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Sip Error 480 No Routes Found

Dain Bramaged (Avaya Search tool http://tinyurl.com/bas1234 ) ______________________________________ RE: SIP Configuration johnromani (IS/IT--Management) (OP) 28 Jan 14 08:35 Heçlo Bas1234, i tried to change from LAN1 to LAN2 and any others SIP 923 / No DNS results / Service or option unavailable It is usually shown when: the server information is not provided correctly; the transport type is not selected as necessary; I put on the SIP URI the complete Number (551149494949) (these are not the real numbers) Regards, Cesar Romani Avaya ACSS | APSS (SME/Data)| ACIS (Data) Comptia Convergence + RE: SIP I'll ask my network provider if they received >> these message tomorrow morning. http://idroprofessional.com/sip-error/sip-error-503.html

Here's the output. Thanks! we need to send the first 6 digits including the 6. lol I never thought that we have to change the ITSP Domain...

Here's Why Members Love Tek-Tips Forums: Talk To Other Members Notification Of Responses To Questions Favorite Forums One Click Access Keyword Search Of All Posts, And More... Do you have an outbound accept template on the SIP trunk that allows (matches) the call patterns the user is dialing outbound?You could run a "debug voice summary", and see what Dowble check if the audio codecs that you are using are supported.

  • session target ipv4:10.0.1.62 incoming called-number .T voice-class codec 1  voice-class sip profiles 1 dtmf-relay rtp-nte fax-relay ecm disable no fax-relay sg3-to-g3 fax protocol pass-through g711ulaw!dial-peer voice 11 voip description -------------[SERVICOS] translation-profile
  • That will narrow things down to either >> an Asterisk configuration or a network routing issue. >> >> There is not really a caller, I'm trying to use Asterisk as an
  • I replaced numbers and letters with **** for privacy reasons.

After talking to the provider and doing some traces, they determined I needed to change the IP of the SIP server on their network, in my 7060. Cesar Romani Avaya ACSS | APSS (SME/Data)| ACIS (Data) Comptia Convergence + RE: SIP Configuration amriddle01 (Programmer) 2 Feb 14 09:11 I think you need a better and less demanding provider, It should be the "domain" of the SIP message; one the UAS has authorized. By routing calls based on this information, rather than the standard dialed number identification service (DNIS) information, more flexible and user-centered networks can be created.

More about audio codecs. That works fine. Like Show 0 Likes (0) Actions Re: Outbound through SIP trunk DID gouc Aug 30, 2013 1:02 PM (in response to gerkhs) You may want to look at this document. http://sipforum.org/pipermail/discussion/2010-September/007827.html Is this CUBE?

Calls fail with SIP error 503, I-SUP errors 34 or 38:If our platform replies back with 503 it usually means the gateway trying to process the call can't due to "issues", Hello. … error-passthru no update-callerid early-offer forced Scanner Error Samsung 4300 The Canon imageFormula DR-C225W document scanner can connect by Wi-Fi for network scanning … That makes it tied, within the We cannot drop the 6 since all cell phones need a 6 in front where the trunk is located. Resources Join | Indeed Jobs | Advertise Copyright © 1998-2016 ENGINEERING.com, Inc.

For example, if the public number is 645-5111, and you are dialing 64551112 and only sending 455-1112, it won't match. I actually need the call to go out trunk T01 and not T03. Codec payload      : 255 (tx), 255 (rx)Negotiated Dtmf-relay    : 0Dtmf-relay Payload       : 0 (tx), 0 (rx)Source IP Address (Media): 10.55.52.25Source IP Port    (Media): 22990Destn  IP Address (Media):  - Destn  IP Port    it works!!!

This indicates that the request(call, register, etc) does not reach the VoIP server or the response does not reach Zoiper. have a peek at these guys Report the issue to your provider for more assistance. Yes NoSend feedback Sorry we couldn't be helpful. Error 102 Error 102 reffers to SIP 408 "Request timeout".

One of these is the actual response by the PBX/VoIP server to your (registration) request. By joining you are opting in to receive e-mail. Then on the SIP Line>Transport tab, populate which LAN to use for Network Topology Info. http://idroprofessional.com/sip-error/error-code-480-att.html Due to this the Service Provider rejects the call.

Hello. … error-passthru no update-callerid early-offer forced SIP Trunk /// SIP/2.0 480 No Routes Found. SIP 409 / Temporary failure There is a temporary network failure. Here's the debug voice summary output.

This will set the Public IP Address and Firewall/NAT Type for you.

when the destination does not wish to participate in the call, or cannot do so. You can not post a blank message. If the issue remains unaffected or if it did not exist before (appeared recently), you should contact your ITSP or network administrator to check if any changes had been introduced to More … Sms Error Code 3 The Witcher 3 PS4 Error Code CE-34878-0 Workaround Given, New Video Explores Its Beautiful World Bookmark the permalink.

Are the numbers you are dialing reachable via other trunks? On a 7060, I got 2 FXO lines that are used by several users. Cesar Romani Avaya ACSS | APSS (SME/Data)| ACIS (Data) Comptia Convergence + RE: SIP Configuration Bas1234 (TechnicalUser) 27 Jan 14 17:37 On lan 1 or 2 fill in the stun part. http://idroprofessional.com/sip-error/sip-error-503-xlite.html Depending on the server setup you may need to use a different outbound proxy.Contact your system administrator or VoIP provider for more assistance.

session protocol sipv2 session target ipv4:172.26.154.70 voice-class codec 1  voice-class sip profiles 1 dtmf-relay sip-notify fax-relay ecm disable fax rate disable fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback We’ll also assume you agree to the way we use cookies and are ok with it as described in our Privacy Policy, unless you choose to disable them altogether through your If so can you post the config?Chris See More 1 2 3 4 5 Overall Rating: 0 (0 ratings) Log in or register to post comments Leonardo Tadeu Tue, 05/15/2012 - Because on SIP URI tab, all the configs (by credential, internal data,..) is going by 192.168.x.x; Regards, Cesar Romani Avaya ACSS | APSS (SME/Data)| ACIS (Data) Comptia Convergence + RE: SIP

SIP 480 / Temporarily unavailable / No user response The account you are trying to dial appears to be unavailable. Now if only I could remember what I did... Here's the output. Join UsClose HomeInboxActivityLinksTechnical DocumentsSoftware NotificationsFeature RequestsGetting Started VideoFeedbackBrowseContentPeoplePlacesLog inSearch All Places > NetVanta > NetVanta 7000 Series > Discussions Please enter a title.

Make sure that the proper account is selected. Altering STUN and RPort for the affected account could help. I'm assuming that is a 7-digit pattern they are dialing for local calls? Search New support ticket Check ticket status 0115 88 000 88 (option3) Solution home General SIP SIP Error Codes & SIP Trunk Troubleshooting Modified on: Mon, 16 Mar, 2015 at 11:17

I need to know how a change the ip address in "from" field. Close Reply To This Thread Posting in the Tek-Tips forums is a member-only feature. If STUN doesn't do well try manually to change it which is a pain but there are only 4 or 5 settings there and it is not open Internet usually so