[SR-Users] Issue with Asterisk interconnection for VoiceMail

[Igor Potjevlesch]

Hello Daniel,

I tried and it works! Thank you.

So, as suggested, I write $du=$null before call RELAY route.

Regards,

Igor.

-----Message d'origine-----
De : Igor Potjevlesch [mailto:igor.potjevlesch at gmail.com] 
Envoyé : lundi 18 mai 2015 16:42
À : 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Users] Issue with Asterisk interconnection for VoiceMail

Hello Daniel,

I can try this. But there are cases where lookup is called and the
redirection to VoiceMail is working fine.

Could it be an issue with a missing "append_branch()" instruction?

Regards,

Igor.

-----Message d'origine-----
De : sr-users [mailto:sr-users-bounces at lists.sip-router.org] De la part de
Daniel Tryba Envoyé : vendredi 15 mai 2015 10:58 À :
sr-users at lists.sip-router.org Objet : Re: [SR-Users] Issue with Asterisk
interconnection for VoiceMail

On Friday 15 May 2015 10:30:45 Igor Potjevlesch wrote:
> Then, the request goes to RELAY. Here is the issue:
> sometimes, the request is forwarded to the IP of the UA (the one 
> initially
> contacted) instead of the IP of Asterisk.

Unset $du ($du=$null) when routing to voicemail.

It is set after a lookup():
http://www.kamailio.org/wiki/cookbooks/4.1.x/pseudovariables#du_-_destinatio
n_uri

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