Re: [asterisk-users] Moving User Agent To Remote Location
Hello Ishfaq, and Isrlgb, The canreinvite value for UA friend entries are set to no, and for the OpenSIPS peer entry it's set to yes. I do have esternip and localnet cid set in sip.conf. I did not want to start a new email, but part of my problem right now is that OpenSIPS is in charge of performing the AUTH and REGISTER. This is fine for peers with static host definition, but not for the dynamic ones. Is it possible to have the fullcontact realtime info in sip_buddies populated upon initial INVITE? This is my first problem right now. After that will come RTP, and Codec issues... PS I have seen fullcontact info get populated with the correctly in the past, just can't get it to do it every time Thanks for your help!!! Nick. On 1/4/13, isr...@gmail.com isr...@gmail.com wrote: Did you set externip and localnet in your sip conf ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote: Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. Hi Is your directmedia/canreinvite (depending on version) set to no? Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote: Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. Hi Is your directmedia/canreinvite (depending on asterisk version) set to no? Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
Did you set externip and localnet in your sip conf ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Moving User Agent To Remote Location
Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, January 03, 2013 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Moving User Agent To Remote Location Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. I'm going to vote for the RTP issue. If you are establishing a call but have no audio, you are getting the 5060 port, but not the 1-2 range that RTP normally expects. A better practice is to allocate 4 ports per line you expect to use in rtp.conf (1-2 would allow 2500 lines; much more that most folks need and more holes to monitor). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 Just for testing purposes, and deduce my way from there? Right now I am trying to call the phone from my softphone. That being said, I currently I am not able to reach the remote extension from my location here. I think this is the root of the problem here: -- Executing [1003@context-from-toronto:1] Dial(SIP/OpenSIPS-0009, SIP/1003, 20) in new stack Really destroying SIP dialog '06775f8653ff88b47cfa9ec123abdd89@127.0.0.1:0' Method: INVITE [Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [1003@context-from-toronto:2] Wait(SIP/OpenSIPS-0009, 1) in new stack -- Executing [1003@context-from-toronto:3] Answer(SIP/OpenSIPS-0009, ) in new stack Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP It's actually not able to create the SIP channel between the two UA? I will try taking opensips out of the picture and work outwards... N. On 1/3/13, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, January 03, 2013 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Moving User Agent To Remote Location Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. I'm going to vote for the RTP issue. If you are establishing a call but have no audio, you are getting the 5060 port, but not the 1-2 range that RTP normally expects. A better practice is to allocate 4 ports per line you expect to use in rtp.conf (1-2 would allow 2500 lines; much more that most folks need and more holes to monitor). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 do you mean 1_000_8 ? Markus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 do you mean 1_000_8 ? Markus I think he means 10007. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
On Thu, Jan 3, 2013 at 2:21 PM, Nick Khamis sym...@gmail.com wrote: [Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) Can you check that the registration is happening correctly? Try `sip show peers` or `sip show peer [destination phone]`. Usually cause 20 means the peer isn't registered. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
Oooops yes of course 10004-10007!! Simple math does not come easy anymore... Anyhow, I singled out Opensips and I have two way audio form UA(local) - UA(remote) but not from UA - Siptrunk. That being said maybe a small diagram of the architecture. Please don't laugh!!! :) I know having a block of static IPs would make like easier however UA (Remote) - Router (Remote) - Internet - Router (Local) - OpenSIPS+RTPProxy - Asterisk Port forwarding (Remote): 5060, and 1-5 to UA Port Forwarding (Local): 5060. and 1-5 to OpenSIPS) No Audio Port Forwarding (Local): 5060. and 1-5 directly to Asterisk Two Way Audio Cheers Guys! Nick On 1/3/13, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
Just for grins, run netstat -anp on the call using just Asterisk and then again with OpenSIPS in the mix. It sounds like OpenSIPS or your RTPproxy is block the audio channels. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
To Answer Some of You Questions: Please not that I replace the true domain wtih example, and the true ip for the remote UA with public-ip. Nothing against no one here, just don't know who else would read this email in the future!!! PS: The public IP of the remote UA is correct. SIP Show Peers: Name/username HostDyn Forcerport ACL Port Status Realtime 1002/1002@toronto.example. 192.168.2.13 N5060 UNKNOWNCached RT 1003/1003@toronto.example. -public-ip- D N5060 OK (86 ms) Cached RT Peers look registered correctly. This has now become a sip proxy issue :S. Thank you so much for your time guys!!! N. On 1/3/13, Nick Khamis sym...@gmail.com wrote: Oooops yes of course 10004-10007!! Simple math does not come easy anymore... Anyhow, I singled out Opensips and I have two way audio form UA(local) - UA(remote) but not from UA - Siptrunk. That being said maybe a small diagram of the architecture. Please don't laugh!!! :) I know having a block of static IPs would make like easier however UA (Remote) - Router (Remote) - Internet - Router (Local) - OpenSIPS+RTPProxy - Asterisk Port forwarding (Remote): 5060, and 1-5 to UA Port Forwarding (Local): 5060. and 1-5 to OpenSIPS) No Audio Port Forwarding (Local): 5060. and 1-5 directly to Asterisk Two Way Audio Cheers Guys! Nick On 1/3/13, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
