RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
If I am understanding correctly your setup looks like this.. {Asterisk}--[NAT]--Internet--[NAT]--{X-Lite} If this is correct then you are going to have major problems getting it to work.. Your RTP traffic is going to get very confused.. You need to get Asterisk onto a Public IP address.. I have seen many try and get the double NAT setup to work but I haven't yet heard of anyone getting it right.. Hi, thanks for that. after implementing yours and wipeout's suggestions (thank you both), x-lite changed its default codecs to G711a. which is great... and a way forward. but it still does not play sound when the 1000 is dialed. my * is behind nat. and my test pc is as well. Here are my settings: sip.conf [senad] type=friend secret=blah host=dynamic dtmfmode=inband thanks senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Littlepage Sent: 08 August 2003 19:50 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue Change the allow=all in sip.conf to allow=alaw and see if that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, August 08, 2003 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue Hi, X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but i can not hear it. Tried many times with the same result. After quite few tries * complains about: - WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 43 (Response) WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to determine sequence number from '' - Has anyone had the same problem? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
Thanks. I will move * on one of our internet servers. That should take away all NAT/PAT issues. Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: 09 August 2003 08:37 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue If I am understanding correctly your setup looks like this.. {Asterisk}--[NAT]--Internet--[NAT]--{X-Lite} If this is correct then you are going to have major problems getting it to work.. Your RTP traffic is going to get very confused.. You need to get Asterisk onto a Public IP address.. I have seen many try and get the double NAT setup to work but I haven't yet heard of anyone getting it right.. Hi, thanks for that. after implementing yours and wipeout's suggestions (thank you both), x-lite changed its default codecs to G711a. which is great... and a way forward. but it still does not play sound when the 1000 is dialed. my * is behind nat. and my test pc is as well. Here are my settings: sip.conf [senad] type=friend secret=blah host=dynamic dtmfmode=inband thanks senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Littlepage Sent: 08 August 2003 19:50 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue Change the allow=all in sip.conf to allow=alaw and see if that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, August 08, 2003 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue Hi, X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but i can not hear it. Tried many times with the same result. After quite few tries * complains about: - WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 43 (Response) WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to determine sequence number from '' - Has anyone had the same problem? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
Change the allow=all in sip.conf to allow=alaw and see if that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, August 08, 2003 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue Hi, X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but i can not hear it. Tried many times with the same result. After quite few tries * complains about: - WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 43 (Response) WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to determine sequence number from '' - Has anyone had the same problem? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
You might look into PAT to forward your RTP traffic to the asterisk boxes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Saturday, August 09, 2003 2:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue If I am understanding correctly your setup looks like this.. {Asterisk}--[NAT]--Internet--[NAT]--{X-Lite} If this is correct then you are going to have major problems getting it to work.. Your RTP traffic is going to get very confused.. You need to get Asterisk onto a Public IP address.. I have seen many try and get the double NAT setup to work but I haven't yet heard of anyone getting it right.. Hi, thanks for that. after implementing yours and wipeout's suggestions (thank you both), x-lite changed its default codecs to G711a. which is great... and a way forward. but it still does not play sound when the 1000 is dialed. my * is behind nat. and my test pc is as well. Here are my settings: sip.conf [senad] type=friend secret=blah host=dynamic dtmfmode=inband thanks senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Littlepage Sent: 08 August 2003 19:50 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue Change the allow=all in sip.conf to allow=alaw and see if that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, August 08, 2003 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue Hi, X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but i can not hear it. Tried many times with the same result. After quite few tries * complains about: - WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 43 (Response) WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to determine sequence number from '' - Has anyone had the same problem? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
Thanks WEll, not sure how to do that.Anyway. I will move * on one of our internet servers. That should take away all NAT/PAT issues. Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Littlepage Sent: 09 August 2003 11:53 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue You might look into PAT to forward your RTP traffic to the asterisk boxes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Saturday, August 09, 2003 2:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue If I am understanding correctly your setup looks like this.. {Asterisk}--[NAT]--Internet--[NAT]--{X-Lite} If this is correct then you are going to have major problems getting it to work.. Your RTP traffic is going to get very confused.. You need to get Asterisk onto a Public IP address.. I have seen many try and get the double NAT setup to work but I haven't yet heard of anyone getting it right.. Hi, thanks for that. after implementing yours and wipeout's suggestions (thank you both), x-lite changed its default codecs to G711a. which is great... and a way forward. but it still does not play sound when the 1000 is dialed. my * is behind nat. and my test pc is as well. Here are my settings: sip.conf [senad] type=friend secret=blah host=dynamic dtmfmode=inband thanks senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Littlepage Sent: 08 August 2003 19:50 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue Change the allow=all in sip.conf to allow=alaw and see if that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, August 08, 2003 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue Hi, X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but i can not hear it. Tried many times with the same result. After quite few tries * complains about: - WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 43 (Response) WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to determine sequence number from '' - Has anyone had the same problem? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
Hi, thanks for that. after implementing yours and wipeout's suggestions (thank you both), x-lite changed its default codecs to G711a. which is great... and a way forward. but it still does not play sound when the 1000 is dialed. my * is behind nat. and my test pc is as well. Here are my settings: sip.conf [senad] type=friend secret=blah host=dynamic dtmfmode=inband thanks senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nathan Littlepage Sent: 08 August 2003 19:50 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue Change the allow=all in sip.conf to allow=alaw and see if that works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, August 08, 2003 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue Hi, X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but i can not hear it. Tried many times with the same result. After quite few tries * complains about: - WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 43 (Response) WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to determine sequence number from '' - Has anyone had the same problem? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users