Hi List; I established an SIP IP Trunk between Asterisk and another softswitch (asterisk registered on the softswitch successfully) and I saw this on the softswitch.
Pablo Allietti
2007-Oct-26 14:51 UTC
[asterisk-users] Everyone is busy/congested: IP Trunk
On Fri, Oct 26, 2007 at 06:55:12AM -0700, bilal ghayyad wrote:> 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 [9617565116 at EgyptInternationalVoIP:1] > Dial("SIP/EgyptOeratorSIP-09f9bed0", > "SIP/9617565116 at EgyptAlooNet") 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! Mail has the best spam protection around > http://mail.yahoo.com > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users---end quoted text--- -- .- Pablo Allietti E-mail: pablo at lacnic.net | LACNIC Phone : +598 2 6042222 | http://LACNIC.NET
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 tobe> via this softswitch (ofcourse, the softphone willsend> for asterisk and asterisk should route to the > softswitch based on the extensions.conf > configurations. > > But, always I receive this message (and the calldoes> not even reach to the softswitch, it is not sended > from Asterisk to the softswitch): > > Executing [9617565116 at EgyptInternationalVoIP:1] > Dial("SIP/EgyptOeratorSIP-09f9bed0", > "SIP/9617565116 at EgyptAlooNet") 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 issuch> 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! Mail has the best spam protection around http://mail.yahoo.com
On 10/27/07, bilal ghayyad <bilmar_gh at yahoo.com> 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.Hi i see 2 causes 1. it could be Dialplan issue ( check how the provider accept the call, 1 or just USA number) 2 provider blocked account check network trace to get more info ngrep should be the ideal tool to check the errors in network trace ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071029/d9430cc7/attachment.htm
joakimsen at gmail.com
2007-Oct-29 21:54 UTC
[asterisk-users] Everyone is busy/congested: IP Trunk
No: register => abc:123 at xyz [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 <bilmar_gh at yahoo.com> 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 [9617565116 at EgyptInternationalVoIP:1] > > Dial("SIP/EgyptOeratorSIP-09f9bed0", > > "SIP/9617565116 at EgyptAlooNet") 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! Mail has the best spam protection around > http://mail.yahoo.com > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Gabriel Natale
2007-Oct-30 12:43 UTC
[asterisk-users] Everyone is busy/congested: IP Trunk
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: <joakimsen at gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: Monday, October 29, 2007 6:54 PM Subject: Re: [asterisk-users] Everyone is busy/congested: IP Trunk> No: > > register => abc:123 at xyz > > [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 <bilmar_gh at yahoo.com> 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 [9617565116 at EgyptInternationalVoIP:1] >> > Dial("SIP/EgyptOeratorSIP-09f9bed0", >> > "SIP/9617565116 at EgyptAlooNet") 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! Mail has the best spam protection around >> http://mail.yahoo.com >> >> _______________________________________________ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
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: <joakimsen at gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: Monday, October 29, 2007 6:54 PM Subject: Re: [asterisk-users] Everyone is busy/congested: IP Trunk> No: > > register => abc:123 at xyz > > [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 <bilmar_gh at yahoo.com>wrote:>> Hi Pablo; >> >> How the IP address will be wrong, and asterisk ableto>> 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 Asteriskand>> > another softswitch (asterisk registered on the >> > softswitch successfully) and I saw this on the >> > softswitch. >> > >> > >From firefly softphone, I was need to do a callto>> 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 notsended>> > from Asterisk to the softswitch): >> > >> > Executing [9617565116 at EgyptInternationalVoIP:1] >> > Dial("SIP/EgyptOeratorSIP-09f9bed0", >> > "SIP/9617565116 at EgyptAlooNet") 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 numberof>> > 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 codecmatter,>> > then call should be send to the softswitch, andthe>> > softswitch will gives an error related to thecodec>> > missmatch. >> > >> > Any help? >> > >> > Regards >> > Bilal Ghayad__________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Baji Panchumarti
2007-Nov-06 17:26 UTC
[asterisk-users] Everyone is busy/congested: IP Trunk
after a copious loss of follicles :-), I finally got outbound working.
Basically the channel statement in the call file needs to have the
number to be called. For eg., in test.call format the statement
as follows :
Channel: SIP/3012345678@<your-sip-provider>
And there is no need for a DIAL statement in extensions.conf
unless you need to dial an additional number / extension.
Then in sip.conf you need a para that matches <your-sip-provider>
with the relevant auth info.
These two wiki pages, they were very helpful in figuring out a
solution to the problem :
http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out
http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message
hth,
-baji.
--
On Oct 30, 2007 8:43 AM, Gabriel Natale wrote:
> 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
Vivek Shrivastava
2007-Nov-07 00:46 UTC
[asterisk-users] Everyone is busy/congested: IP Trunk
yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti <baji.panchumarti at gmail.com> wrote:> > after a copious loss of follicles :-), I finally got outbound working. > > Basically the channel statement in the call file needs to have the > number to be called. For eg., in test.call format the statement > as follows : > > Channel: SIP/3012345678@<your-sip-provider> > > And there is no need for a DIAL statement in extensions.conf > unless you need to dial an additional number / extension. > > Then in sip.conf you need a para that matches <your-sip-provider> > with the relevant auth info. > > These two wiki pages, they were very helpful in figuring out a > solution to the problem : > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message > > hth, > > -baji. > > -- > > On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: > > > 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 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071106/0b506444/attachment.htm
Hi Friends; Actually I would appreciate if Vivek can advise if the VPN resolved the RTP packets in the SIP Trunk between Asterisk and another softswitch? In other words, openvpn helpful in NAT cases in what exactly? As without VPN, I was able to establish a call but without voice or with complete noise (nothing understood) :) - So if NAT resolve this issue for the SIP Trunk, then I can proceed forward, as really now I do not have any other attempt to try.
Vivek Shrivastava
2007-Nov-10 11:41 UTC
[asterisk-users] Everyone is busy/congested: IP Trunk
Well, unfortunately i did not dig much into "why/how it worked" with openvpn, but it did work for me with default setup.I think you may need to set constant ports instead of random ports. Thanks, Vivek On 11/9/07, bilal ghayyad <bilmar_gh at yahoo.com> wrote:> > Hi Friends; > > Actually I would appreciate if Vivek can advise if the > VPN resolved the RTP packets in the SIP Trunk between > Asterisk and another softswitch? In other words, > openvpn helpful in NAT cases in what exactly? As > without VPN, I was able to establish a call but > without voice or with complete noise (nothing > understood) :) - So if NAT resolve this issue for the > SIP Trunk, then I can proceed forward, as really now I > do not have any other attempt to try. > > From the other side, I think that baji is talking > about something else than the IP Trunk, he is talking > about outbound (which is related to using an > application to run an outside call, which is used > usually in campaign in contact centers and so on), I > think nthis case differs that placing a calls via IP > Trunk or even outside call but the caller who will do > it (and not the application). > > Lastly, Mr. Amit helped me when he gave me a > configuration to be done for the SIP Trunk, as in his > method, I did not register on the softswitch, I send > directly, and the connectioned succeed, but as I said: > with complete voice (actually nothing understood, i > feel it is complete RTP situation), the test was by > letting Asterisk behind NAT (private IP) and sending > to a softswitch in anther country has a public IP > address. Is it NAT issue, so VPN can resolve? > > Note: anyone knows if h323 works better in the IP > trunk? > > Regards > Bilal > > ---------------------------------- > yeah i found openvpn helpful in NAT cases. > > -Vivek > > > On 11/6/07, Baji Panchumarti > <baji.panchumarti at gmail.com> wrote: > > > > after a copious loss of follicles :-), I finally got > outbound > working. > > > > Basically the channel statement in the call file > needs to have the > > number to be called. For eg., in test.call format > the statement > > as follows : > > > > Channel: SIP/3012345678@<your-sip-provider> > > > > And there is no need for a DIAL statement in > extensions.conf > > unless you need to dial an additional number / > extension. > > > > Then in sip.conf you need a para that matches > <your-sip-provider> > > with the relevant auth info. > > > > These two wiki pages, they were very helpful in > figuring out a > > solution to the problem : > > > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out > > > > > > > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message > > > > hth, > > > > -baji. > > > > -- > > > > On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: > > > > > 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 > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071110/ec2b51bc/attachment.htm