<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:10pt"><div><br></div><div style="font-family: times new roman,new york,times,serif; font-size: 10pt;">Hello,<br>Thank you for your reply.<br>Ok it works; now i have an other issue:<br>My asterisk(A) communicate with an other asterisk(B) via SIP protocole, my user (U) communicate with asterisk(A) via IAX2 protocole; i tried as you said but unfortunatly it does not work; <br>here is my configuration :<br>sip.conf:<br>[general]<br>register => login:secret@domaine<br><br>[outbound]<br>type=peer<br>host=hostname<br>insecure=invite<br>nat=yes<br>qualify=yes<br>secret=secret<br>username=username<br>canreinvite=no<br>disallow=all<br>allow=ulaw<br>allow=gsm<br>allow=alaw<br><br>iax.conf<br>[500]<br>username = 500<br>secret = secret<br>context = outbound<br>type = friend <br>host = dynamic <br>disallow =
all <br>allow = gsm <br>allow = ulaw <br>allow = alaw <br>transfer = no <br><br><br>extension.conf<br>[outbound]<br>exten =>XXX.,1,Dial(SIP/login@outbound/${EXTEN})<br><br>But it does not work.<br>I'm not sure if the configuration is ok.<br>I showed the sip peers, it says that asterisk A is registred to asterisk B.<br><br>Any suggestion<br><br><br><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><font face="Tahoma" size="2"><hr size="1"><b><span style="font-weight: bold;">De :</span></b> khalid touati <khalidtouati@gmail.com><br><b><span style="font-weight: bold;">À :</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b><span style="font-weight: bold;">Envoyé le :</span></b> Jeu 8 juillet 2010, 7h 09min 18s<br><b><span style="font-weight: bold;">Objet :</span></b> Re: [asterisk-users] Communication IAX2
>SIP>IAX2<br></font><br><br><br><div class="gmail_quote">2010/7/8 Kyle Kienapfel <span dir="ltr"><<a rel="nofollow" ymailto="mailto:doctor.whom@gmail.com" target="_blank" href="mailto:doctor.whom@gmail.com">doctor.whom@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div><div></div><div class="h5">On Wed, Jul 7, 2010 at 10:34 AM, Adil Zaaraoui <<a rel="nofollow" ymailto="mailto:adilzeaaraoui@yahoo.fr" target="_blank" href="mailto:adilzeaaraoui@yahoo.fr">adilzeaaraoui@yahoo.fr</a>> wrote:<br>
> Dear list.<br>
><br>
> Is it possible to use both IAX2 and SIP protocole during a dial?<br>
><br>
> Illustration:<br>
><br>
> I have peer A communicate with my Asterisk using IAX2 protocole.<br>
> I have peer B communicate with my Asterisk using SIP protocole.<br>
><br>
> A and B are both registred to the same Asterisk.<br>
><br>
> So is it possible that peer A communicate with peer B and vice versa?<br>
><br>
> if yes how can i achieve that?<br>
><br>
> Best regards<br>
><br></div></div></blockquote></div><br>yes it is possible, all you have to to is use the Dial application according to what technology your peer is using, and asterisk do the translation for you automatically:<br><br>
exten = XXX,1,Dial(IAX2/A,20,) and exten = XXX,1,Dial(SIP/B,20,)<br><br>you can also use one Dial that try to reach this extension using both technologies.<br><br>exten = XXX,1,Dial(IAX2/A&SIP/A,,) (or if A=555 for ex: exten = 555,Dial(IAX2/${EXTEN}&SIP/${EXTEN},,)<br>
<br>you can find this and more on the future of telephony book.<br><br>Good luck!!<br><br>------------------<br>Abdullah<br>
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