Dear Amit; Special thanks for your greate help and support.
Sorry for delaying in reply, I was busy during this week. It worked with very poor and noise voice, and disconnect after around 5 seconds, but it worked in the direct mode (by using trustip=yes so Asterisk does not register on the softswitch). But maybe this voice problem was because of the network (I will explain my network situation). Before I explain my network situation, I would like to know why in registering mode (by using register => directive and letting asterisk registering on the softswitch), Asterisk was registering successfully but call was not arrive for the softswitch (does not know if Asterisk sent it or did not send it). The question: is there a kind of packets negotiation during the SIP registeration that determine the facility of call exchaning? The softswitch ables to receive calls from any SIP endpoint, why this does not do happen with Asterisk if Asterisk registered? But it receive and manipulate the calls if Asterisk work via trustip (without registeration)?!! Actually, when Asterisk was registering on the softswitch, I was see the registeration on the softswitch, but I did not see even the call attempt. Regarding to my network status (that might be the reason of having very poor and noise voice and disconnecting the line after around 5 second), actually the softswitch in public IP address and it is located in Germany, while the Asterisk in Kuwait and it is behind NAT (a private IP address), and the softphone also have a private IP address (in the same LAN with the Asterisk), so the softphone was registering on the Asterisk, when the softphone send the call for Asterisk then Asterisk was sending it for the the softswitch in Germany via the SIP Trunk. Do u think that because Asterisk Nated? In that case, do u think the VPN will resolve the problem (VPN between Asterisk network in Kuwait, and the Softswitch network in Germany)? Or there is a settings should be done? Regards Bilal ---------------------------- I have the same problem. I trying with more 4 SIP providers, the account is registering, receive inboud calls, but can`t make outbound calls for "congestion". Can be the out call id the problem? Thanks Gabriel ----- Original Message ----- From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Monday, October 29, 2007 6:54 PM Subject: Re: [asterisk-users] Everyone is busy/congested: IP Trunk > No: > > register => abc:[EMAIL PROTECTED] > > [peer] > host=zzz > > Its possible to make mistakes and typos you know. Maybe you can post > your config file and we can help you. > > On 10/26/07, bilal ghayyad <[EMAIL PROTECTED]> wrote: >> Hi Pablo; >> >> How the IP address will be wrong, and asterisk able to >> do registeration on the destination? >> >> If the IP address wrong, so I will not be able to >> register on that IP address. >> >> Regards >> Bilal >> >> > Hi List; >> >> >> Ip address to destination? >> >> Unable to create channel of type SIP (cause 3 - No >> route to destination) >> >> i think you have the wrong ip information >> >> >> >> > >> > I established an SIP IP Trunk between Asterisk and >> > another softswitch (asterisk registered on the >> > softswitch successfully) and I saw this on the >> > softswitch. >> > >> > >From firefly softphone, I was need to do a call to >> be >> > via this softswitch (ofcourse, the softphone will >> send >> > for asterisk and asterisk should route to the >> > softswitch based on the extensions.conf >> > configurations. >> > >> > But, always I receive this message (and the call >> does >> > not even reach to the softswitch, it is not sended >> > from Asterisk to the softswitch): >> > >> > Executing [EMAIL PROTECTED]:1] >> > Dial("SIP/EgyptOeratorSIP-09f9bed0", >> > "SIP/[EMAIL PROTECTED]") is new stack >> > >> > Unable to create channel of type SIP (cause 3 - No >> > route to destination) >> > >> > Everyone is busy/congested at this time (1:0/0/1) >> > >> > Anyone faced that? >> > >> > Is it related to a paramater that control number of >> > allowed channels per IP trunk? Maybe I have such >> > parameters is 0 ? I do not know even if there is >> such >> > parameter. >> > >> > At the softswitch, I do not see even any attempt >> > (nothing related to the dialed number), so why >> > Asterisk does not send the called number to the >> > softswitch and why asterisk assume there is not >> > available channel? >> > >> > The softphone codec is g729a and the softswitch >> > support such codec. Also, if it is a codec matter, >> > then call should be send to the softswitch, and the >> > softswitch will gives an error related to the codec >> > missmatch. >> > >> > Any help? >> > >> > Regards >> > Bilal Ghayad __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! 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