[asterisk-users] SER/Asterisk interworking mailing list.
Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in duplicated effort and lack of specialised response. This is mainly due, I think, to the fact that detailed Asterisk experience - while common - is not a prerequisite for working with the SER products, while for Asterisk people SER can often be a next step in scalability and VoIP service delivery platform enhancement that they are just getting into. And so on. There's pollution in the respective discursive spaces; a lot of Asterisk people posting to the SER lists ask a lot of Asterisk-specific questions in addition to any they may have about SER which can be construed as potentially off-topic by some members, and the opposite is true on the Asterisk lists when detailed, involved discussion about SER occurs. We need to capture that discussion that exists at the overlap and is specifically concerned with making these two systems work together, requiring somewhat detailed and esoteric understanding of both and a community of user support and knowledge that focuses on both of these conceptual and product universes. Toward that end, I am hosting a new mailing list with this succinct purpose, if slightly unwieldy name, and encourage all interested to join. It is called 'SER-Asterisk-Interwork' and can be accessed for subscription here: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork The archives are available here: http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ You can post to the list at: [EMAIL PROTECTED] It's the same GNU Mailman stuff you are already used to. While it could be argued that this cross-product discussion is valuable to retain in both communities, I think there is considerable benefit to creating a specialised mailing list that focuses specifically on this integration path and the unique interoperation and configuration issues it creates. I think it would be good to get some of this discussion off of the SER and Asterisk-specific mailing lists where it has somewhat marginal relevance at times and refocus it. If you agree and are interested in this topic, you are invited to join the list. One last note: The SER/OpenSER community has been in a state of flux recently, with OpenSER undergoing a name change to Kamailio and subsequently seeing a fork. The incumbent Kamailio project is now in the process of merging with the original SER project. The choice of nomenclature for list is not meant to imply an endorsement of or affinity for the IPTel SER project per se. It is just that right now it serves the aim of terseness to use a common denominator, to refer to this family of projects as the SER ecosystem. Whether you are a SER, OpenSER, Kamailio, or OpenSIPS user, you are part of that SER ecosystem. That is why the list is named what it is. Thank you, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER/Asterisk interworking mailing list.
Good work, I am sure this will be endorsed by many and will be useful for lots of small VoIP user who are ready to expand. Only problem I have seen is that people who have done (deployed) this type of integration does not share complete solution mainly because of compititive disadvantage. But keeping the information at one place will definitely help. I am also working on a 'howto' on integrating Asterisk with Ser that will describe step by step instructions on the deployment of asterisk. I have tons of many things in my plate but targeting to finish within next week or so. -Jai Buy unmetered VoIP DID from DidForSale.com On Wed, Nov 5, 2008 at 9:04 AM, Alex Balashov [EMAIL PROTECTED]wrote: Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in duplicated effort and lack of specialised response. This is mainly due, I think, to the fact that detailed Asterisk experience - while common - is not a prerequisite for working with the SER products, while for Asterisk people SER can often be a next step in scalability and VoIP service delivery platform enhancement that they are just getting into. And so on. There's pollution in the respective discursive spaces; a lot of Asterisk people posting to the SER lists ask a lot of Asterisk-specific questions in addition to any they may have about SER which can be construed as potentially off-topic by some members, and the opposite is true on the Asterisk lists when detailed, involved discussion about SER occurs. We need to capture that discussion that exists at the overlap and is specifically concerned with making these two systems work together, requiring somewhat detailed and esoteric understanding of both and a community of user support and knowledge that focuses on both of these conceptual and product universes. Toward that end, I am hosting a new mailing list with this succinct purpose, if slightly unwieldy name, and encourage all interested to join. It is called 'SER-Asterisk-Interwork' and can be accessed for subscription here: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork The archives are available here: http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ You can post to the list at: [EMAIL PROTECTED] It's the same GNU Mailman stuff you are already used to. While it could be argued that this cross-product discussion is valuable to retain in both communities, I think there is considerable benefit to creating a specialised mailing list that focuses specifically on this integration path and the unique interoperation and configuration issues it creates. I think it would be good to get some of this discussion off of the SER and Asterisk-specific mailing lists where it has somewhat marginal relevance at times and refocus it. If you agree and are interested in this topic, you are invited to join the list. One last note: The SER/OpenSER community has been in a state of flux recently, with OpenSER undergoing a name change to Kamailio and subsequently seeing a fork. The incumbent Kamailio project is now in the process of merging with the original SER project. The choice of nomenclature for list is not meant to imply an endorsement of or affinity for the IPTel SER project per se. It is just that right now it serves the aim of terseness to use a common denominator, to refer to this family of projects as the SER ecosystem. Whether you are a SER, OpenSER, Kamailio, or OpenSIPS user, you are part of that SER ecosystem. That is why the list is named what it is. Thank you, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
That's not actually true. SER is very much alive and well and under constant development. How do I KNOW it's constant development (other than the chatter on the mailing list)? Because things keep changing in CVS, but there never seems to be a 'release' version. Just a release candidate. ;) Seriously, though... this seems to be a popular misconception. I hear it a lot. Where did you come across the information that SER is no longer developed? N. Alex Balashov wrote: No, the issue isn't my value or preference. The issue is that SER is no longer maintained or developed and has not been for several years. Tobias Wolf wrote: Alex Balashov schrieb: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Well, i am not getting the correct meaning of 'defunct', but from the last part of your suggestion i guess you value Kamailio/OpenSIPS more than SER. Are there some hard reasion for this. I am in the process of deciding which SIP server i want to use with Asterisk and just made a go at SER. Compilation was a little rough but it was manageable. I threw away every module which funtionality i didn't wanted at after it just worked. I was able to register SIP phones at the server and configure an outgoing rule so that every call that could not be handled by the SIP server would go to Asterisk. But i confess, that i didn't looked at the other two projects ... Maybe they are so much better. Can you please write one or two aspects that makes me understand better why this two projects are the better choice ? Thank you very much ... Tobias On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. -- #Joseph ___ -- 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] SER + Asterisk
SIP wrote: Seriously, though... this seems to be a popular misconception. I hear it a lot. Where did you come across the information that SER is no longer developed? That seems to be a consequence of looking at the releases. Anyway, I spoke too soon in saying that there's absolutely nothing going on with the project whatsoever in terms of development. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
Alex Balashov wrote: SIP wrote: Seriously, though... this seems to be a popular misconception. I hear it a lot. Where did you come across the information that SER is no longer developed? That seems to be a consequence of looking at the releases. Anyway, I spoke too soon in saying that there's absolutely nothing going on with the project whatsoever in terms of development. Yes... I'll agree the releases are a bit... odd. SER 0.9.6 (or possibly 0.9.7 -- I'm never sure) was the last actual 'release' labeled stable. However, SER 2.0 rc1 has been available for over a year now, and hasn't been granted that 'stable' label, even though I gather it's no more unstable than 0.9.6/7. All the while, work on SER 2.1 is commencing long before there's been a release of SER 2.0. It's incredibly difficult to follow. But this is where the OpenSER (now OpenSIPS) and SER projects differed in their ideology most often -- that of releases and documentation. SER was always a bit sparse on both, preferring to make up for it by way of solid innovations in the core code. Unfortunately, it's a bit like the tale of Seymour Cray. Here was a man who was convinced that if you built a supercomputer, people would buy it because it's the fastest thing out there, and building peripherals and/or software for it as part of the business plan was a waste of time and money. This ideological difference is why he left Control Data. This is why he was encouraged out of Cray Research. And this is why his final company, Cray Computer Corp failed -- that sort of missed idea that people will buy technology simply for the sake of having better technology. I see a lot of parellels there with OpenSIPS and SER. OpenSIPS is a stable plaform that has dozens of modules and documentation galore on how to mesh the system with this, that, and the other. SER has rock-solid, incredibly innovative core code, but prefers to leave the writing of modules and documentation as an exercise for the user, thereby making it perhaps overly difficult for anyone to implement or integrate. N. ___ -- 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] SER + Asterisk
SIP wrote: I see a lot of parellels there with OpenSIPS and SER. OpenSIPS is a stable plaform that has dozens of modules and documentation galore on how to mesh the system with this, that, and the other. SER has rock-solid, incredibly innovative core code, but prefers to leave the writing of modules and documentation as an exercise for the user, thereby making it perhaps overly difficult for anyone to implement or integrate. Perhaps, although I would argue that the core of OpenSIPS/Kamailio is by far the most useful. The modules simply provide an easier API and automation for some tasks that would otherwise have to be done manually in the route script or in one's own database, but they're mostly fairly trivial. Sometimes the modules' approach to a given task creates more problem than it solves. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
Alex Balashov schrieb: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Well, i am not getting the correct meaning of 'defunct', but from the last part of your suggestion i guess you value Kamailio/OpenSIPS more than SER. Are there some hard reasion for this. I am in the process of deciding which SIP server i want to use with Asterisk and just made a go at SER. Compilation was a little rough but it was manageable. I threw away every module which funtionality i didn't wanted at after it just worked. I was able to register SIP phones at the server and configure an outgoing rule so that every call that could not be handled by the SIP server would go to Asterisk. But i confess, that i didn't looked at the other two projects ... Maybe they are so much better. Can you please write one or two aspects that makes me understand better why this two projects are the better choice ? Thank you very much ... Tobias On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. -- #Joseph ___ -- 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] SER + Asterisk
No, the issue isn't my value or preference. The issue is that SER is no longer maintained or developed and has not been for several years. Tobias Wolf wrote: Alex Balashov schrieb: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Well, i am not getting the correct meaning of 'defunct', but from the last part of your suggestion i guess you value Kamailio/OpenSIPS more than SER. Are there some hard reasion for this. I am in the process of deciding which SIP server i want to use with Asterisk and just made a go at SER. Compilation was a little rough but it was manageable. I threw away every module which funtionality i didn't wanted at after it just worked. I was able to register SIP phones at the server and configure an outgoing rule so that every call that could not be handled by the SIP server would go to Asterisk. But i confess, that i didn't looked at the other two projects ... Maybe they are so much better. Can you please write one or two aspects that makes me understand better why this two projects are the better choice ? Thank you very much ... Tobias On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. -- #Joseph ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
Alex Balashov wrote: No, the issue isn't my value or preference. The issue is that SER is no longer maintained or developed and has not been for several years. The above statement is totally false. SER is indeed an ongoing project which is actively maintained. If you subscribed to the SERDEV mailing list you would know that. The latest update is just from last week: ser-2.0.1+cvs20081014_src.tar.gz 14-Oct-2008 06:26 2.5M Andres. ___ -- 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] SER + Asterisk
On 10/18/08 05:28, Grey Man wrote: As far as I'm aware SER (and it's derivatives) cannot initiate outbound registraitions. They can do the opposite and act as a SIP Registrar. For outbound registrations you should be able to use Asterisk. Regards, Greyman. Yes, I use Asterisk for iax outside registration but not for sip from Asterisk; I don't want to make a swizz cheese (open so many ports) out of my firewall. I can not use stun with Asterisk. I have my stand alone sip phone registered with the provider but I can only register it with one provider so no Asterisk access. What are my best options? I was thinking that something like nathelper with SER would be of any use to me but I see it might not be the case, it is only for helping to register clients IN not OUT. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
On 10/18/08 10:39, ram wrote: On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote: I would gladly go with any of the newer packages if I only could. I'm just working with what I can find in portage; I'm sure it will be eventually available. It will first show up via overlay. What I'm trying to do is to register SER to my VoIP provider via stun.fwdnet.net and connect SER with Asterisk, I just need some simple practical example; and upgrade will come with time. I'm sure it is possible even with old SER. Suggesting what is newer is not going to help me much :-) Hi Joseph you can use UAC Module to register with provider and make calls using SER/Openser/OpensSIPs or you can do other way is SER as registrar and Asterisk act a b2bua ( you can register with provider) let me know if it helps your need Ram Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
No, a proxy cannot *initiate* anything. ram wrote: On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 10/17/08 23:23, Kristian Kielhofner wrote: On 10/17/08, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. I would gladly go with any of the newer packages if I only could. I'm just working with what I can find in portage; I'm sure it will be eventually available. It will first show up via overlay. What I'm trying to do is to register SER to my VoIP provider via stun.fwdnet.net http://stun.fwdnet.net/ and connect SER with Asterisk, I just need some simple practical example; and upgrade will come with time. I'm sure it is possible even with old SER. Suggesting what is newer is not going to help me much :-) Hi Joseph you can use UAC Module to register with provider and make calls using SER/Openser/OpensSIPs or you can do other way is SER as registrar and Asterisk act a b2bua ( you can register with provider) let me know if it helps your need Ram ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
Doesn't mean it's not defunct. Joseph wrote: I'm using Gentoo and the only package I was able to find in portage was SER; I could compile manually but it is harder to upgrade and keep track of dependencies. -- #Joseph On 10/17/08 22:42, Alex Balashov wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. You cannot use the UAC module to register. The proxy is an event-driven element, by definition; it cannot initiate anything, nor can it itself possess UAC credentials. What you can do with the UAC module is take advantage of the proxy's ability to statelessly or statefully forward calls, branch calls, and reply to particular feedback by mimicking some of the behaviour of a UAC and/or sending an authentication digest in response to a registration or proxy challenge. But you can't use it to register with a provider as such. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On 10/18/08 13:51, Alex Balashov wrote: Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. No, my firewall does not support NAT gateway translation, it is freesco -- #Joseph ___ -- 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] SER + Asterisk
Joseph wrote: On 10/18/08 13:51, Alex Balashov wrote: Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. No, my firewall does not support NAT gateway translation, it is freesco Well, you *can* use the proxy to provide near-end NAT traversal. The UAC module won't help much here; your best bet is to statefully relay the REGISTER messages and the corresponding challenges. There is a nathelper module that can help you fix up the contact bindings if it they contain RFC1918 addresses. However, it should be emphasised in no uncertain terms that your UAC (Asterisk) must originate the request and relay it through the proxy; the proxy cannot originate it itself. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On 10/18/08 15:31, Alex Balashov wrote: Joseph wrote: On 10/18/08 13:51, Alex Balashov wrote: Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. No, my firewall does not support NAT gateway translation, it is freesco Well, you *can* use the proxy to provide near-end NAT traversal. The UAC module won't help much here; your best bet is to statefully relay the REGISTER messages and the corresponding challenges. There is a nathelper module that can help you fix up the contact bindings if it they contain RFC1918 addresses. However, it should be emphasised in no uncertain terms that your UAC (Asterisk) must originate the request and relay it through the proxy; the proxy cannot originate it itself. Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On 10/18/08 16:10, Alex Balashov wrote: Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. I'm trying to find a good manual for SER with decent examples for beginners but don't have much luck. I think these package OpenSER OpenSIPS SER are not so common as it is hard to understand them. The manual on their web-page is just dry plain language without examples so it makes it harder to understand. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
Joseph wrote: On 10/18/08 16:10, Alex Balashov wrote: Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. I'm trying to find a good manual for SER with decent examples for beginners but don't have much luck. I think these package OpenSER OpenSIPS SER are not so common as it is hard to understand them. The manual on their web-page is just dry plain language without examples so it makes it harder to understand. There is not really a lot of good conceptual introduction to OpenSER, although Flavio Goncalves' book (Building Scalable Telephony Applications With OpenSER) may be somewhat of aid. The documentation primarily serves those that already know what they are doing, kind of like programmers that just need an API reference. But basically, it is admittedly a lot of work to figure out an extremely polymorphic and idiosyncratic environment just to solve a relatively simple problem. I recommend contracting someone to take care of it for you, or stealing a recipe from somewhere. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On 10/18/08 16:48, Alex Balashov wrote: [snip] There is not really a lot of good conceptual introduction to OpenSER, although Flavio Goncalves' book (Building Scalable Telephony Applications With OpenSER) may be somewhat of aid. The documentation primarily serves those that already know what they are doing, kind of like programmers that just need an API reference. But basically, it is admittedly a lot of work to figure out an extremely polymorphic and idiosyncratic environment just to solve a relatively simple problem. I recommend contracting someone to take care of it for you, or stealing a recipe from somewhere. I totally agree with you, it is hard to understand, I've been telling all alone that programmers shouldn't write manuals :-) I'm in a stage that even if I stole a recipe from someone I wouldn't know what to do with it :-/ -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
On Sat, Oct 18, 2008 at 4:48 PM, Alex Balashov [EMAIL PROTECTED]wrote: Joseph wrote: On 10/18/08 16:10, Alex Balashov wrote: Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. I'm trying to find a good manual for SER with decent examples for beginners but don't have much luck. I think these package OpenSER OpenSIPS SER are not so common as it is hard to understand them. The manual on their web-page is just dry plain language without examples so it makes it harder to understand. There is not really a lot of good conceptual introduction to OpenSER, although Flavio Goncalves' book (Building Scalable Telephony Applications With OpenSER) may be somewhat of aid. The documentation primarily serves those that already know what they are doing, kind of like programmers that just need an API reference. But basically, it is admittedly a lot of work to figure out an extremely polymorphic and idiosyncratic environment just to solve a relatively simple problem. I recommend contracting someone to take care of it for you, or stealing a recipe from somewhere. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 Jeremy McNamara wrote some helpful tidbits as well. http://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-registar-sip-proxy-and-far-end-nat-traversal-for-media/ This one is a Configuration Wizard I haven't tried it out yet but certainly will at some point http://www.jeremy-mcnamara.com/2007/02/22/seropenser-configuration-wizard/ If someone wrote a nice webmin module with all the configuration options as check boxes and fill in the blanks, that would be very NICE! -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] SER + Asterisk
Steve Totaro wrote: If someone wrote a nice webmin module with all the configuration options as check boxes and fill in the blanks, that would be very NICE! The problem with simply doing a GUI frontend to *SER is that it's very polymorphic far too extensible; there are far too many potential applications, and those applications are far too customised and situation-specific. That's why the routing script takes the character that it does, because it wishes to have as few cookie-cutter characteristics as possible. That having been said, there are plenty of common use cases of the product which probably deserve GUI implementation. But it needs to be understood that they are just common use cases, nothing more, and represent an infinitesimal fraction of conceivable -- and routine -- applications. The product is far too low-level to be able to say what it does even in the loose ways in which we routinely attribute certain functional goals or traits to Asterisk. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On Sat, Oct 18, 2008 at 5:35 PM, Alex Balashov [EMAIL PROTECTED]wrote: Steve Totaro wrote: If someone wrote a nice webmin module with all the configuration options as check boxes and fill in the blanks, that would be very NICE! The problem with simply doing a GUI frontend to *SER is that it's very polymorphic far too extensible; there are far too many potential applications, and those applications are far too customised and situation-specific. That's why the routing script takes the character that it does, because it wishes to have as few cookie-cutter characteristics as possible. That having been said, there are plenty of common use cases of the product which probably deserve GUI implementation. But it needs to be understood that they are just common use cases, nothing more, and represent an infinitesimal fraction of conceivable -- and routine -- applications. The product is far too low-level to be able to say what it does even in the loose ways in which we routinely attribute certain functional goals or traits to Asterisk. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 Kind of like SwitchVox, FreePBX, Thirdlane.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] SER + Asterisk
Steve Totaro wrote: Kind of like SwitchVox, FreePBX, Thirdlane.. I don't know that I'd make that comparison. I would say that in general, OpenSER is more low-level and amorphous and multipurpose than Asterisk or any GUI that wraps it. Asterisk has many applications and uses and niches, but these are all uses that capitalise on the sort of thing that Asterisk is. The genus of thing that it is on a technical level and the role it plays is fairly well-understood, even if there are many things you can do with that particular type of thing. OpenSER is hard to pin down like that. The closest you can come to it is to say that it is a proxy/UAS, and what does that really get you? It's used in situations that offer far less taxonomic resemblance to each other than sundry appropriations of Asterisk do. Yes, there's no argument that there are many things OpenSER does that can be driven by a GUI. But at the same time, that approach is somewhat antithetical to its basic nature. Its roles cannot be usefully anticipated nearly as well or as much. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. -- #Joseph ___ -- 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] SER + Asterisk
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. -- #Joseph ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
Re: [asterisk-users] SER + Asterisk
I'm using Gentoo and the only package I was able to find in portage was SER; I could compile manually but it is harder to upgrade and keep track of dependencies. -- #Joseph On 10/17/08 22:42, Alex Balashov wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. ___ -- 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] SER + Asterisk
On 10/17/08 23:23, Kristian Kielhofner wrote: On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. I would gladly go with any of the newer packages if I only could. I'm just working with what I can find in portage; I'm sure it will be eventually available. It will first show up via overlay. What I'm trying to do is to register SER to my VoIP provider via stun.fwdnet.net and connect SER with Asterisk, I just need some simple practical example; and upgrade will come with time. I'm sure it is possible even with old SER. Suggesting what is newer is not going to help me much :-) -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
As far as I'm aware SER (and it's derivatives) cannot initiate outbound registraitions. They can do the opposite and act as a SIP Registrar. For outbound registrations you should be able to use Asterisk. Regards, Greyman. ___ -- 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] SER + Asterisk
On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote: On 10/17/08 23:23, Kristian Kielhofner wrote: On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. I would gladly go with any of the newer packages if I only could. I'm just working with what I can find in portage; I'm sure it will be eventually available. It will first show up via overlay. What I'm trying to do is to register SER to my VoIP provider via stun.fwdnet.net and connect SER with Asterisk, I just need some simple practical example; and upgrade will come with time. I'm sure it is possible even with old SER. Suggesting what is newer is not going to help me much :-) Hi Joseph you can use UAC Module to register with provider and make calls using SER/Openser/OpensSIPs or you can do other way is SER as registrar and Asterisk act a b2bua ( you can register with provider) let me know if it helps your need Ram ___ -- 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] Ser, asterisk and ip2ipgw
Hi, i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull b2bua server, any suggestion to howto or documentation to asterisk integration and b2b use? ty in advance. -- Riccardo Cupardo ___ -- 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] Ser, asterisk and ip2ipgw
Riccardo Cupardo wrote: Hi, i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull b2bua server, any suggestion to howto or documentation to asterisk integration and b2b use? Well, Asterisk is a B2BUA. And it keeps state. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER / Asterisk and mediapath
Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can anyone give me any hints? I wasn't sure if using canreinvite since I wasn't sure if that would affect the caller's interaction in the IVR. Thanks ___ --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] SER / Asterisk and mediapath
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can anyone give me any hints? I wasn't sure if using canreinvite since I wasn't sure if that would affect the caller's interaction in the IVR. Hi yes can canreinvite does the job depends on peer compatability ram ___ --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] SER+Asterisk integration
I have ser sitting on my iptables nat box and my asterisk box on the lan
. Ser does forwarding so that any requests (register,invite,ack,...) to
the nat box at 5060 r sent to my asterisk box on the lan .I can register
from outside
to my asterisk box but there is only one way audio , reason being that
when the asterisk box sends a sip packet whith session description the sdp
part of the sip packet is not natted .I have tried the following :
if(src_ip==10.0.0.0/255.0.0.0){
force_rtp_proxy();
encode_contact(enc_prefix,wanip);
sdp_mangle_ip(10.0.0.0/255.0.0.0,wanip);
};
and it does not work because my ethernet dump shows that the contact in
sdp is
not mangled.
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Re: [asterisk-users] SER+Asterisk integration
Have a check through:http://www.voip-info.org/wiki-NAT+and+VOIPhttp://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
RegardsRobOn 03/09/06, Siqhamo Sifo [EMAIL PROTECTED] wrote:
I have ser sitting on my iptables nat boxand my asterisk box on the lan. Ser does forwarding so that any requests (register,invite,ack,...) tothe nat box at 5060 r sent to my asterisk box on thelan .I can register
from outsideto my asterisk box but there is only one way audio , reason being thatwhen the asterisk box sends a sip packet whith session description the sdppart of the sip packet is not natted .I have tried the following:
if(src_ip==10.0.0.0/255.0.0.0){force_rtp_proxy(); encode_contact(enc_prefix,wanip); sdp_mangle_ip(
10.0.0.0/255.0.0.0,wanip);};and it does not work because my ethernet dump shows that the contact insdp is
not mangled.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] SER+Asterisk integration
have a look at the nathelper examples in SER distribution. This is from
an rather old installation of mine.
--
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test(3)) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == REGISTER || ! search(^Record-Route:)) {
xlog(L_ERR, LOG: Someone trying to register from private
IP, rewriting\n);
# This will work only for user agents that support symmetric
# communication. We tested quite many of them and majority i
s
# smart enough to be symmetric. In some phones it takes a co
nfiguration
# option. With Cisco 7960, it is called NAT_Enable=Yes, with
kphone it is
# called symmetric media and symmetric signalling.
fix_nated_contact(); # Rewrite contact with source IP of sig
nalling
if (method == INVITE) {
fix_nated_sdp(1); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6);# Mark as NATed
};
};
..
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
--
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[asterisk-users] SER + Asterisk PSTN calls don't hung up
Hi, I'm deploying a SER + Asterisk architecture, where SER is used to manage acc, users database and sip routing, and Asterisk is used for voicemail and PSTN gateway. The system is already able to make and receive calls from the PSTN, although, only after the call has been established it can be hung up with success; when it is still ringing, if any side hungs up the call, it still keeps ringing on the other side. Observing with Ethereal, we concluded that in this erroneous cases, the CANCEL SIP request isn't transmitted from the SER to Asterisk (if cancelled from the VoIP side) being transmitted a 404 User Not Found message from SER to Sip Phone. If hung from the PSTN side, the sip phone keeps calling after that, and ends calling by time-out being observed a 486 Busy Here status message from Asterisk to SER and then from SER to sip phone. Any help, please? Regards, Ricardo. ___ --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
Re: [asterisk-users] SER + Asterisk PSTN calls don't hung up
Problem solved.
It was needed to insert the following code in ser.cfg:
-
if (method==CANCEL) {
route(1);
break;
}
-
and also:
-
exten = _0.,2,Busy
exten = _0.,3,Hangup
-
Ricardo.
Ricardo Carvalho wrote:
Hi,
I'm deploying a SER + Asterisk architecture, where SER is used to
manage acc, users database and sip routing, and Asterisk is used for
voicemail and PSTN gateway.
The system is already able to make and receive calls from the PSTN,
although, only after the call has been established it can be hung up
with success; when it is still ringing, if any side hungs up the call,
it still keeps ringing on the other side. Observing with Ethereal, we
concluded that in this erroneous cases, the CANCEL SIP request isn't
transmitted from the SER to Asterisk (if cancelled from the VoIP side)
being transmitted a 404 User Not Found message from SER to Sip
Phone. If hung from the PSTN side, the sip phone keeps calling after
that, and ends calling by time-out being observed a 486 Busy Here
status message from Asterisk to SER and then from SER to sip phone.
Any help, please?
Regards,
Ricardo.
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[Asterisk-Users] SER Asterisk with DID incoming and out going
Hi all I have badly NATed Clients proble with one way Voice After reading some documents people ask me to use STUN Server But still i have some problem with one way Voice I have setup like below iam trying with 2 extensions 1 extention in the same LAN where the * installed 2 extension in different network, NATed IP , 3. both the side iam use SIPURA 4. i have 2 DID from provider 5. i have route them to appropriate extensions Iam able to make calls in and out but the problem where iam setting up server have limited bandwidth So i have installed G729 codec So i want to make RTP so i made setup caninvite=yes since my provider support that option then my NAT Clients have One way Voice problem So after Reading some DOCS SER, should be able to do this Job so SER can be integrated with *, if yes can any one point me to some URL thanks ram ___ --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
Re: [Asterisk-Users] SER Asterisk with DID incoming and out going
ram wrote: Hi all I have badly NATed Clients proble with one way Voice After reading some documents people ask me to use STUN Server But still i have some problem with one way Voice use stun on dinamic ip :) I have setup like below iam trying with 2 extensions 1 extention in the same LAN where the * installed 2 extension in different network, NATed IP , 3. both the side iam use SIPURA 4. i have 2 DID from provider 5. i have route them to appropriate extensions Iam able to make calls in and out but the problem where iam setting up server have limited bandwidth So i have installed G729 codec So i want to make RTP so i made setup caninvite=yes canreinvite=no nat=yes since my provider support that option then my NAT Clients have One way Voice problem So after Reading some DOCS SER, should be able to do this Job so SER can be integrated with *, if yes can any one point me to some URL thanks ram ___ --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 -- Поздрави, Андрей Сотиров ___ --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
Re: [Asterisk-Users] SER Asterisk with DID incoming and out going
Hi thanks for the reply ya the default is NAT=YES only if i keep reinvite=no, the my server b/w consuming lot since i have bottleneck of server bandwidth ram On 3/16/06, Andrei Sotirov [EMAIL PROTECTED] wrote: ram wrote: Hi all I have badly NATed Clients proble with one way Voice After reading some documents people ask me to use STUN Server But still i have some problem with one way Voiceuse stun on dinamic ip :) I have setup like below iam trying with 2 extensions 1 extention in the same LAN where the* installed 2 extension in different network, NATed IP , 3. both the side iam use SIPURA 4. i have 2 DID from provider 5. i have route them to appropriate extensions Iam able to make calls in and out but the problem where iam setting up server have limited bandwidth So i have installed G729 codec So i want to make RTP so i made setup caninvite=yescanreinvite=nonat=yes since my provider support that option then my NAT Clients have One way Voice problem So after Reading some DOCS SER, should be able to do this Job so SER can be integrated with *, if yes can any one point me to some URL thanks ram ___ --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--Поздрави,Андрей Сотиров___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] SER ,Asterisk and MWI
hello, I am trying to pass MWI from Asterisk to SER.my user agents register with Ser.i am not able to figure out how to do this. i added the changes for mailbox in sip.conf for ser peer entry. [ser] type=friend mailbox=XYZ also changes in chan_sip.c for asterisk but not seeing the notify messages hitting my ser server. any suggestions please, i would highly appreciate. Thanks, AA ___ --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
[Asterisk-Users] SER + Asterisk
When using Asterisk and SER together, should SER place calls to the PSTN, and Asterisk only deal with special features such as voicemail, queues, autoattendants, etc? Or should SER be used ONLY as a proxy/registrar, and all calls be routed to Asterisk so that Asterisk places the calls to the PSTN? Cheers! -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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
RE: [Asterisk-Users] SER Asterisk combination to get around NAT
Mark, Thanks for your response. The typical deployment is a single server in the customer location directly on the end of an ADSL link with two Ethernet interfaces, 1 to the ADSL modem and the other to the LAN. The LAN side is fine and is as normal but many customers have remote users or remote small offices that may have more than one SIP device behind NAT. What I am trying to establish is how successful SER is at allowing multiple remote SIP devices behind a remote NAT router to interact with Asterisk and what issues need to be taken into account such as MWI and or codec's. I have been using Asterisk for quite some time but have not played with SER yet and so does anyone have some sample SER configs to work in this type of deployment. Stuart -Original Message- From: Mark John Buenconsejo [mailto:[EMAIL PROTECTED] Sent: 18 November 2005 06:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SER Asterisk combination to get around NAT Importance: High Hello Stuart, we have, and I would be happy to help you setup both Asterisk and SER on a consultancy basis. You can find more information about me here: http://mark.teamcebu.com Basically, it requires SER to forward the SIP messages to Asterisk, and that SER be configured as one of the SIP channels on Asterisk. What happens is: from the LAN Phone, it connects to SER and then SER forwards it to Asterisk Asterisk will connect to the actual destination As soon as Asterisk is able to connect to the destination, it then replies to the phone that the call is connected At this point, the actual call connections are made (asterisk-to-phone and asterisk-to-destination) and then Asterisk bridges the asterisk-to-destination and asterisk-to-phone connections The bridged call mechanism on Asterisk works around the NAT limitations In this setup, it will appear that the Phone is connecting to Asterisk (LAN side), and that the destination is talking to Asterisk (Live side), and Asterisk passes the RTP packets back-and-forth. There are a few considerations though, such as codec supports. As much as possible use the same codec for each leg of the call, otherwise the call quality deteriorates during transcoding. By the way, we're using this with up to 12 simultaneous calls in our setup (a small call center), using either iLBC and G.729 codec. Anyway, let me know if you need further help. :) Or if you have some more specific questions. Thanks! Mark Stuart Hirst wrote: Has anyone successfully used SER and Asterisk together on the same server to get around NAT traversal issues. I have looked at many of the NAT traversal topics which either involve commercial products and significant costs or solutions such as STUN or proprietary systems such as xten. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/174 - Release Date: 17/11/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/174 - Release Date: 17/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk combination to get around NAT
Stuart Hirst ha scritto: Has anyone successfully used SER and Asterisk together on the same server to get around NAT traversal issues. I have looked at many of the NAT traversal topics which either involve commercial products and significant costs or solutions such as STUN or proprietary systems such as xten. I've installed ser + mediaproxy + asterisk without much trouble following the docs you find at www.onsip.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER Asterisk combination to get around NAT
Has anyone successfully used SER and Asterisk together on the same server to get around NAT traversal issues. I have looked at many of the NAT traversal topics which either involve commercial products and significant costs or solutions such as STUN or proprietary systems such as xten. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/174 - Release Date: 17/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER+ASTERISK
No ! Asterisk should send the invite request to sip proxy . Harry --- Walter Willis [EMAIL PROTECTED] a écrit : the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER+ASTERISK
Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER+ASTERISK
the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello,I wish to setup this scheme:ser-0.9.4asterisk-1.2-bêtapolycom ip300 phonesasterisk 5050-- |firewall+nat|-192.168.ser 5060---My ip phones use ser as outbound sip proxy and asterisk as sip registrar server.Ser Forward REGISTER requests to asterisk however whena phone try to send an invite message then asterisksend icmp to private ip (host=dynamic in sip.conf)Is it possible to solve this problem ? RegardsHarry___Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez cette version sur http://fr.messenger.yahoo.com___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER+ASTERISK
Hello Walter, The ser an asterisk run in the same box. What do you mean redirect host + port :) Sip agents send sip requests to ser (outbound proxy) and this one to asterisk ! sip agents are both registered on ser and asterisk. Please to explain me how asterisk redirect the requests. Regards Harry --- Walter Willis [EMAIL PROTECTED] a écrit : the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER+ASTERISK
you could wait infinitely or try users list..On 11/4/05, harry gaillac [EMAIL PROTECTED] wrote: Hello Walter,The ser an asterisk run in the same box.What do you mean redirect host + port :) Sip agents send sip requests to ser (outbound proxy)and this one to asterisk !sip agents are both registered on ser and asterisk.Please to explain me how asterisk redirect therequests.Regards Harry--- Walter Willis [EMAIL PROTECTED] a écrit : the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello,I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phonesasterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez cette version sur http://fr.messenger.yahoo.com___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER+ASTERISK
my bad you are.. lol didnt realize.. On 11/4/05, Jimmy Smith [EMAIL PROTECTED] wrote: you could wait infinitely or try users list..On 11/4/05, harry gaillac [EMAIL PROTECTED] wrote: Hello Walter,The ser an asterisk run in the same box.What do you mean redirect host + port :) Sip agents send sip requests to ser (outbound proxy)and this one to asterisk !sip agents are both registered on ser and asterisk.Please to explain me how asterisk redirect therequests. Regards Harry--- Walter Willis [EMAIL PROTECTED] a écrit : the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phonesasterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez cette version sur http://fr.messenger.yahoo.com___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER+ASTERISK voicemail
Hello, I set SER as sip proxy and ASTERISK as voicemail server (ARA) and serweb as TUI (telephone user interface) . Serweb | Ua---ser---asterisk voicemail | | Mysql DB I add user agents with address sip:[EMAIL PROTECTED] + aliases sip:[EMAIL PROTECTED] where 123 is mailbox I can forward voice messages to Asterisk with failure route for status 408 or 486. However I can't do it for offline users because of SER look for addresses like sip:[EMAIL PROTECTED] not sip:[EMAIL PROTECTED] where 123 is mailbox How could I solve this problem if possible ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER + ASTERISK voicemail
Hello,
Thanks for help it's ok with static file
voicemail.conf
However something is wrong with ARA .
app_voicemail search entries in voicemail.conf ?!
I set apps/Makefile for USE_ODBC_STORAGE.
Regards
Harry
//
Connected to Asterisk CVS-HEAD currently running on
serveur1 (pid = 2584)
Verbosity is at least 3
-- Executing VoiceMail(SIP/asterisk-8db8, b84)
in new stack
Aug 29 16:11:40 WARNING[7947]: app_voicemail.c:2602
leave_voicemail: No entry in voicemail config file
for '84'
Aug 29 16:11:50 WARNING[7947]: pbx.c:2336
__ast_pbx_run: Timeout, but no rule 't' in context
'loc al'
serveur1*CLI odbc show
Name: asterisk
DSN: asterisk
Connected: yes
serveur1*CLI
///
--- Steve Blair [EMAIL PROTECTED] a écrit :
You'll want some rules in your sip.conf to handle
the connection from
SER. A
starting point might be:
[ser ip addr:ser port ?= 5060]
type=peer
context=my sip context name
tos=lowdelay; tos delay
allow=ulaw ; dtmfmode=inband
only works with ulaw
or alaw!
dtmfmode=inband; Choices are
inband, rfc2833, or info
You'll then want some rules in extensions.conf to
accept the call and
redirect it
to mailboxes defined in your voicemail.conf or in
MySQL. Something like:
[general]
context=my sip context name
switch = Realtime/my sip context
name@extensions
static=yes
[my sip context name]
exten = _uX,1,VoiceMail(${EXTEN}@my sip
context name)
exten = _X,1,VoiceMail(${EXTEN}@my sip
context name)
exten = _bX,1,VoiceMail(${EXTEN}@my sip
context name))
exten = #,2,Hangup ; Hang
them up.
Steve
harry gaillac wrote:
Hello,
I try set Ua---SERAsterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to
asterisk
voicemail in failure route.
How may I configure extensions.conf and ser.cfg ?
I have been trying without success!
Regards
Harry
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[Asterisk-Users] SER + ASTERISK voicemail
Hello, I try set Ua---SERAsterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://mail.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER + ASTERISK voicemail
You'll want some rules in your sip.conf to handle the connection from
SER. A
starting point might be:
[ser ip addr:ser port ?= 5060]
type=peer
context=my sip context name
tos=lowdelay; tos delay
allow=ulaw ; dtmfmode=inband only works with ulaw
or alaw!
dtmfmode=inband; Choices are inband, rfc2833, or info
You'll then want some rules in extensions.conf to accept the call and
redirect it
to mailboxes defined in your voicemail.conf or in MySQL. Something like:
[general]
context=my sip context name
switch = Realtime/my sip context name@extensions
static=yes
[my sip context name]
exten = _uX,1,VoiceMail(${EXTEN}@my sip context name)
exten = _X,1,VoiceMail(${EXTEN}@my sip context name)
exten = _bX,1,VoiceMail(${EXTEN}@my sip context name))
exten = #,2,Hangup ; Hang them up.
Steve
harry gaillac wrote:
Hello,
I try set Ua---SERAsterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.
How may I configure extensions.conf and ser.cfg ?
I have been trying without success!
Regards
Harry
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[Asterisk-Users] SER, Asterisk, SIP proxy, routing, redirection - confused
I am fairly new to Asterisk / VOIP and have been playing around with it for long enough to have a whole lot of questions so far without answers. Presently Im running Asterisk (v.1.0.7) on a Debian Sarge installation with 2 soft phones (for testing purposes). A live deployment will probably have a dozen-odd extensions. I wish to have both SIP and PSTN services exposed to the outside and will probably install an appropriate Digium card to allow me to connect PSTN lines. We pay ransom to Microsloth for our company network. I am reading that Asterisk does not provide SIP proxying services however proxy services are very important (one reference said critical) to routing in SIP as it provides for dynamic rewriting, redirection and inter-domain routing. In Asterisk, how are these functions meant to work? As far as I can tell, it cannot perform inter-domain routing as it has no proxying capability but apparently provides redirection and rewriting services. Am I going to require the services of SER (perhaps in a gateway role) in order to achieve any or all of these functions or will Asterisk alone provide it? I have been reading the SER documentation and it seems to be very capable however I think that establishing the dial plan and voicemail in Asterisk may be a simpler and clearer process. So my next question may be how are people deploying Asterisk with a separate proxy server? Early on I was reading that a proxy is mainly useful in a large environment (thousands of extensions) in order to reduce the load on the Asterisk server however this doesnt seem to mesh with what Im reading now about a proxy providing SIP routing services. To date, I have only been able to set up Asterisk with fixed extension numbers with no facility for authenticating a particular user at a terminal. Being able to tell Asterisk where a particular user is and direct calls to them is one of the core capabilities of SIP and is one of the key reasons why we want to deploy it into our office. Yet Ive seen no documentation on how to do this. As you can probably gather, Im rather confused about how to develop / deploy a VOIP solution. There is much written about the topic however they seem to say conflicting things Any help would be appreciated. Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER Asterisk SIP =513 Message Too Big
Title: SER Asterisk SIP =513 Message Too Big
Using Asterisk 1.0.9
When I try to make an outgoing call with SIP I get the message 513 Message too big back from SER. Any ideas what I am doing wrong?
Debug below.
SER and Asterisk are running on the same Server
SER is on port 5060
Asterisk is on port 5061
In my extension.conf I have the line
SERADDRESS=192.219.85.57:5060
in Globals
and am using
exten =_5XXX,2,Dial(sip/${EXTEN:[EMAIL PROTECTED])
to dial out.
Here is the sip debug.
-- Executing Ringing(H323/ip$192.219.85.57:2680/5746, ) in new stack
-- Executing Dial(H323/ip$192.219.85.57:2680/5746, sip/[EMAIL PROTECTED]:5060) in new stack
We're at 192.219.85.57 port 13054
Answering/Requesting with root capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: 223 sip:[EMAIL PROTECTED]:5061;tag=as01e72172
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5061
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 25 Jul 2005 14:33:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 30548 30548 IN IP4 192.219.85.57
s=session
c=IN IP4 192.219.85.57
t=0 0
m=audio 13054 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 192.219.85.57:5060
-- Called [EMAIL PROTECTED]:5060
Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: 223 sip:[EMAIL PROTECTED]:5061;tag=as01e72172
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 192.219.85.57:5060 Noisy feedback tells: pid=19732 req_src_ip=192.219.85.57 req_src_port=5061 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1
9 headers, 0 lines
Sip read:
SIP/2.0 513 Message too big
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: 223 sip:[EMAIL PROTECTED]:5061;tag=as01e72172
To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 192.219.85.57:5060 Noisy feedback tells: pid=19732 req_src_ip=192.219.85.57 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==11
9 headers, 0 lines
-- Got SIP response 513 Message too big back from 192.219.85.57
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: 223 sip:[EMAIL PROTECTED]:5061;tag=as01e72172
To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab
Contact: sip:[EMAIL PROTECTED]:5061
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.219.85.57:5060
== No one is available to answer at this time
Incoming calls from a soft SIP phone to SER and then through to asterisk work fine.
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[Asterisk-Users] ser+asterisk problem
hello I am using ser with asterisk asterisk on 5070 (on back end) ser on 5060 (on front end) i am getting all requests at asterisk. i tried by changing asterisk port bindport=5090 but still getting all requests from sjphone at asterisk. can any one tell what is the reason regrads Kamran __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ser+asterisk problem
On 19/05/05, Kamran Ahmad [EMAIL PROTECTED] wrote: hello I am using ser with asterisk asterisk on 5070 (on back end) ser on 5060 (on front end) i am getting all requests at asterisk. i tried by changing asterisk port bindport=5090 but still getting all requests from sjphone at asterisk. can any one tell what is the reason Did you restart Asterisk - that's a complete restart, not just a 'reload' Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER Asterisk and NAT
I have been trying to setup Asterisk in combination with SER on the same box as a PBX with SIP clients. I would like to have it available for both external and internal users so I have the box setup with external and internal IP address. I am running into all kinds of troubles with this configuration, specifically with forwarding voicemail to Asterisk from SER. Does anyone have a similar setup that is working? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER + Asterisk
I'm working with SER + Asterisk. I was told that to have SER push calls to multiple Asterisk servers, I can use the LCR Module, I'll just give all the Asterisk servers the same weight/price, and SER will randomly send outbound requests to each Asterisk server. It's not truly equally balanced, so one server could get more calls while the other has spare resources. So although it does increase the number of simul. outbound calls that can be made, it still doesn't make me feel good knowing it's not perfectly load balanced. Can somebody help elaborate? Is there a better way to get SER to evenly balance between the Asterisk servers? I wonder if I use Asterisk's ability to limit the number of simult. calls, if Asterisk gets more than, lets say 300, calls, then it would reject calls, I wonder if SER would then try sending it to the other Asterisk server, which may have available channels. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ser - asterisk configs anyone?
I have searched high and low for these, but to no avail, nothing useful back from google, nothing I could find on this mailing list, or voip-user.org. Does anyone have any good urls and or pointers which will assist in configuring SIP Express Router and Asterisk talking to each other on the same machine? Many thanks, Spencer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser - asterisk configs anyone?
This may help, I just happen to be a google searching master :) http://www.voip-info.org/wiki-Asterisk+at+large -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of G.Marshall Sent: Wednesday, April 06, 2005 11:06 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ser - asterisk configs anyone? I have searched high and low for these, but to no avail, nothing useful back from google, nothing I could find on this mailing list, or voip-user.org. Does anyone have any good urls and or pointers which will assist in configuring SIP Express Router and Asterisk talking to each other on the same machine? Many thanks, Spencer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ser - asterisk -cisco gateway
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hi,
we have the ser sip-proxy for registration and we forwarding
the call to our cisco gateway and it works.
but now we will forwarding the calls to the asterisk and
the asterisk shoud forward the calls to our gw (via sip not h323).
how must i configure the asterisk
ser.cfg
if(uri =~sip:1024#){
~ log(1,Forwarding to Asterisk\n);
~ setflag(1);
~ rewritehostport(192.168.1.3:5061);
~ t_relay();
}
asterisk
thanks hans
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFCS/e6ouYj3oyEw4wRAoseAKCffEjSqxRGPmZaJYawqdoFrVjURACdHIXt
98DkG/axeJ4Gp6ENnMd0shk=
=ik0/
-END PGP SIGNATURE-
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[Asterisk-Users] ser, asterisk and conferencing
Hi List, Can I use asterisk to enable call conferencing? I'm using ser for the UA's to register, can I do something like if they dial a certain digits, it will forward it asterisk and use asterisks meetme feature? can i do meetme using only sip? Sorry for my terms, hope you understand my question. Regards, Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser, asterisk and conferencing
Hi ron, Of course you can make meetme, what you need is a zaptel device or, if you haven't any hardware, the ztdummy device. Install it (google), compile asterisk again, define an extension and it should work, more or less ;-))! Greetings, Mario -Original Message- From: ron [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 31. März 2005 16:07 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ser, asterisk and conferencing Hi List, Can I use asterisk to enable call conferencing? I'm using ser for the UA's to register, can I do something like if they dial a certain digits, it will forward it asterisk and use asterisks meetme feature? can i do meetme using only sip? Sorry for my terms, hope you understand my question. Regards, Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser, asterisk and conferencing
Ron, can I suggest a little more research next time. Everything you are asking is already very well documented on the wiki. The answer to your question is Yes - Asterisk can do call conferencing. http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ron Sent: Thursday, March 31, 2005 9:07 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ser, asterisk and conferencing Hi List, Can I use asterisk to enable call conferencing? I'm using ser for the UA's to register, can I do something like if they dial a certain digits, it will forward it asterisk and use asterisks meetme feature? can i do meetme using only sip? Sorry for my terms, hope you understand my question. Regards, Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ser - asterisk -cisco gateway
I am about to embark down the path of hooking up SER to Asterisk so my
understanding may be incorrect. I hope an expert will correct me if I'm
wrong.
My understanding is this:
1. Asterisk doesn't strictly forward calls in the way you suggest. It acts
as a UA and bridges the call. Have a look at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20not-proxy for
more.
2. To do what you want I think you need to set each sip client up in
sip.conf. The article at http://www.voip-info.org/wiki-Asterisk+at+large
discusses voicemail integration between Asterisk and SER but should be
helpful for you.
Regards
Cameron
- Original Message -
From: hans [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 01, 2005 1:14 AM
Subject: [Asterisk-Users] ser - asterisk -cisco gateway
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hi,
we have the ser sip-proxy for registration and we forwarding
the call to our cisco gateway and it works.
but now we will forwarding the calls to the asterisk and
the asterisk shoud forward the calls to our gw (via sip not h323).
how must i configure the asterisk
ser.cfg
if(uri =~sip:1024#){
~ log(1,Forwarding to Asterisk\n);
~ setflag(1);
~ rewritehostport(192.168.1.3:5061);
~ t_relay();
}
asterisk
thanks hans
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Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFCS/e6ouYj3oyEw4wRAoseAKCffEjSqxRGPmZaJYawqdoFrVjURACdHIXt
98DkG/axeJ4Gp6ENnMd0shk=
=ik0/
-END PGP SIGNATURE-
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[Asterisk-Users] ser+asterisk - security
Hi there, I'm using ser and asterisktogether. Asterisk for voice mail etc and ser forregistration of the users usig database.I can restrict forwarding callsfrom another sip proxy to ser(using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Thanks in advance, Pavel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser+asterisk - security
Pavel Siderov - Hostmates wrote: I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Use iptables on the asterisk machine to only allow SIP traffic from the machine with SER? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ser+asterisk - security
Hi Andreas, it's impossible to use iptables due to the reason that audio flows through asterisk and users won't be able to communicate w/ *... I've tried that. Regards, Pavel - Original Message - From: Andreas Sikkema [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 10:40 AM Subject: RE: [Asterisk-Users] ser+asterisk - security Pavel Siderov - Hostmates wrote: I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Use iptables on the asterisk machine to only allow SIP traffic from the machine with SER? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser+asterisk - security
[EMAIL PROTECTED] wrote: it's impossible to use iptables due to the reason that audio flows through asterisk and users won't be able to communicate w/ *... I was thinking of just the SIP port. I am assuming that asterisk protects its RTP ports from processing traffic from a third party. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER - Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)
Hello all! I googled lists.digium.com and ser mailing list, but did
not find any working configuration of asterisk used as voicemail for
SER. This is my config
if (uri==myself) {
if (method==REGISTER) {
save(location);
log (1, Registered\n);
break;
};
if (lookup(location)) {
log (1, *** IP to IP call *);
if (method == INVITE){
setflag (1);
t_on_failure(1);
t_relay();
sl_send_reply (180, Ringing);
setflag (1);
break;
}
if (!t_relay()) {
sl_send_reply(404, Not Found);
break;
};
#};
break;
};
failure_route[1] {
revert_uri();
forward(69.70.x.x,5060);
break();
}
Asterisk sip.conf:
[ser]
host=69.70.x.x
context=ser
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1
allow=gsm
allow=ilbc
nat=yes
extensions.conf:
[ser]
include = vm
include = messagecenter
[vm]
exten = _9.,1,VoiceMail(u${EXTEN})
exten = _9.,2,Hangup
[messagecenter]
exten = 555,1,Answer
exten = 555,2,Wait(1)
exten = 555,3,VoiceMailMain(default)
exten = 555,4,Hangup
exten = _555X.,1,Answer; can dial 555exten
to skip 'mailbox' prompt. Useful for speedial.
exten = _555X.,2,Wait(1)
exten = _555X.,3,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
exten = _555X.,4,Hangup
All SER calls 9xxx must go to asterisk, and it does, but I get the
following in aster log:
to 69.70.7.174:5060
Mar 6 18:41:36 WARNING[3539]: chan_sip.c:695 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for seqno 1
(Non-critical Response)
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav49, 0x814cb60
-- x=1, open writing:
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: gsm, 0x814d068
-- x=2, open writing:
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav, 0x8144980
Mar 6 18:41:45 WARNING[3539]: app.c:619 ast_play_and_record: No audio
available on SIP/69.70.x.x-08149a98??
-- User hung up
== Spawn extension (ser, 900, 1) exited non-zero on 'SIP/69.70.x.x-08149a98'
Destroying call '[EMAIL PROTECTED]'
If I use rewritehostport instead of forward, the call does not reach asterisk:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
t_relay()
break();
SER log:
4(11513) *** IP to IP call * 1(11506) ERROR:
t_forward_nonack: no branched for fwding
1(11506) ERROR: w_t_relay (failure mode): forwarding failed
3(11512) *** IP to IP call * 2(11509) Bye
Is there a way to do append_branch([EMAIL PROTECTED]) ?
Anyone did it? Reply pls with your config files!!
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Re: [Asterisk-Users] SER - Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)
If I use rewritehostport instead of forward, the call does not reach asterisk:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
t_relay()
break();
SER log:
Your failure route should read:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
append_branch(); ==YOU MISSED THIS
t_relay()
break();
--
Andres
Network Admin
http://www.telesip.net
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Re: [Asterisk-Users] SER/Asterisk consultants in Denver
Keith Burns wrote: Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Keith, My name is Michael Welter, and I have been installing Asterisk systems for two years. You may call me on 303-718-2804. Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER/Asterisk consultants in Denver
On Friday 18 February 2005 13:44, Michael Welter wrote: Keith Burns wrote: Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] [... quoted signature deleted ...] Hello Keith, My name is Michael Welter, and I have been installing Asterisk systems for two years. You may call me on 303-718-2804. You failed the intelligence test. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER/Asterisk consultants in Denver
What part of please contact me at [EMAIL PROTECTED] did you not understand? -Matthew - Original Message - From: Michael Welter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 18, 2005 7:44 AM Subject: Re: [Asterisk-Users] SER/Asterisk consultants in Denver Keith Burns wrote: Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello Keith, My name is Michael Welter, and I have been installing Asterisk systems for two years. You may call me on 303-718-2804. Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER/Asterisk consultants in Denver
Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER Asterisk Voicemail
Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk Voicemail
The sipsak way simply lites the MWI (or not) to indicate a message is waiting. You need to provide instructions in extensions.conf that route the call into voicemailmain. I use exten = 68007,1,VoicemailMain exten = 68007,2,Hangup -Steve Aisling O'Driscoll wrote: Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER + asterisk
Hi everybody! this is third day I'm supposed to work on some telecomunications solution. We have SIP Express Router to maintain and redirect incomming calls to asterisk. The problem is that we (i mean my company) have to run some prepaid solution with asterisk. I'm wondering if modified prepaid solution would work correctly in such environment? I'd be thankfull for any suggestions... regards Cyprian Zawadzki -- The paranoids' way... // Networked Electronic ___ ___ ___ ___ (___ ( ___ Unit Responsible for | )|___)| )| )| )|| | Online Troubleshooting | / |__ |__/ ||__/ |__ | |__ and Intensive Calculation ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ser + Asterisk DMZ
Hi all I am in this strange situation: we had ser configured to relay calls to numbers to asterisk extensions and all used to work nicely, with both ser and asterisk running on the same machine with public ip (ser on port 5060 and * on 5061). We had to move temporarily our server to another provider which put our server on a dmz, so that now we have our server with private ip but reachable from the outside via port forwarding on a public ip. Now every communication with asterisk is mute, calls are relayed by ser, connections estabilished, but no voice either with sip or with demo-echotest (* log says he is playing echotest but I can't hear anything!). I thought this was a dmz firewall + rtp problem but ports in rtp.conf are open with forwarding (udp). This is current network situation: myserver: private ip 10.0.0.229, ser running on port 5060, asterisk on 5061 (sip), rtp ports 5082-5092. Public ip 82.184.xx.xx with udp forwarding on above ports to myserver private ip Ser listens to private ip address and forwards to asterisk on private ip (can't forward to public address). bindaddress in sip.conf =private ip or 0.0.0.0 (mute in both cases), can't bind on public ip. Intresting part in sip.conf [general] port = 5061 ; Port to bind to ;bindaddr = 10.0.0.229 ; Address to bind SIP channel to bindaddr = 0.0.0.0 context = 82.184.xx.xx ; Default context for incoming calls srvlookup = no ; Enable DNS SRV lookups on outbound calls ;;; tried with or without following lines, still mute :-( autocreatepeer=yes externip=82.184.xx.xx register = asterisk:[EMAIL PROTECTED]/100 ;asterisk actually registers on ser! realm=82.184.xx.xx ;;; tried also with public ip host, nat=no, canreinvite=yes, type=peer [asterisk] type=friend secret=x username=asterisk host=10.0.0.229 nat=yes canreinvite=no ;dtmfmode=rfc2833 I hope someone can give me a hint to resolve this crappy situation thanks a lot -- Giovanni Balasso giaso apud supereva.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER + Asterisk Attended Call Transfer
Hi All ! First I was having trouble using attended call transfer using asterisk but thatnks to you guys I was able to make it work by adding 't' in options and applying the patch. Now I am using SER along with asterisk. SER works as SIP proxy and Asterisk performs all the necessary PBX functionalities. Can anybody guide me how to make attended call transfers work in this scenario if the SIP phone doesnot support attended call transfers. I'll be waiting for any valueable feedback. Thanks, Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER + Asterisk
Hi, since a while I try get Asterisk and SER work together. But until now I have no success. I want to use Asterisk as Gateway to the old telephone world. Is there somebody who can give me a small example of the ser.cfg and the Asterisk config files. This will be very nice. Thanks Bastian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER -- Asterisk , RTP Question.
Hello. I trying to use SER with Asterisk together. I have a question regarding the RTP path. If i make a call from one of my endpoints registered in SER Server, and that call in particular is forwarded to Asterisk and then to a PSTN-GW, Does the media goes through Asterisk?? is there a way to avoid this ?? Here es my extension.conf [TO_PSTN] exten = _00562.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _00562.,2,Hangup() Thanks in advance Ricardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER/Asterisk PSTN Call Transfer Issue.
Hi We have a phone system consisting primarily of SER and Asterisk, and are having trouble transferring inbound calls from the PSTN. We believe the problem is basically that because our phones register with SER, the Asterisk box never sees the call from the original callee to the new callee. i.e. caller -- PSTN -- Asterisk -- SER -- callee's SIP phone callee's SIP phone -- SER -- new callee's SIP phone when the original caller hangs up, the transfer is requested, however Asterisk does not know anything about the second call shown above, and so we see an error about supervised transfer requested; callid not found. We have found that if, in our SER config, we route 8extension through Asterisk, having stripped off the leading '8', and on to the SIP phone via SER, then it works, because Asterisk then 'knows' about the new call, since the second call above becomes callee's SIP phone -- SER -- Asterisk -- SER -- new callee's SIP phone Of course, this is a rather inelegant fix, especially when we actually have more than one PSTN gateway. Does anyone know if there is a more flexible fix, either on the SER or Asterisk side, that might fix our problem? thanks peter -- peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/ -- engineering hosting services for email, web and voip -- -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER + Asterisk
Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance, easier administration, or what? And, if it is better, then why to use Asterisk with it? Is it because of PSTN and/or IAX2 access, or the voicemail, IVR, meetme and such? Thanks, Marconi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ser+ asterisk
hi list,
i want to use the astersik in conjunction with
the ser
so i followed the instructions provided on the
voip-info.org site
but when calling from one user to another it gives me
problem in the asterisk cli that
failed user authentication
my aim of doing this is to use the asterisk with ser
for the features which it provides to the ser
registerd
users such as prepaid billing,siph323 gateway
so my configartions are as follows
ser.cfg of ser is
if (method==INVITE)
{
setflag(1);
if(uri=~^sip:2.*)
{
rewritehost(*.*.*.118);
t_relay();
break;
}
};
and then in the sip.conf
[general]
context=internal
autocreatepeer=yes
[Provider]
type=friend
username=XX
secret=XXX
host=*.*.*.119
and extension.conf
[EMAIL PROTECTED] asterisk]# cat extensions.conf
[globals]
SERADDRESS=*.*.*.119:5060
[general]
static=yes
writeprotect=no
[internal]
exten =
_XX,1,Dial(SIP/[EMAIL PROTECTED],20,r)
i have used * for innocence of ipaddress purpose
so please guide me
with regards
ravi kumar kura
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Go to: http://yahoo.shaadi.com/india-matrimony
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Re: [Asterisk-Users] SER Asterisk problem
Welesley Sibelson Dias wrote: Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI -- Executing Dial(SIP/16008-3d17, SIP/16007SIP/16006|20|tr) in new stack -- Called 16007 -- Called 16006 -- SIP/16007-8c24 is ringing -- SIP/16007-8c24 answered SIP/16008-3d17 -- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar 30 13:53:11 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 8 (Response) =3D=3D Spawn extension (sip, 1000, 1) exited non-zero on 'SIP/16008-3d17' Jadylson da Rocha Passos Bomfim I know of a GrandStream bug which generates a wrong ack to the 200 OK asterisk sends on connecting. SER drops this ack and asterisk drops the call, as it should. This is fixed in latest firmware image. Kind regards, Geert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk problem
Geert Nijpels wrote: -- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar I know of a GrandStream bug which generates a wrong ack to the 200 OK asterisk sends on connecting. SER drops this ack and asterisk drops the call, as it should. This is fixed in latest firmware image. At a guess it looks like the bridging is happening to a NAT'd SIP connection and doesn't like the non-routable IPs, stick the following line in your sip.conf for the phone notransfer=yes and see if that fixes your problem... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER Asterisk problem
Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI -- Executing Dial(SIP/16008-3d17, SIP/16007SIP/16006|20|tr) in new stack -- Called 16007 -- Called 16006 -- SIP/16007-8c24 is ringing -- SIP/16007-8c24 answered SIP/16008-3d17 -- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar 30 13:53:11 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 8 (Response) =3D=3D Spawn extension (sip, 1000, 1) exited non-zero on 'SIP/16008-3d17' Jadylson da Rocha Passos Bomfim Redevox Telecom Uberlandia +55 34 3234-7813 S=E3o Paulo +55 11 5055-6888 M=F3vel+55 34 9103-6854 =20 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.643 / Virus Database: 411 - Release Date: 25/3/2004 =20 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.643 / Virus Database: 411 - Release Date: 25/3/2004 =20 ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk
But now i'm stumbling on another problem.. Asterisk seems to want to send the SIP udp packets directly to the SIP clients. In the case of a SIP user/client behind a NAT, this obviously doesn't work. Have you tried reinvite=no in your [ser] section of sip.conf? P ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk
On Sat, 2004-01-17 at 01:33, [EMAIL PROTECTED] wrote: Thanks guys, thought SER had to 'register' to be able to use any Asterisk contexts. But just defining a new entry in the sip.conf with just context ip worked! But now i'm stumbling on another problem.. Asterisk seems to want to send the SIP udp packets directly to the SIP clients. In the case of a SIP user/client behind a NAT, this obviously doesn't work. My guess'd be that this is a problem of your ser configuration (such as a missing record_route()) rather than an issue with *. One thing I would take a look at, would be the incoming INVITE request using sip debug and check whether or not you you find a header field Record-Route: pointing to you SER proxy. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk
[EMAIL PROTECTED] wrote: I'm trying to bundle the powers of Asterisk and SER. Asterisk for pabx functionalities and termination to landline/PSTN, and SER as SIP Gateway/Proxy. With my current configuration the SIP user just adds 0 as a prefix to a number, and the call will go out to PSTN over Asterisk. For this to work I added the rewritehostport() function in SER to point to the Asterisk IP (different from the SER ip). At the moment I just added the following line to my sip.conf (in the [general] section): context=from-sip But my question here is, everyone can (ab)use this by connecting directly to the Asterisk IP. This way they can easily dial out over the PSTN network. Hi, This sounds a very similar problem to me, despite the different context. The 'default' context in the [general] section shouldn't be (ab)usable - set this to something like [bogon-calls]. Then set up a specific peer lower down: [ser] context=sip-legal host=y.y.y.y ; IP address of SER Se this Wiki page for more flesh of my (not yet fully working!) configs: http://voip-info.org/wiki-Asterisk+cisco+FXO Good luck! Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk
Yes, you can keep non-authorized SIP callers from accessing the PSTN by setting up the .conf file correctly as below but you can also run a fire wall on the box that Asterisk runs on. Firewall off SIP ports except for if they come from your SER server. This will work even if Asterisk is broken or misconfigured. Security sould always be applied in multiple layers: use both a belt and suspenders I like the shorewall firewall script. configuration is conceptually easy it uses the cisco-like idea of zones. --- Fran Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: I'm trying to bundle the powers of Asterisk and SER. Asterisk for pabx functionalities and termination to landline/PSTN, and SER as SIP Gateway/Proxy. With my current configuration the SIP user just adds 0 as a prefix to a number, and the call will go out to PSTN over Asterisk. For this to work I added the rewritehostport() function in SER to point to the Asterisk IP (different from the SER ip). At the moment I just added the following line to my sip.conf (in the [general] section): context=from-sip But my question here is, everyone can (ab)use this by connecting directly to the Asterisk IP. This way they can easily dial out over the PSTN network. Hi, This sounds a very similar problem to me, despite the different context. The 'default' context in the [general] section shouldn't be (ab)usable - set this to something like [bogon-calls]. Then set up a specific peer lower down: [ser] context=sip-legal host=y.y.y.y ; IP address of SER Se this Wiki page for more flesh of my (not yet fully working!) configs: http://voip-info.org/wiki-Asterisk+cisco+FXO Good luck! Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk
Thanks guys, thought SER had to 'register' to be able to use any Asterisk contexts. But just defining a new entry in the sip.conf with just context ip worked! But now i'm stumbling on another problem.. Asterisk seems to want to send the SIP udp packets directly to the SIP clients. In the case of a SIP user/client behind a NAT, this obviously doesn't work. SER is configured to use the wonderful RTPProxy + SER nathelper module, and this works flawlessly (using the rewritehostport function). But when I try to call a phone number on the PSTN network from a SIP client behind NAT, SER sends the invites to Asterisk, and Asterisk makes an outbound call to the phone number, the phone rings, but when the pstn user picks up the phone, no sound, and after a while (couple of seconds), the call is dropped. Asterisk spews out the following warning, chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 29898 (Response) Tried searching on the voip-info wiki and mailinglists, but didn't find a way to force Asterisk to use a SIP proxy/SER. Any ideas ? On Fri, Jan 16, 2004 at 12:12:14AM -0800, Chris Albertson wrote: Yes, you can keep non-authorized SIP callers from accessing the PSTN by setting up the .conf file correctly as below but you can also run a fire wall on the box that Asterisk runs on. Firewall off SIP ports except for if they come from your SER server. --- Fran Boon [EMAIL PROTECTED] wrote: [ser] context=sip-legal host=y.y.y.y ; IP address of SER Se this Wiki page for more flesh of my (not yet fully working!) configs: http://voip-info.org/wiki-Asterisk+cisco+FXO ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
