Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]----sip----[asterisk2]----PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo
On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote:> I must use sip, cos we'll use cisco rtp header-compression to save > bandwidth. > > Could you tell me the best way to send calls from asterisk1 to > asterisk2, since I cannot use IAX trunking?Maybe I'm way off base here, but I'm pretty sure that IAX2 trunking will save you more bandwidth than rtp header compression, at least if you've got multiple calls going between the two servers... (Then again, I might be a little biased, since IAX2 trunking was my idea.) Jared Smith
Why not use IAX2 trunking you can accomplish the same results with .. I tried SIP to SIP with asterisk you must do it without passwords. bkw On Mon, 5 Jan 2004, Eduardo Goncalves wrote:> Hi list, > > I have to connect two asterisk box, in this scenario: > > [asterisk1]----sip----[asterisk2]----PSTN > > I must use sip, cos we'll use cisco rtp header-compression to save > bandwidth. > > Could you tell me the best way to send calls from asterisk1 to > asterisk2, since I cannot use IAX trunking? > > Thanks in advance > Eduardo > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
On Mon, 2004-01-05 at 10:24, Eduardo Goncalves wrote:> Hi list, > > I have to connect two asterisk box, in this scenario: > > [asterisk1]----sip----[asterisk2]----PSTN > > I must use sip, cos we'll use cisco rtp header-compression to save > bandwidth.Will rtp header compression necessarily be better than full removal of IP headers? I'm betting rtp headers are already quite small and any compression there is minimal with respect to how IAX trunking combines what would otherwise be many packets into a single packet thus removing about 40 bytes per packet per call over 1.> Could you tell me the best way to send calls from asterisk1 to > asterisk2, since I cannot use IAX trunking?Sounds like you have already made decision to go down a path. Are you truely willing to accept better approaches? -- Steven Critchfield <critch@basesys.com>
I'm trying to setup SIP trunking between 2 asterisk servers. Eventually
there may be up to 5 servers linked together depending on the growth
needed. I have IAX2 trunking working, but I want both.
For simplicity, I have named the two servers, alpha and beta. Extension
7100 is a Polycom IP600 on alpha and extension 7300 is using kphone on
beta. Both ae using SIP. Below are the relvenant parts of
extensions.conf and sip.conf.
;# server alpha
; extensions.conf
[staff]
extension => 7100,Dial(SIP/7100)
extension => 7300,Goto(siptrunk,7300,1)
[siptrunk]
include => siptrunk-beta
[siptrunk-beta]
exten => _73XX,1,Dial(SIP/siptrunk-peer/${EXTEN})
; in sip.conf
[siptrunk-peer]
type=peer
username=siptrunk-peer
secret=password
host=<beta's IP address>
[siptrunk-user]
type=user
username=siptrunk-user
secret=password
host=<beta's IP address>
;# server beta
; extensions.conf
[staff]
extension => 7300,Dial(SIP/7300)
extension => 7100,Goto(siptrunk,7100,1)
[siptrunk]
include => siptrunk-alpha
[siptrunk-alpha]
exten => _71XX,1,Dial(SIP/siptrunk-peer/${EXTEN})
; in sip.conf
[siptrunk-peer]
type=peer
username=siptrunk-peer
secret=password
host=<alpha's IP address>
[siptrunk-user]
type=user
username=siptrunk-user
secret=password
host=<alpha's IP address>
When dialing 7300 from alpha, I get the following:
-- Executing Dial("SIP/7000-e924",
"SIP/siptrunk-peer/7300") in new
stack
-- Called siptrunk-peer/7300
Jun 13 11:10:47 WARNING[18099]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 45a607bc5f66cfb363f2cc565b85fa29@<alpha's IP
address> for seqno 102 (Critical Request)
== No one is available to answer at this time
Any ideas?
Hello, I am attempting to figure out how to set up SIP trunking, between my company and our SIP provider. This is an expermintal project at this time. The SIP provider gave us a Signalling IP address and two Media IP addresses. We supplied them with the IP address of our Asterisk box. When asked what our Usernames and Passwords would be we were told that they were not needed for a SIP trunk. We can use what they call SIP lines that use username/password however because of tarrifing the lines cost more per month than a trunk I have been successfull in making a SIP Line work, but have no idea where to start with a SIP Trunk. We sill be using DID numbers on the SIP trunks. Has anyone had any experiance with this type of configuration an example would be extrememly helpfull. I have search the internet for help and I may have seen a solution but just was not certain what I was looking at, or how to implement as everything that I have seen user a Username/Password combo. Thank you in advance. Thanks Steven __________________________________________________ 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/20060602/9b6d5b95/attachment.htm
It should work as is, just make usre that you have an extension defined (or a catch all) for every DID you have with the provider so that incoming works. On 6/2/06, Steven Haldeman <haldeman_79@yahoo.com> wrote:> > Hello, > > I am attempting to figure out how to set up SIP trunking, between my company > and our SIP provider. This is an expermintal project at this time. The SIP > provider gave us a Signalling IP address and two Media IP addresses. We > supplied them with the IP address of our Asterisk box. When asked what our > Usernames and Passwords would be we were told that they were not needed for > a SIP trunk. We can use what they call SIP lines that use username/password > however because of tarrifing the lines cost more per month than a trunk I > have been successfull in making a SIP Line work, but have no idea where to > start with a SIP Trunk. We sill be using DID numbers on the SIP trunks. > Has anyone had any experiance with this type of configuration an example > would be extrememly helpfull. > > I have search the internet for help and I may have seen a solution but just > was not certain what I was looking at, or how to implement as everything > that I have seen user a Username/Password combo. > > Thank you in advance. > > Thanks > > Steven > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Thank you for your response.
All that I get when I dial in is all circutes are busy and when I dial out 503
errors. Here are my configs. Any ideas would be greatly appreciated. The
provider is using a Tekelec 9000 Class 5 switch if that is any help.
The provider sat us up two accounts one that they are calling a line that uses
authentication usernames and passwords as though we were terminating into a SIP
phone. This account works but is not the prefered solution to our problem. The
other account is what the provider calls a trunk account and it does not use
usernames and passwords. This seems to be the solution for the provider and us.
Thank you,
Steven
sip.conf
[inbound-trunk]
type=friend
context=incoming
insecure=very
host=xxx.xxx.xxx.xxx
outboundproxy=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
defaultip=xxx.xxx.xxx.xxx
disallow=all
allow=ulaw
nat=yes
canreinvite=yes
qualify=yes
extension.conf
[incoming]
exten => NXXNXXXXXX,1,Answer()
exten => NXXNXXXXXX,2,Background(greeting)
exten => NXXNXXXXXX,3,SayDigits(${CALLERIDNUM})
exten => NXXNXXXXXX,4,Dial(SIP/steven)
exten => NXXNXXXXXX,5,Hangup()
C F <shmaltz@gmail.com> wrote:
It should work as is, just make usre that you have an extension
defined (or a catch all) for every DID you have with the provider so
that incoming works.
On 6/2/06, Steven Haldeman wrote:>
> Hello,
>
> I am attempting to figure out how to set up SIP trunking, between my
company
> and our SIP provider. This is an expermintal project at this time. The SIP
> provider gave us a Signalling IP address and two Media IP addresses. We
> supplied them with the IP address of our Asterisk box. When asked what our
> Usernames and Passwords would be we were told that they were not needed for
> a SIP trunk. We can use what they call SIP lines that use username/password
> however because of tarrifing the lines cost more per month than a trunk I
> have been successfull in making a SIP Line work, but have no idea where to
> start with a SIP Trunk. We sill be using DID numbers on the SIP trunks.
> Has anyone had any experiance with this type of configuration an example
> would be extrememly helpfull.
>
> I have search the internet for help and I may have seen a solution but just
> was not certain what I was looking at, or how to implement as everything
> that I have seen user a Username/Password combo.
>
> Thank you in advance.
>
> Thanks
>
> Steven
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
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I also forgot to mention that our provider gave use three different IP
addresses. One IP address for "Signalling" and two addresses for
"Media." If this helps any.
Thank you,
Steven
Steven Haldeman <haldeman_79@yahoo.com> wrote:
Thank you for your response.
All that I get when I dial in is all circutes are busy and when I dial out 503
errors. Here are my configs. Any ideas would be greatly appreciated. The
provider is using a Tekelec 9000 Class 5 switch if that is any help.
The provider sat us up two accounts one that they are calling a line that uses
authentication usernames and passwords as though we were terminating into a SIP
phone. This account works but is not the prefered solution to our problem. The
other account is what the provider calls a trunk account and it does not use
usernames and passwords. This seems to be the solution for the provider and us.
Thank you,
Steven
sip.conf
[inbound-trunk]
type=friend
context=incoming
insecure=very
host=xxx.xxx.xxx.xxx
outboundproxy=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
defaultip=xxx.xxx.xxx.xxx
disallow=all
allow=ulaw
nat=yes
canreinvite=yes
qualify=yes
extension.conf
[incoming]
exten => NXXNXXXXXX,1,Answer()
exten => NXXNXXXXXX,2,Background(greeting)
exten => NXXNXXXXXX,3,SayDigits(${CALLERIDNUM})
exten => NXXNXXXXXX,4,Dial(SIP/steven)
exten => NXXNXXXXXX,5,Hangup()
C F <shmaltz@gmail.com> wrote:
It should work as is, just make usre that you have an extension
defined (or a catch all) for every DID you have with the provider so
that incoming works.
On 6/2/06, Steven Haldeman wrote:>
> Hello,
>
> I am attempting to figure out how to set up SIP trunking, between my
company
> and our SIP provider. This is an expermintal project at this time. The SIP
> provider gave us a Signalling IP address and two Media IP addresses. We
> supplied them with the IP address of our Asterisk box. When asked what our
> Usernames and Passwords would be we were told that they were not needed for
> a SIP trunk. We can use what they call SIP lines that use username/password
> however because of tarrifing the lines cost more per month than a trunk I
> have been successfull in making a SIP Line work, but have no idea where to
> start with a SIP Trunk. We sill be using DID numbers on the SIP trunks.
> Has anyone had any experiance with this type of configuration an example
> would be extrememly helpfull.
>
> I have search the internet for help and I may have seen a solution but just
> was not certain what I was looking at, or how to implement as everything
> that I have seen user a Username/Password combo.
>
> Thank you in advance.
>
> Thanks
>
> Steven
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com _______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
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