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Incoming Call Got Sip Response 500 Internal Server Error

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I was thinking of doing the same, but as a challenge I still want to investigate further. Thanks in advance!! Who can help: henkoegema Oldsterisk Posts: 210Joined: Sun May 14, 2006 2:52 am E-mail henkoegema Top by henkoegema » Thu Dec 20, 2007 3:51 am Noboddy who can help? Mein KontoSucheMapsYouTubePlayNewsGmailDriveKalenderGoogle+ÜbersetzerFotosMehrShoppingWalletDocsBooksBloggerKontakteHangoutsNoch mehr von GoogleAnmeldenAusgeblendete FelderNach Gruppen oder Nachrichten suchen WIKI SUPPORT SEARCH LOGIN REGISTER Brekeke Forum Index » Brekeke SIP Server Forum Got SIP response 500 "Internal Server Error" http://colvertgroup.com/500-internal/iis-7-500-internal-server-error-asp.php

Style Default Style Help Home Top RSS Terms and Rules Forum software by XenForo™ ©2010-2015 XenForo Ltd. Not that I've had reported. Reply URL 1 Mike Mc. ● 2 years ago Ossovh wrote:I am facing the same issue and even G711a doesn't work. It's in sip.cfg: Make sure it says 1 and not 0 Quote: |-----Original Message----- |From: [email protected] |[mailto:[email protected]] On Behalf Of |Doug Lytle |Sent: Sunday, February 26, 2006

Sip 500 Internal Server Error Cisco

I have exactly the same problem with Polycom phones while transferring... I am using chandave's instructions. Contact your VoIP Provider for more assistance on this issue, as the issue is likely happening server-side.

  • Release Release 7.0(2) Associated CDETS# None.
  • Whats wrong?
  • Thanks!
  • It is routing properly.
  • srl100 Newsterisk Posts: 28Joined: Thu Feb 01, 2007 5:51 am Website Top by henkoegema » Tue Dec 18, 2007 10:01 am srl100 wrote:Enable sip debugging and post your logs.
  • Each extension on the server is logging "-- Got SIP response 500 "Internal Server Error" back from 10.1xx.xx.xx:5060" every few seconds.
  • I had a similar problem with another provider in México and I configured the asterisk with the next line (sip.conf): [general] useragent=name of the client that the provider gives me This
  • The funny thing is that it has been working well (some time ago now) henkoegema Oldsterisk Posts: 210Joined: Sun May 14, 2006 2:52 am E-mail henkoegema Top by henkoegema »
  • Thanks Mike.

My VoIP provider is Poivy.com. CK -------------- next part -------------- An HTML attachment was scrubbed... Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Sip 2.0 500 Server Internal Error Asterisk As I said, you can send a log if you'd like us to verify it.

NOTHING was changed on my site, since it stopped working. Sip 2.0 500 Server Internal Error Audiocodes gremln007 Hello all, I finally got VoiceStick working under Asterisk (name your trunk i2telecom.com and nothing else!!!) but it isn't working exactly right. voipsolutionsllc, Jul 18, 2006 #4 gremln007 voipsolutionsllc said: While I don't have a problem using a softphone with Voicstick and Asterisk... http://www.asteriskguru.com/archives/image-vp195021.html sip.conf. ;register to voipraider register => hoegema1946:[email protected][voipraider-out]type=peerusername=hoegema1946secret=mysecrethost=sip.Voipraider.comrealm=sip.Voipraider.comfromuser=00324763788xxcontext=defaultcanreinvite=noinsecure=veryqualify=300nat=yesport=5060dtmfmode=inbanddisallow=allallow=alawallow=ulaw routes-outgoing.confexten => _000234.,1,NoOP(Time=${STRFTIME(${EPOCH},,%H)}.${STRFTIME(${EPOCH},,%M)})exten => _000234.,n,Dial(SIP/voipraider-out/${EXTEN:1})exten => _000234.,n,Congestion() This is CLI output with debug off.: -- Starting simple switch on 'Zap/1-1' -- Executing [[email protected]:1] NoOp("Zap/1-1",

Please contact your VoIP Provider or Server Administrators about this issue. (I do note that you have only ONE codec enabled on your Mobile Network, which would not result in an Sip/2.0 503 Service Unavailable I only can accept calls in wifi bot not make. Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Follow-Ups: Re: SIP response 500 "Server Internal Error" From: Danny Nicholas B. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Back to top Display posts from previous: All Posts1 Day7 Days2 Weeks1

Sip 2.0 500 Server Internal Error Audiocodes

mberlant, Jul 9, 2006 #2 bbbeavis "name your trunk i2telecom.com" Thanks, I needed that I'm registered now, too. http://lists.digium.com/pipermail/asterisk-users/2010-August/252267.html Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 3 6. Sip 500 Internal Server Error Cisco Asterisk Forums Please hold while I try that extension. Sip 500 Internal Server Error Avaya So, no problems arise from this?

I'll try it and see if it works and post back. http://colvertgroup.com/500-internal/iis-debug-500-internal-server-error.php Comments have been locked on this page! UA (phone), gateway or other hardware/software involved: Polycom IP 335 & 450, FreePBX (Asterisk 1.8.5) 5. So, no problems arise from this? Cisco Cube Sip/2.0 500 Internal Server Error

Get always error message 500. (it works ok with their own software client phone. Idefisk Tools Tutorials Reviews VoIP Providers Archives AsteriskGuru ArchivesMailing List Archives FAQ Search Memberlist Usergroups Register Profile Log in to check your private messages Log in [Asterisk-Users] Internal Server Error Neither with Asterisk nor with a softphone. http://colvertgroup.com/500-internal/iis-6-asp-500-internal-server-error.php SJPhone complains about cannot negotiate codec.

Rates are different when using a SIP client. 480 Temporarily Unavailable Strange... Currently 4.13/512345 Rating: 4.1/5 (24 votes cast) Retrieved from "http://docwiki.cisco.com/wiki/SIP_Troubleshooting:_SIP_Calls_Receives_500_Internal_Server_Error_%22Routing_Failed%22_Event" Category: Unified CVP, Release 7.0(2) Views Page Leave a Comment View Source History Personal tools Log in Navigation Main Page Recent

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Reply URL 0 Arnas ● 3 years ago Ha, only one codec worked! Looking out for solutions. BTW, have you found a way to make the polycoms keep the volume settings (handset, ringers and speaker) after they have been restarted or rebooted? |-----Original Message----- |From: [email protected] |[mailto:[email protected]] On Any clues?

Doesn't seem to cause problems, but annoying just the same. It registers and I see the call hitting the CLI (turned on debug also). It only persist in wifi mode. check over here Not sure what to post to diagnose; however, I noticed this after the call disconnects: asterisk1*CLI> <-- SIP read from 206.165.50.116:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.0.122:5060;branch=z9hG4bK7b3a9747;rport=5060 From: ;tag=as638b51fb

All Rights Reserved. Re: SIP response 500 "Server Internal Error" [Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index] Subject: Re: SIP response 500 "Server Internal Error" From: asterisk What codec are you using? Please contact your supplier on this matter.Kind regards,Customer service For me it's difficult to believe that it is a hardware problem, because when I use other betamax providers such as voipbuster Quote: BTW, have you found a way to make the polycoms keep the volume settings (handset, ringers and speaker) after they have been restarted or rebooted?

That's good to know. Im in México.I have a public IP so I think this error isnt a nat problem.I had a similar problem with another provider in México and I configured the asterisk with Skip to content Wiki Blog Forums Mailing Lists Contact Us Advanced search Forums have moved to https://community.asterisk.org Board index ‹ Asterisk ‹ Asterisk Support RSS RSS Change font size FAQ SIP Stay logged in VoIP Forum Home Forums > General Service Provider Forums > Other Providers > Home All Content Residential Voip Business Voip Hardware Mobile Voip News Spotlight How to Forums

or use dial plan at brekeke sip server to forward subscribe requests to correct servers which handle these requests. Second, it looks like the error has something to do with turning the voicemail light on/off on the polycom phones. Doesn't seem to cause |problems, but annoying just the same. | |Doug | |-- |Ben Franklin quote: | |"Those who would give up Essential Liberty to purchase a |little Temporary Safety, Thanks!500 errors are server side errors that are proxied to Bria from a remote location.

Possible Cause A SIP Trunk is not configured to receive or send calls to the IP Address from the source of the SIP INVITE event. henkoegema Oldsterisk Posts: 210Joined: Sun May 14, 2006 2:52 am E-mail henkoegema Top by srl100 » Tue Dec 18, 2007 7:55 am Enable sip debugging and post your logs. SIP Debug shows (on FreepBX Server) : <--- SIP read from UDP:10.130.33.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.140.1.13:5060;branch=z9hG4bK187da513 From: "Unknown" ;tag=as2b6117f5 To: Call-ID: [email protected]:5060 CSeq: 102 OPTIONS Server: Brekeke I get "Internal server error (500)" message.

Reply URL 0 Arnas ● 3 years ago but i can call in 3G mode. I'm not on the same phone system, |so don't have daily interaction with it. | |> BTW, have you found a way to make the polycoms keep the volume |> settings henkoegema Oldsterisk Posts: 210Joined: Sun May 14, 2006 2:52 am E-mail henkoegema Top by fractalspace » Tue Dec 25, 2007 8:16 pm Just so you know, the advertized rates are I did not put useragent under [general] but under [voipraider-out] type=peer username=hoegema1946 secret=mysecret host=sip.voipraider.com realm=voipraider.com fromdomain=sip.voipraider.com fromuser=00324763788xx useragent=VoipRaider4.01build476 context=default canreinvite=no insecure=invite qualify=300 nat=yes port=5060 dtmfmode=inband disallow=all allow=alaw allow=ulaw According to their

So I am (nearly) convinced the the error is not coming from my site.