SIP-account not working | Thirdlane.com

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hello,

we use a 25 user lizens for the pbxmanager 6.0.1.72. We have create a sipp-acount. Over the cli, the command "sip show registry" is the account ist registered. We have create a inbound route for this sip-account (trunk). We can not make a call from extern or from intern. The calls intern works fine.
When we make then configfiles manually (like HowTo from Astersik) its work fine. Where can i find a HowTo for the pbxmanager and for the german sip provider?

We need help.

Sorry my english is ....., a my german is better.

Thanks for help.

Frank

Submitted by eeman on Tue, 05/12/2009 Permalink

run 'sip show peers' and see if all the accounts appear correctly

Submitted by evendo on Wed, 05/13/2009 Permalink

the command 'sip show peers' show all internal registered sip phones (ok). The problems are the incomming or outgoing calls by a external sip-provider. By the command 'sip set debug ip xxxx' we can monitor the call from external (by sip provider) . The call is switch to the 'Contact: ' and the phone tone indicates busy.

Thanks for help.

Frank

Submitted by eeman on Wed, 05/13/2009 Permalink

are you saying your provider is incorrectly sending SIP headers placing the DID in the Contact: header? It sounds as though you are experiencing an interop problem between asterisk and your provider's platform. You can parse the sip header and redirect the call but if your outbound calls are failing its not going to be quite as easy of a fix, your going to have to alter the custom trunk headers etc. to add appropriate headers for the callerid info.

Submitted by evendo on Wed, 05/20/2009 Permalink

To check whether the SIP-phone providers may be the config files from the pbx manager removed and replaced by a separate configuration. With this configuration could easily be on the sip-provider will be phoned. Here is an excerpt from the trace:

-- Goto (macro-tl-dialout-base,dial-SIP,7)

-- Executing [dial-SIP@macro-tl-dialout-base:7] GotoIf("SIP/100-08204c20", "1?NoOpt") in new stack

-- Goto (macro-tl-dialout-base,dial-SIP,10)

-- Executing [dial-SIP@macro-tl-dialout-base:10] GotoIf("SIP/100-08204c20", "0?noarg") in new stack

-- Executing [dial-SIP@macro-tl-dialout-base:11] Dial("SIP/100-08204c20", "SIP/03055731xxxxx@SIPGATE0306098xxxx|60") in new stack

-- Called 0305573xxxx@SIPGATE0306098xxxx

[May 15 10:32:05] WARNING[3671]: chan_sip.c:12679 handle_response_invite: Received response: "Forbidden" from '"e-vendo" ;tag=as5a568568'

-- SIP/SIPGATE0306098xxxx-08208c28 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)

-- Executing [dial-SIP@macro-tl-dialout-base:12] Goto("SIP/100-08204c20", "dial-CONGESTION|1") in new stack

-- Goto (macro-tl-dialout-base,dial-CONGESTION,1)

-- Executing [dial-CONGESTION@macro-tl-dialout-base:1] Goto("SIP/100-08204c20", "next|1") in new stack

-- Goto (macro-tl-dialout-base,next,1)

-- Executing [next@macro-tl-dialout-base:1] Set("SIP/100-08204c20", "i=6") in new stack

-- Executing [next@macro-tl-dialout-base:2] Goto("SIP/100-08204c20", "onetrunk|1") in new stack

Thank for help

Frank

Submitted by eeman on Wed, 05/20/2009 Permalink

[May 15 10:32:05] WARNING[3671]: chan_sip.c:12679 handle_response_invite: Received response: "Forbidden" from '"e-vendo" ;tag=as5a568568'

as long as your upstream trunk is denying access you will not be able to call. You will need to contact your provider and supply sip captures.