Re: [asterisk-users] stanaphone issues. can someone verify my config?
Sorry, being really busy recently and only now have the time to dedicate to this (finished uni for the summer break) The asterisk is running on the machine that does the nat for the network here at home, it is set as the dmz on the adsl router so all ports should be coming into it. I have done a sip debug and copied it (and sanitized it) and put it here - well up till all the retrys start to appear. ; richards stanaphone incoming ;register = 0892: (MY PASSWORD)@sip.stanaphone.com/0892 register = 0892: (MY PASSWORD)@sip.stanaphone.com/101 (tried it both ways, having the stanaphone number as extension makes no difference) 101 just goto's a thing that answers, plays a voice and thenputs it on hold which work on all other sip providers. [stanaphone-richard] type=friend username=0892 secret=(MY PASSWORD) host=sip.stanaphone.com allow=all ;allow=g729 ;allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Monday, September 24, 2007 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stanaphone issues. can someone verify my config? any firewall in between? On 9/18/07, Richard [EMAIL PROTECTED] wrote: Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0 http://192.168.0.0/255.255.0.0 ; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [ [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [ [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error) The lack of RTP everywhere makes it look to be a nat issue, but I have done everything I can think of to have that work, and the config is the same other then host, username and password on italk which is working fine. I have googled for the Maximum retries exceeded on transmission - I could only see some stuff related
Re: [asterisk-users] stanaphone issues. can someone verify my config?
And I forgot the pastebin link - DOH - http://pastebin.com/m782bcee4 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Sent: Monday, December 17, 2007 12:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] stanaphone issues. can someone verify my config? Sorry, being really busy recently and only now have the time to dedicate to this (finished uni for the summer break) The asterisk is running on the machine that does the nat for the network here at home, it is set as the dmz on the adsl router so all ports should be coming into it. I have done a sip debug and copied it (and sanitized it) and put it here - well up till all the retrys start to appear. ; richards stanaphone incoming ;register = 0892: (MY PASSWORD)@sip.stanaphone.com/0892 register = 0892: (MY PASSWORD)@sip.stanaphone.com/101 (tried it both ways, having the stanaphone number as extension makes no difference) 101 just goto's a thing that answers, plays a voice and thenputs it on hold which work on all other sip providers. [stanaphone-richard] type=friend username=0892 secret=(MY PASSWORD) host=sip.stanaphone.com allow=all ;allow=g729 ;allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Monday, September 24, 2007 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stanaphone issues. can someone verify my config? any firewall in between? On 9/18/07, Richard [EMAIL PROTECTED] wrote: Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0 http://192.168.0.0/255.255.0.0 ; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [ [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [ [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error
Re: [asterisk-users] stanaphone issues. can someone verify my config?
any firewall in between? On 9/18/07, Richard [EMAIL PROTECTED] wrote: Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error) The lack of RTP everywhere makes it look to be a nat issue, but I have done everything I can think of to have that work, and the config is the same other then host, username and password on italk which is working fine. I have googled for the Maximum retries exceeded on transmission - I could only see some stuff related to broken sip phones, not a voip server. Alternativly, since it seems that stanaphone is a bit of a hit and miss from some other reading, is there any other functional US inwards provider for free that doesn't need a credit card that works well with asterisk? The softphone works, but I really need to get it going to my phones in the house instead. Soft client was closed when testing the asterisk. Many thanks. Richard Malcolm-Smith... ___ 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
[asterisk-users] stanaphone issues. can someone verify my config?
Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error) The lack of RTP everywhere makes it look to be a nat issue, but I have done everything I can think of to have that work, and the config is the same other then host, username and password on italk which is working fine. I have googled for the Maximum retries exceeded on transmission - I could only see some stuff related to broken sip phones, not a voip server. Alternativly, since it seems that stanaphone is a bit of a hit and miss from some other reading, is there any other functional US inwards provider for free that doesn't need a credit card that works well with asterisk? The softphone works, but I really need to get it going to my phones in the house instead. Soft client was closed when testing the asterisk. Many thanks. Richard Malcolm-Smith... ___ 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
