[asterisk-users] multiple sip trunks with the same ITSP
HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name Example register=myaccou...@sip.myitsp.com/line1 register=myaccou...@sip.myitsp.com/line2 [line1] type=peer username=myaccount1 host=sip.myitsp.com [line2] type=peer username=myaccount2 host=sip.myitsp.com If sip.myitsp.com directs a call to asterisk with a request line of: INVITE line1@mybindaddr SIP/2.0 then it is matched to the line2 peer whereas it would probably be better matched to the line1 peer -- Best regards Antony моб (066) 919-75-33 моб (063) 656-43-40 satski...@gmail.com mail%3asatski...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP trunks between same pair of asterisk box
If it is just matter of billing you can pass billing related info in additional SIP headers on single trunk. If you must need multiple trunk you can add multiple IPs of different subnet class to both interfaces and configure asterisk to listen of all IPs. Then use one trunk per IP Subnet class. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, July 21, 2011 3:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple SIP trunks between same pair of asterisk box Hello, for billing purpose between a multitenant asterisk box and another asterisk, I am in the need to maintain multiple SIP trunks between them. Usually I use insecure=invite,port but I had to remove or the trunks will be selected based on IP address and not with username/password. I had to use the fromuser option or asterisk will try to authenticate the call using the CID and not the username, but this break the outbound CID of the client. Both are asterisk 1.6 Is there any other solution from multiple SIP trunks between two asterisk boxes? Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple SIP trunks between same pair of asterisk box
Hello, for billing purpose between a multitenant asterisk box and another asterisk, I am in the need to maintain multiple SIP trunks between them. Usually I use insecure=invite,port but I had to remove or the trunks will be selected based on IP address and not with username/password. I had to use the fromuser option or asterisk will try to authenticate the call using the CID and not the username, but this break the outbound CID of the client. Both are asterisk 1.6 Is there any other solution from multiple SIP trunks between two asterisk boxes? Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple sip trunks
I thought so- the fact the server has 20 different registry entries to 20 different account all at the same ITSP shouldn't matter? Can't see any DDI info in the SIP headers unfortunately :( John 2009/12/14 meetmecall i...@meetmecall.nl The easiest solution to deal with this is to have one context with different extensions for the different numbers and route the incoming calls from there. It should look something like this (not a tested piece of asterisk script, just an example to give the idea). Hope it helps :-) Erik de Wild [all_trunks] exten = 31592123456,1,Goto(trunk1,s,1) exten = 31592123457,1,Goto(trunk1,s,1) exten = 31592123458,1,Goto(trunk1,s,1) exten = 3159212,1,Goto(trunk2,s,1) exten = 31592123334,1,Goto(trunk2,s,1) exten = 31592123335,1,Goto(trunk2,s,1) On 14 dec 2009, at 10:39, Olle E. Johansson wrote: 11 dec 2009 kl. 23.21 skrev John Taylor: I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? This is an effect of the Asterisk architecture. We've had many discussions on how to change it, but right now the peer matching on IP/Port can't separate various instances from each other, since they all have the same IP/port. Asterisk simply goes for the first match, which happens to be the last entry with the IP/port in the sip.conf file. /Olle ___ -- 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 ___ -- 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
Re: [asterisk-users] multiple sip trunks
In that case, you're going to have to talk to your provider. They SHOULD be able to easily send the DID with the call... -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Taylor Sent: Tuesday, December 15, 2009 5:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] multiple sip trunks I thought so- the fact the server has 20 different registry entries to 20 different account all at the same ITSP shouldn't matter? Can't see any DDI info in the SIP headers unfortunately :( John 2009/12/14 meetmecall i...@meetmecall.nlmailto:i...@meetmecall.nl The easiest solution to deal with this is to have one context with different extensions for the different numbers and route the incoming calls from there. It should look something like this (not a tested piece of asterisk script, just an example to give the idea). Hope it helps :-) Erik de Wild [all_trunks] exten = 31592123456,1,Goto(trunk1,s,1) exten = 31592123457,1,Goto(trunk1,s,1) exten = 31592123458,1,Goto(trunk1,s,1) exten = 3159212,1,Goto(trunk2,s,1) exten = 31592123334,1,Goto(trunk2,s,1) exten = 31592123335,1,Goto(trunk2,s,1) On 14 dec 2009, at 10:39, Olle E. Johansson wrote: 11 dec 2009 kl. 23.21 skrev John Taylor: I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? This is an effect of the Asterisk architecture. We've had many discussions on how to change it, but right now the peer matching on IP/Port can't separate various instances from each other, since they all have the same IP/port. Asterisk simply goes for the first match, which happens to be the last entry with the IP/port in the sip.conf file. /Olle ___ -- 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 ___ -- 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
Re: [asterisk-users] multiple sip trunks
11 dec 2009 kl. 23.21 skrev John Taylor: I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? This is an effect of the Asterisk architecture. We've had many discussions on how to change it, but right now the peer matching on IP/Port can't separate various instances from each other, since they all have the same IP/port. Asterisk simply goes for the first match, which happens to be the last entry with the IP/port in the sip.conf file. /Olle ___ -- 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
Re: [asterisk-users] multiple sip trunks
snip
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
/snip
Your provider is probably sending the DID in the SIP header TO: field. This was
discussed on the list last week to at a reasonable level of detail but
generally speaking, you want to dump all of the calls into a context like
[FromSIP] and then have all calls parsed based on the to: field with something
like this:
(credit for this goes to someone at asterisk-info.org, but I didn't write down
who...)
[FromSIP]
;DIDs
exten = 888555,1,Dial(SIP/EXTENSION,10)
;parser
exten = i,1,Goto(FromSIP|s|1)
exten = s,1,Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
exten = s,n,Goto(FromSIP|${calldest:1}|1)
Then you can set up an exten for each incoming DID that will handle the calls
directly within this same context. Turn on sip debugging and high verbosity at
the cli to help yourself see what's going on with this...
-Dave
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Re: [asterisk-users] multiple sip trunks
The easiest solution to deal with this is to have one context with different extensions for the different numbers and route the incoming calls from there. It should look something like this (not a tested piece of asterisk script, just an example to give the idea). Hope it helps :-) Erik de Wild [all_trunks] exten = 31592123456,1,Goto(trunk1,s,1) exten = 31592123457,1,Goto(trunk1,s,1) exten = 31592123458,1,Goto(trunk1,s,1) exten = 3159212,1,Goto(trunk2,s,1) exten = 31592123334,1,Goto(trunk2,s,1) exten = 31592123335,1,Goto(trunk2,s,1) On 14 dec 2009, at 10:39, Olle E. Johansson wrote: 11 dec 2009 kl. 23.21 skrev John Taylor: I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? This is an effect of the Asterisk architecture. We've had many discussions on how to change it, but right now the peer matching on IP/Port can't separate various instances from each other, since they all have the same IP/port. Asterisk simply goes for the first match, which happens to be the last entry with the IP/port in the sip.conf file. /Olle ___ -- 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
Re: [asterisk-users] multiple sip trunks
Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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
Re: [asterisk-users] multiple sip trunks
On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote: Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? yes, Asterisk does support multiple registry entries... if it didn't ... it would be just a crippled sip endpoint lets say more ... Asterisk can do whatever you want it to do (within reason and technical boundaries); just code it in or request a feature Martin Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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 ___ -- 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
Re: [asterisk-users] multiple sip trunks
I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf Thanks John 2009/12/11 Martin asteriskl...@callthem.info: On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote: Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? yes, Asterisk does support multiple registry entries... if it didn't ... it would be just a crippled sip endpoint lets say more ... Asterisk can do whatever you want it to do (within reason and technical boundaries); just code it in or request a feature Martin Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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 ___ -- 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
Re: [asterisk-users] multiple sip trunks
I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf No. SIP uses authentication (well, I guess you can not use authentication). Asterisk (and almost any SIP gateway) will correctly match the call to the trunk based on the authentication. Even if you didn't send any authentication info, asterisk will try to match the call as a guest call. It is common practice to not allow unauthenticated SIP traffic. - Noah ___ -- 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
Re: [asterisk-users] multiple sip trunks
I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? John 2009/12/11 Noah Miller noahisaacmil...@gmail.com: I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf No. SIP uses authentication (well, I guess you can not use authentication). Asterisk (and almost any SIP gateway) will correctly match the call to the trunk based on the authentication. Even if you didn't send any authentication info, asterisk will try to match the call as a guest call. It is common practice to not allow unauthenticated SIP traffic. - Noah ___ -- 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
[asterisk-users] multiple sip trunks
I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John ___ -- 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
Re: [asterisk-users] multiple sip trunks
- John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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
