[asterisk-users] SIP over TCP/TLS for 1.4 branch
Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Bryan Ekelund WHI Solutions, Inc. bekel...@whisolutions.com STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- 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] SIP over TCP/TLS for 1.4 branch
On Mon, Nov 23, 2009 at 4:05 PM, Ekelund, Bryan bekel...@whisolutions.com wrote: Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Just file a bug with Microsoft and ask them to support SIP over UDP. Problem solved. ___ -- 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] SIP over TCP/TLS for 1.4 branch
And take the easy way out? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, November 23, 2009 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP/TLS for 1.4 branch On Mon, Nov 23, 2009 at 4:05 PM, Ekelund, Bryan bekel...@whisolutions.com wrote: Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Just file a bug with Microsoft and ask them to support SIP over UDP. Problem solved. ___ -- 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 STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify WHI Solutions immediately at g...@whisolutions.com, and destroy all copies of this message and any attachments. ___ -- 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] SIP over TCP
That's excellent! So in theory one could not make Asterisk compatible SIP softphone in Flash (since Flash only supports TCP). Nice... BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Tuesday, June 24, 2008 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP No, TCP for media as well. I though that was the whole point of SIP over TCP. Michael On Mon, 23 Jun 2008 16:59:00 +0200, Asterisk wrote: Hi, But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right? BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, June 22, 2008 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
On Tue, 24 Jun 2008 12:36:52 +0200, Asterisk wrote: That's excellent! So in theory one could not make Asterisk compatible SIP softphone in Flash (since Flash only supports TCP). Nice... BR, Alex I beleive that this hs already been done, although I can't recall by whom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Tuesday, June 24, 2008 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP No, TCP for media as well. I though that was the whole point of SIP over TCP. Michael On Mon, 23 Jun 2008 16:59:00 +0200, Asterisk wrote: Hi, But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right? BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, June 22, 2008 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
Michael Graves [EMAIL PROTECTED] writes: No, TCP for media as well. I though that was the whole point of SIP over TCP. Hopefully not. RTP over TCP would be entirely pointless. RTP needs packetization, doesn't mind packet loss (within reason) but hates retransmissions. TCP doesn't provide packetization, guarantees against packet loss, but retransmits. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
Hi, But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right? BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, June 22, 2008 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
No, TCP for media as well. I though that was the whole point of SIP over TCP. Michael On Mon, 23 Jun 2008 16:59:00 +0200, Asterisk wrote: Hi, But you can only route SIP signalization over TCP. Audio stream must still go thru UDP, right? BR, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, June 22, 2008 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP over TCP On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over TCP
Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
On 6/22/08, Michael Graves [EMAIL PROTECTED] wrote: Ok, so now that it's possible to implement SIP over TCP instead of UDP why would I want to do this? Beyond simply integration with M$ OCS. And what are the implications for management of QoS? I would expect that lost packets would be less of a factor. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Michael, The main advantages for SIP over TCP that I know of (in no particular order): - Better compatibility with NAT devices (it seems some of them don't do UDP well) - Support for TLS - Support for packet fragmentation (to support large/diverse SDPs, headers, etc) I'm sure there are other ones but that's all I can think of this early on a Sunday morning... -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP development in 1.6 branch?
On Thu, Jun 19, 2008 at 3:50 PM, Paul Belanger [EMAIL PROTECTED] wrote: List, Could anybody speak to the status of development in 1.6 branch? I know support for SIP over TCP is pretty new / experimental but it seems active development of it has slowed or stopped in recent months. Is that a correct statement? Is SIP over TCP more a community project or something headed from Digium? I only ask to get a pulse of its status; not harp or demand people to work on it. Like everybody else, we have some dependencies on SIP over TCP, and have a few bugs open against it. Personally, I would love to help develop or submit patches for the bugs but would need a mentor for that. Either way, just looking to get some more info about the development status of it. I can't speak about the status of SIP/TCP development in Asterisk, but I can say the following: . I've tested Asterisk SIP/TCP and SIP/TLS against a variety of SIP implementations (acting as SIP peers) in a lab setting and things look okay. . I ran into a bug when I register a SIP end-point using SIP/TCP (http://bugs.digium.com/view.php?id=12282). . I think some of the challenges relating to deploying Asterisk SIP/TCP in production environments will be - connection management and NAT traversal. I think certain design thought needs to be put in SIP/TCP feature design to combat these issues. -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over TCP development in 1.6 branch?
List, Could anybody speak to the status of development in 1.6 branch? I know support for SIP over TCP is pretty new / experimental but it seems active development of it has slowed or stopped in recent months. Is that a correct statement? Is SIP over TCP more a community project or something headed from Digium? I only ask to get a pulse of its status; not harp or demand people to work on it. Like everybody else, we have some dependencies on SIP over TCP, and have a few bugs open against it. Personally, I would love to help develop or submit patches for the bugs but would need a mentor for that. Either way, just looking to get some more info about the development status of it. Thanks again, PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP development in 1.6 branch?
On Thu, 2008-06-19 at 15:50 -0400, Paul Belanger wrote: List, Could anybody speak to the status of development in 1.6 branch? I know support for SIP over TCP is pretty new / experimental but it seems active development of it has slowed or stopped in recent months. Is that a correct statement? Is SIP over TCP more a community project or something headed from Digium? I only ask to get a pulse of its status; not harp or demand people to work on it. Like everybody else, we have some dependencies on SIP over TCP, and have a few bugs open against it. Personally, I would love to help develop or submit patches for the bugs but would need a mentor for that. Either way, just looking to get some more info about the development status of it. Can be brief about it: it just works! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over TCP
Hello guys, I thought it would be neat if we had a SIP client for Asterisk working in Adobe Flash, but as far as I know, Flash only supports TCP. I know that Asterisk (at least v1.6) can handle SIP communication over TCP, but I was wondering is there a possibility to route audio stream over TCP too? Regards, Alex ___ -- 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] SIP over TCP
On 13/02/2008, Raj Jain [EMAIL PROTECTED] wrote: SIP over TCP is included in 1.6. http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co Thanks all! :o) ___ -- 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] SIP over TCP
I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. ___ -- 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] SIP over TCP
Looks like it is part of the 1.6 Beta. From the Change Log: 2008-01-18 22:04 + [r99080-99085] Russell Bryant [EMAIL PROTECTED] * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) On Feb 13, 2008 4:21 PM, Razza [EMAIL PROTECTED] wrote: I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. ___ -- 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] SIP over TCP
SIP over TCP is included in 1.6. http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co On Feb 13, 2008 5:21 PM, Razza [EMAIL PROTECTED] wrote: I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. ___ -- 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 -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org ___ -- 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] SIP over TCP
Hy all, I am Marta from Spain. I have just start working and my first project is with Asterisk. And I am a bit lost I am interested in testing sip over tcp. I have read in http://www.sineapps.com/news.php?rssid=1777 that there are some implementations. Is this the most recent version? Are there any other developments in this area? Is it really working? How can I test it? Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP over TCP
SPANISH IOLLOWS...sorry about that folks! -- hola Marta, bienvenida al mundo de las PBX y Asterisk. Tambien enhorabuena en tu andadura profesional. Como se que al principio es dificil no me importaria echarte una mano para que despegue tu proyecto laboral :) Toma nota de mi email y escribeme sin compromiso. Un saludo. Jonathan GF On 9/21/07, dadsadsadf dsadasdsa [EMAIL PROTECTED] wrote: Hy all, I am Marta from Spain. I have just start working and my first project is with Asterisk. And I am a bit lost… I am interested in testing sip over tcp. I have read in http://www.sineapps.com/news.php?rssid=1777 that there are some implementations. Is this the most recent version? Are there any other developments in this area? Is it really working? How can I test it? Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] SIP over TCP: latest news?
Mimmus wrote: Hi, I know it is a FAQ but I'm interested in latest news (if any...) about SIP over TCP support in Asterisk. I found this: https://savannah.nongnu.org/projects/asterisk-tcp/ but I'm not able to understand if project is active and what is its level of development. Thanks Mimmus The patch has been added to Digium BTS, and it's waiting for the new socket interface according to bug report #4903. http://bugs.digium.com/view.php?id=4903 /Mikael ___ --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] SIP over TCP: latest news?
Hi, I know it is a FAQ but I'm interested in latest news (if any...) about SIP over TCP support in Asterisk. I found this: https://savannah.nongnu.org/projects/asterisk-tcp/ but I'm not able to understand if project is active and what is its level of development. Thanks Mimmus ___ --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
