I have tried several SIP clients with the same results. As mentioned I have checked to make sure NAT=YES is in my config files. Are there any port's that have to be looked at in the router at the server end other than the SIP port? All of the clients I have tried work behind the router. Bill Hurlock From: Bill Hurlock Sent: Sunday, February 09, 2014 6:27 PM To: 'app_rpt-users@ohnosec.org' Subject: SIP Help Needed I use CSIP Simple on my cell Phone. When I'm on my local WiFi all is well. If I go on the road and log into a WiFi system somewhere. I can register the account but it only seems to be a inward path only. When I watch CLI remotely, I see the connect, Answer, Wait, Playback going out OK but hear nothing on the Cell phone end. Than I see some CLI error messages that seem to point to the fact it isn't getting info back from the Cell phone end after which it processes a disconnect. Is this a port issue at the distant end of the link or something I have wrong on the Server end?? As I mentioned it works perfect when logged in on WiFI on the same network as the server. IAX clients seem to be having a similar issue connecting from the cell however IAXRpt works OK from the cell phone network using my PC to connect home to the server. Any ideas on this as I know people are using SIP OK out in the field. Bill Hurlock CPCommunications 856-234-1661 Office 856-264-1010<http://www.totalrf.com/> Cell www.cpcomms.com<http://www.cpcomms.com/> WA2TQI
You also need the router to handle the RTP ports as that's where the audio is carried. The actual ports (a pair on each call) used are assigned pseudo randomly within the range defined in rtp.conf. And of course, the client needs the same range set. On February 11, 2014 7:24:01 AM EST, Bill Hurlock <bill.hurlock@cpcomms.com> wrote:
I have tried several SIP clients with the same results. As mentioned I have checked to make sure NAT=YES is in my config files. Are there any port's that have to be looked at in the router at the server end other than the SIP port? All of the clients I have tried work behind the router.
Bill Hurlock
From: Bill Hurlock Sent: Sunday, February 09, 2014 6:27 PM To: 'app_rpt-users@ohnosec.org' Subject: SIP Help Needed
I use CSIP Simple on my cell Phone. When I'm on my local WiFi all is well. If I go on the road and log into a WiFi system somewhere. I can register the account but it only seems to be a inward path only. When I watch CLI remotely, I see the connect, Answer, Wait, Playback going out OK but hear nothing on the Cell phone end. Than I see some CLI error messages that seem to point to the fact it isn't getting info back from the Cell phone end after which it processes a disconnect. Is this a port issue at the distant end of the link or something I have wrong on the Server end?? As I mentioned it works perfect when logged in on WiFI on the same network as the server. IAX clients seem to be having a similar issue connecting from the cell however IAXRpt works OK from the cell phone network using my PC to connect home to the server. Any ideas on this as I know people are using SIP OK out in the field.
Bill Hurlock CPCommunications 856-234-1661 Office 856-264-1010<http://www.totalrf.com/> Cell www.cpcomms.com<http://www.cpcomms.com/> WA2TQI
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I looked through the example rtp.conf and can’t see any way to define port usage as you mentioned. Would you have a sample file I could look at? Bill Hurlock From: Robert A. Poff [mailto:wb3awj@comcast.net] Sent: Tuesday, February 11, 2014 8:09 AM To: Bill Hurlock; app_rpt-users@ohnosec.org Subject: Re: [App_rpt-users] SIP Help Needed You also need the router to handle the RTP ports as that's where the audio is carried. The actual ports (a pair on each call) used are assigned pseudo randomly within the range defined in rtp.conf. And of course, the client needs the same range set. On February 11, 2014 7:24:01 AM EST, Bill Hurlock <bill.hurlock@cpcomms.com<mailto:bill.hurlock@cpcomms.com>> wrote: I have tried several SIP clients with the same results. As mentioned I have checked to make sure NAT=YES is in my config files. Are there any port’s that have to be looked at in the router at the server end other than the SIP port? All of the clients I have tried work behind the router. Bill Hurlock From: Bill Hurlock Sent: Sunday, February 09, 2014 6:27 PM To: 'app_rpt-users@ohnosec.org' Subject: SIP Help Needed I use CSIP Simple on my cell Phone. When I’m on my local WiFi all is well. If I go on the road and log into a WiFi system somewhere. I can register the account but it only seems to be a inward path only. When I watch CLI remotely, I see the connect, Answer, Wait, Playback going out OK but hear nothing on the Cell phone end. Than I see some CLI error messages that seem to point to the fact it isn’t getting info back from the Cell phone end after which it processes a disconnect. Is this a port issue at the distant end of the link or something I have wrong on the Server end?? As I mentioned it works perfect when logged in on WiFI on the same network as the server. IAX clients seem to be having a similar issue connecting from the cell however IAXRpt works OK from the cell phone network using my PC to connect home to the server. Any ideas on this as I know people are using SIP OK out in the field. Bill Hurlock CPCommunications 856-234-1661 Office 856-264-1010<http://www.totalrf.com/> Cell www.cpcomms.com<http://www.cpcomms.com/> WA2TQI ________________________________ App_rpt-users mailing list App_rpt-users@ohnosec.org<mailto:App_rpt-users@ohnosec.org> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users To unsubscribe from this list please visit http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the "Unsubscribe or edit options button" You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem. -- Sent from my Android phone with K-9 Mail. Please excuse my brevity and typing errors.
Here's what is in mine verbatim. Ah....... I just looked into an ACID install of mine. Didn't see rtp.conf there at all. You may need to create it. When I first starting to add extensions, I used the Asterisk book as a guide. Very helpful. Here's the online version : http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.ht... It covers Asterisk 1.8, but most of the config stuff still applies to 1.4 I find. Anyhow, here's what I have in rtp.conf on my node that has SIP phones attached: ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; These are the addresses where your system will RECEIVE audio and video stream$ ; If you have connections across a firewall, make sure that these are open. ; rtpstart=10000 rtpend=10999 And here's a link to a page concerning rtp.conf : http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf Hope this helps.
Right after I hit send I saw that in SRC there was a 2nd rtp.conf example that listed what you have sent. I have added this to my RTP.conf file and and added these ports to my router config. I’ll be out of the house and will check later today. Thanks for your help. Bill Hurlock From: Robert A. Poff WB3AWJ [mailto:wb3awj@comcast.net] Sent: Wednesday, February 12, 2014 11:10 AM To: Bill Hurlock Cc: app rpt-users Subject: Re: [App_rpt-users] SIP Help Needed Here's what is in mine verbatim. Ah....... I just looked into an ACID install of mine. Didn't see rtp.conf there at all. You may need to create it. When I first starting to add extensions, I used the Asterisk book as a guide. Very helpful. Here's the online version : http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.ht... It covers Asterisk 1.8, but most of the config stuff still applies to 1.4 I find. Anyhow, here's what I have in rtp.conf on my node that has SIP phones attached: ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; These are the addresses where your system will RECEIVE audio and video stream$ ; If you have connections across a firewall, make sure that these are open. ; rtpstart=10000 rtpend=10999 And here's a link to a page concerning rtp.conf : http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf Hope this helps.
I hate to re-hash this, but setting up IAX for your smart phone will save you lots of time and grief. The ports are already open and don’t have the routing issues. Setup Steps. 1. cd /etc/asterisk 2. mkdir custom 3. cd custom 4. nano iax.conf 5. paste this into the file. [kc7wsu] ; Change this to your desired username for the client. I use callsign. type=friend context=smartphone host=dynamic auth=md5 secret=Passw0rd1 ; Change to your desired password disallow=all allow=ulaw allow=g726aal2 allow=gsm codecpriority=host context=radio-secure transfer=no callerid="KC7WSU" ; Change to the callsign you would like to show up. 6. Save and exit file. (CTRL+X), Yes 7. nano extensions.conf 8. paste this into the file. Update 29205 with your node number. [smartphone] exten=29205,1,answer() exten=29205,n,Playback(rpt/node) exten=29205,n,Playback(digits/2) exten=29205,n,Playback(digits/9) exten=29205,n,Playback(digits/2) exten=29205,n,Playback(digits/0) exten=29205,n,Playback(digits/5) exten=29205,n,rpt(29205|P) 9. Save and exit file. (CTRL+X), Yes 10. astres.sh Regards, Chris Andrist, KC7WSU On Feb 12, 2014, at 10:00 AM, Bill Hurlock <bill.hurlock@cpcomms.com> wrote:
Right after I hit send I saw that in SRC there was a 2nd rtp.conf example that listed what you have sent. I have added this to my RTP.conf file and and added these ports to my router config. I’ll be out of the house and will check later today. Thanks for your help.
Bill Hurlock
From: Robert A. Poff WB3AWJ [mailto:wb3awj@comcast.net] Sent: Wednesday, February 12, 2014 11:10 AM To: Bill Hurlock Cc: app rpt-users Subject: Re: [App_rpt-users] SIP Help Needed
Here's what is in mine verbatim. Ah....... I just looked into an ACID install of mine. Didn't see rtp.conf there at all. You may need to create it.
When I first starting to add extensions, I used the Asterisk book as a guide. Very helpful. Here's the online version :http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.ht... It covers Asterisk 1.8, but most of the config stuff still applies to 1.4 I find.
Anyhow, here's what I have in rtp.conf on my node that has SIP phones attached: ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; These are the addresses where your system will RECEIVE audio and video stream$ ; If you have connections across a firewall, make sure that these are open. ;
rtpstart=10000 rtpend=10999
And here's a link to a page concerning rtp.conf :http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf
Hope this helps.
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Thanks for the help. I now have IAX and SIP working OK. IAX issue wasn’t port defines, it needed type to be type=friend instead of type=user. Also needed permit=0.0.0.0/0.0.0.0 after the deny statement in the SIP user statement. Bit of a problem on 3g/4g with excessive jitter sometimes. Bill Hurlock From: Robert A. Poff [mailto:wb3awj@comcast.net] Sent: Tuesday, February 11, 2014 8:09 AM To: Bill Hurlock; app_rpt-users@ohnosec.org Subject: Re: [App_rpt-users] SIP Help Needed You also need the router to handle the RTP ports as that's where the audio is carried. The actual ports (a pair on each call) used are assigned pseudo randomly within the range defined in rtp.conf. And of course, the client needs the same range set. On February 11, 2014 7:24:01 AM EST, Bill Hurlock <bill.hurlock@cpcomms.com<mailto:bill.hurlock@cpcomms.com>> wrote: I have tried several SIP clients with the same results. As mentioned I have checked to make sure NAT=YES is in my config files. Are there any port’s that have to be looked at in the router at the server end other than the SIP port? All of the clients I have tried work behind the router. Bill Hurlock From: Bill Hurlock Sent: Sunday, February 09, 2014 6:27 PM To: 'app_rpt-users@ohnosec.org' Subject: SIP Help Needed I use CSIP Simple on my cell Phone. When I’m on my local WiFi all is well. If I go on the road and log into a WiFi system somewhere. I can register the account but it only seems to be a inward path only. When I watch CLI remotely, I see the connect, Answer, Wait, Playback going out OK but hear nothing on the Cell phone end. Than I see some CLI error messages that seem to point to the fact it isn’t getting info back from the Cell phone end after which it processes a disconnect. Is this a port issue at the distant end of the link or something I have wrong on the Server end?? As I mentioned it works perfect when logged in on WiFI on the same network as the server. IAX clients seem to be having a similar issue connecting from the cell however IAXRpt works OK from the cell phone network using my PC to connect home to the server. Any ideas on this as I know people are using SIP OK out in the field. Bill Hurlock CPCommunications 856-234-1661 Office 856-264-1010<http://www.totalrf.com/> Cell www.cpcomms.com<http://www.cpcomms.com/> WA2TQI ________________________________ App_rpt-users mailing list App_rpt-users@ohnosec.org<mailto:App_rpt-users@ohnosec.org> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users To unsubscribe from this list please visit http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the "Unsubscribe or edit options button" You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem. -- Sent from my Android phone with K-9 Mail. Please excuse my brevity and typing errors.
participants (4)
-
Bill Hurlock -
Chris Andrist -
Robert A. Poff -
Robert A. Poff WB3AWJ