace the [!at] with a @): 99500500 [!at] 192.168.1.10 (replace the [!at] with a @):5120>..CSeq: > 1 INVITE..Supported: replaces..Content-T > ype: application/sdp..Content-Length: 315....v=0..o=- 91081239 > 73141710 IN > IP4 192.168.1.10..s=SIP CALL..c=I

Suche Forum VoIP-Anbieter sipgate [Problem] Server error occurred (1/SL) (500) Wenn dies Ihr erster Besuch hier ist, lesen Sie bitte zuerst die Hilfe - H�ufig gestellte Fragen durch. Sie m�ssen sich vermutlich registrieren, bevor Sie Beitr�ge verfassen k�nnen. Klicken Sie oben auf 'Registrieren', um den Registrierungsprozess zu starten. Sie k�nnen auch jetzt schon Beitr�ge lesen. Suchen Sie sich einfach das Forum aus, das Sie am meisten interessiert. Ergebnis 1 bis 5 von 5 Thema: Server error occurred (1/SL) (500) Themen-Optionen Druckbare Version zeigen Thema abonnieren… Anzeige Linear-Darstellung Zur Hybrid-Darstellung wechseln Zur Baum-Darstellung wechseln 16.09.2013,22:07 #1 :=) Profil Beitr�ge anzeigen Private Nachricht IPPF-Aufsteiger Registriert seit 17.06.2007 Beitr�ge 36 Server error occurred (1/SL) (500) Hallo ich habe seit gestern folgendes Problem. Meine Sipgate Rufnummer empf�ngt Anrufe von aussen aber kan kann keine Gespr�che mehr f�hren. Meine Anlge ist wie folgt aufgebaut. --Kabelanschluss Primacom mit Thompson Kabelmodem -- an Lan1 ist via Kabel eine FB 7390 verbunden. -- Die FB ist im IP-Client Modus ... d.h. sie nutzt an Lan1 das Internet vom Thompson und verteilt es auch via WLAN & restlichen LAN -- In der FB ist Sipgate eingerichtet und ein DECT Telefon eingerichtet. Hat bisher auch immer alles geklappt. Konnte telefonieren und angerufen werden. Nur seit gestern bekomme ich beim Anruf folgende Fehlermeldung: Internettelefonie mit 0170xxxxxxxx@sipgate.de �ber sipgate.de war nicht erfolgreich. Ursache: Server error occurred (1/SL) (500) Kann mir wer zu der Fehlermeldung was sagen ? Anrufen kann man mich komischerweise. bzw da klingelt das Telefon. Update: So mal ein kleines Update. Wie ich ja bereits berichtet habe kann man meine Nummer anrufen. Leider h�rt der Anrufer mich dann nicht ... ich h�re aber ihn. Wenn ich oft nacheinander probiere jemanden von meinem Sipgate Telefon anzurufen klappt auch irgendwann die Verbindung. Dann h�re ich den Anrufer und der Anrufer h�rt auch mich. Update 2: Um mal auszuschliessen das es an Sipgate liegt habe ich mal Personal-VoIP als anbieter getestet. Gleiche Problem. Komischweise werden die Nummern aber ordentlich in der FB regestriert. Update 3: Habe eben mit Primacom telefoniert um mal nachzufragen ob was ge�ndert wurde. Antwort: Nein, alles beim alten. SIP wird nicht unterbunden und ge�ndert wurde auch nichts. Klingt ja fast als wenn der Fehler in mei

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Server Error Occurred sl p sorted by date thread subject author Hi Daniel Thanks for the answer but I relatedl already tested if the problem is there and even commenting the prefix out the error is exactly the same That is why I test it using pstn gw ip XX XX XX pstn gw port And route PSTN Without changing original PSTN routing I'm getting the same problem Could you think on something else that can be affecting that Helena From sr-users-bounces at lists sip-router org mailto sr-users-bounces at lists sip-router org On Behalf Of Daniel-Constantin Mierla Sent martes de

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