Re: [asterisk-users] can't call through voip provider
It is not that easy to give the answer. There are lots of itsp typical ways of registration and you haven't provide the info needed to help you out. You need a register line in the general part of sip.conf. It should look something like (mine looks like this register = DID:SECRET:username@ipness.net:6060 And you need a sip entry in sip.conf. For me it looks something like [DID] type=friend host=ipness.net fromuser=DID fromdomain=ipness.net username=username secret=secret insecure=very context=inbound port=6060 qualify=2000 canreinvite=no disallow=all ;allow=ulaw allow=alaw But your provider might need other settings. So ask your provider. If you are on public IP and not behind NAT you should use nat=no From the sip message I make up that the You didn't provide debug info but copied and paste a sip message. If you would like people to help you, you have to provide proper info. CLI output, sip.conf (without passwords and IP adress info) and the sip messages will be helpful. Are you aware of the fact that you need to open UDP ports and not TCP. Your provider should be able to tell you how to configure such an account on an asterisk box, or at least help you to figure it out. A serious ITSP must have customers using Asterisk. If you have no idea what you are doing my advice is to start reading Asterisk: The future of telephony, freely available on http://www.asteriskdocs.org/ . VERY SERIOUS WARNING: Don't put the credentials of a sip account in a mail to a mailing list. People might use your account to call satelite lines for EUR 7,50 per minute. This kind of mistakes might bankcrupt you :-( I hope this helps. Erik On 19 nov 2009, at 22:36, Landy Landy wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 7:51 AM Ok. I do NOT have ports 1-2 opened in. I guess I I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- 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 ___ -- 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] can't call through voip provider
Erik. I already solved this problem and posted it. I was reloading all the setting but, it wasn't changing the provider's ip info. After doing a restart now everything worked. Thanks any ways for your help. --- On Fri, 11/27/09, meetmecall i...@meetmecall.nl wrote: From: meetmecall i...@meetmecall.nl Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, November 27, 2009, 9:51 AM It is not that easy to give the answer. There are lots of itsp typical ways of registration and you haven't provide the info needed to help you out. You need a register line in the general part of sip.conf. It should look something like (mine looks like this register = DID:SECRET:username@ipness.net:6060 And you need a sip entry in sip.conf. For me it looks something like [DID] type=friend host=ipness.net fromuser=DID fromdomain=ipness.net username=username secret=secret insecure=very context=inbound port=6060 qualify=2000 canreinvite=no disallow=all ;allow=ulaw allow=alaw But your provider might need other settings. So ask your provider. If you are on public IP and not behind NAT you should use nat=no From the sip message I make up that the You didn't provide debug info but copied and paste a sip message. If you would like people to help you, you have to provide proper info. CLI output, sip.conf (without passwords and IP adress info) and the sip messages will be helpful. Are you aware of the fact that you need to open UDP ports and not TCP. Your provider should be able to tell you how to configure such an account on an asterisk box, or at least help you to figure it out. A serious ITSP must have customers using Asterisk. If you have no idea what you are doing my advice is to start reading Asterisk: The future of telephony, freely available on http://www.asteriskdocs.org/ . VERY SERIOUS WARNING: Don't put the credentials of a sip account in a mail to a mailing list. People might use your account to call satelite lines for EUR 7,50 per minute. This kind of mistakes might bankcrupt you :-( I hope this helps. Erik On 19 nov 2009, at 22:36, Landy Landy wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 7:51 AM Ok. I do NOT have ports 1-2 opened in. I guess I I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- 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
Re: [asterisk-users] can't call through voip provider
Hello. I have my server running for about 30 days. Every time I did some changes to my sip.conf file I did reload in the cli. I thought this would change the new values. Somehow it wasn't. I decided to do a restart now and that used my new settings. The same settings I've been posting here the past week and weren't working. After restarting asterisk I'm able to use my provider via asterisk to make calls. I would like to thank those who helped me. --- On Fri, 11/20/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, November 20, 2009, 8:53 AM Sorry to bother you again with my problem but, is that I can't figure out what's going on with my setup. I have no idea of why my asterisk server is not communicating with my provider's. I've searched, googled, and can't find my solution. I've followed many tutorials but can't get anywhere. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:53 PM Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan 0 0 ACCEPT udp -- eth0 eth10.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 0 0 ACCEPT udp -- eth0 eth10.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 162 ACCEPT udp -- eth1 eth00.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 36 2372 ACCEPT udp -- eth1 eth00.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 0 0 ACCEPT tcp -- eth1 eth00.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 3 144 ACCEPT tcp -- eth1 eth00.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 and now the debug: etransmitting #5 (NAT) to 190.80.152.200:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport Max-Forwards: 70 From: 102 sip:77...@190.80.152.200;tag=as5084570c To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.152.200 Call-ID: 22569d3b767276276c6c65c84b314...@190.80.152.200 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 22:53:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 135722140 135722140 IN IP4 190.80.152.200 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.200 t=0 0 m=audio 10074 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I'm already frustrated with this. --- On Thu, 11/19/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:11 PM On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.com wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. Who is your voipprovider? Did they give you the settings you're using in your sip.conf? Also, you've got some typos in your sip config (insucure = insecure, careinvite = canreinvite). You could try something like this: [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=yes canreinvite=yes insecure=very disallow=all allow=ulaw allow=alaw -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows
Re: [asterisk-users] can't call through voip provider
Sorry to bother you again with my problem but, is that I can't figure out what's going on with my setup. I have no idea of why my asterisk server is not communicating with my provider's. I've searched, googled, and can't find my solution. I've followed many tutorials but can't get anywhere. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:53 PM Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan 0 0 ACCEPT udp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 0 0 ACCEPT udp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 1 62 ACCEPT udp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 36 2372 ACCEPT udp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 0 0 ACCEPT tcp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 0 0 ACCEPT tcp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 0 0 ACCEPT tcp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 3 144 ACCEPT tcp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 and now the debug: etransmitting #5 (NAT) to 190.80.152.200:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport Max-Forwards: 70 From: 102 sip:77...@190.80.152.200;tag=as5084570c To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.152.200 Call-ID: 22569d3b767276276c6c65c84b314...@190.80.152.200 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 22:53:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 135722140 135722140 IN IP4 190.80.152.200 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.200 t=0 0 m=audio 10074 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I'm already frustrated with this. --- On Thu, 11/19/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:11 PM On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.com wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. Who is your voipprovider? Did they give you the settings you're using in your sip.conf? Also, you've got some typos in your sip config (insucure = insecure, careinvite = canreinvite). You could try something like this: [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=yes canreinvite=yes insecure=very disallow=all allow=ulaw allow=alaw -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows- ___ -- 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 ___ -- 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] can't call through voip provider
Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- 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] can't call through voip provider
Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 7:51 AM Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- 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] can't call through voip provider
On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.comwrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. Who is your voipprovider? Did they give you the settings you're using in your sip.conf? Also, you've got some typos in your sip config (insucure = insecure, careinvite = canreinvite). You could try something like this: [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=yes canreinvite=yes insecure=very disallow=all allow=ulaw allow=alaw -- Thanks, --Warren Selby http://www.selbytech.com ___ -- 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] can't call through voip provider
Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan 0 0 ACCEPT udp -- eth0 eth10.0.0.0/00.0.0.0/0 udp dpts:5060:5070 0 0 ACCEPT udp -- eth0 eth10.0.0.0/00.0.0.0/0 udp dpts:1:10100 162 ACCEPT udp -- eth1 eth00.0.0.0/00.0.0.0/0 udp dpts:5060:5070 36 2372 ACCEPT udp -- eth1 eth00.0.0.0/00.0.0.0/0 udp dpts:1:10100 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/00.0.0.0/0 tcp dpts:5060:5070 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/00.0.0.0/0 tcp dpts:1:10100 0 0 ACCEPT tcp -- eth1 eth00.0.0.0/00.0.0.0/0 tcp dpts:5060:5070 3 144 ACCEPT tcp -- eth1 eth00.0.0.0/00.0.0.0/0 tcp dpts:1:10100 and now the debug: etransmitting #5 (NAT) to 190.80.152.200:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport Max-Forwards: 70 From: 102 sip:77...@190.80.152.200;tag=as5084570c To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.152.200 Call-ID: 22569d3b767276276c6c65c84b314...@190.80.152.200 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 22:53:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 135722140 135722140 IN IP4 190.80.152.200 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.200 t=0 0 m=audio 10074 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I'm already frustrated with this. --- On Thu, 11/19/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:11 PM On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.com wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. Who is your voipprovider? Did they give you the settings you're using in your sip.conf? Also, you've got some typos in your sip config (insucure = insecure, careinvite = canreinvite). You could try something like this: [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=yes canreinvite=yes insecure=very disallow=all allow=ulaw allow=alaw -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows- ___ -- 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] can't call through voip provider
On Thu, Nov 19, 2009 at 4:53 PM, Landy Landy landysacco...@yahoo.comwrote: Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan Did you try the config I provided in my previous email? You should copy / paste it to avoid any typographical errors (which I mentioned what you posted contained). -- Thanks, --Warren Selby http://www.selbytech.com ___ -- 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] can't call through voip provider
What does your Dial command look like? It should be something like
this:
exten = _9.,1,Dial(SIP/voipprovider/${EXTEN:1})
Also, do you have a register statement for voipprovider in sip.conf?
Does sip show registry show that it's registered successfully?
At 2:53 PM on 19 Nov 2009, Landy Landy wrote:
Nothing. I don't know what in the world is going on with my setup.
[...]
I'm already frustrated with this.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't call through voip provider
I have the conf provided in last post.
exten = _9.,1,Dial(SIP/voipprovider/${EXTEN:1})
Yes, I have that in the dialplan.
Does sip show registry show that it's registered
successfully?
*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
0 SIP registrations.
___
-- 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] can't call through voip provider
Hello. Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. --- On Tue, 11/17/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, November 17, 2009, 7:33 AM Thanks for replying. Here is the output of sip set debug peer voipprovider: -- Called 1829257x...@voipprovider Retransmitting #1 (NAT) to myextip:5060: INVITE sip:18292574...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77632...@190.80.152.7 Call-ID: 2908dd00500059761cc66bd81553e...@190.80.152.7 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 190.80.152.7 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 myextip t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Scheduling destruction of SIP dialog '2908dd00500059761cc66bd81553e...@myextip' in 32000 ms (Method: INVITE) By looking at this trace I dont see my provider's ip address anywhere. I guess I'm doing something wrong in my conf. --- On Mon, 11/16/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, November 16, 2009, 9:51 PM On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy landysacco...@yahoo.com wrote: snip I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763x...@default:1] Dial(SIP/102-b6a06a40, SIP/1xxx763x...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx763x...@voipprovider snip We could really use a little more of the CLI output of a failed call. Maybe increase your verbosity to at least 10. Also, what does the SIP debug of a call to the VOIP provider look like (from the cli, type sip set debug peer voipprovider)? -- Thanks, --Warren Selby http://www.selbytech.com
Re: [asterisk-users] can't call through voip provider
On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] can't call through voip provider
Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] can't call through voip provider
What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 ___ -- 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] can't call through voip provider
According to the provider he says he doesn't see anything coming in on their side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new connections. I thought when asterisk starts a communication with a remote server using an unprivate port to port 5060 theres already an ESTABLISHED communication. I don't know if I'm having problems with my firewall script or what but, since there isn't any new connections coming form outside I think I'm ok to accept only ESTABLISHED,RELATED coming in. I don't know but, I'm stuck with this problem and don't know what else to do. --- On Wed, 11/18/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:03 PM What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 ___ -- 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] can't call through voip provider
According to what I know, you have to have 5060 open out and 1-2 open in (you can cut this to as small as 1-10004). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Wednesday, November 18, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't call through voip provider According to the provider he says he doesn't see anything coming in on their side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new connections. I thought when asterisk starts a communication with a remote server using an unprivate port to port 5060 theres already an ESTABLISHED communication. I don't know if I'm having problems with my firewall script or what but, since there isn't any new connections coming form outside I think I'm ok to accept only ESTABLISHED,RELATED coming in. I don't know but, I'm stuck with this problem and don't know what else to do. --- On Wed, 11/18/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:03 PM What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 ___ -- 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 ___ -- 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] can't call through voip provider
Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. Thanks. --- On Wed, 11/18/09, Danny Nicholas da...@debsinc.com wrote: From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] can't call through voip provider To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:18 PM According to what I know, you have to have 5060 open out and 1-2 open in (you can cut this to as small as 1-10004). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Wednesday, November 18, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't call through voip provider According to the provider he says he doesn't see anything coming in on their side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new connections. I thought when asterisk starts a communication with a remote server using an unprivate port to port 5060 theres already an ESTABLISHED communication. I don't know if I'm having problems with my firewall script or what but, since there isn't any new connections coming form outside I think I'm ok to accept only ESTABLISHED,RELATED coming in. I don't know but, I'm stuck with this problem and don't know what else to do. --- On Wed, 11/18/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:03 PM What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 ___ -- 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 ___ -- 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] can't call through voip provider
To be perfectly complete, exactly which inbound ports to open will depend on the phones in use. For example, a Cisco 7940 (using this example because I have one on my desk at the moment), the default ports from the config are: voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 Meaning, you have to have 5060 open (obviously), and all the ports between the start and end media port. Many phones will let you adjust where these boundaries lie, but some won't. You'll need enough range to cover every kind of phone (soft or hard) that you are using. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Thursday, 19 November 2009 09:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't call through voip provider Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. Thanks. --- On Wed, 11/18/09, Danny Nicholas da...@debsinc.com wrote: From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] can't call through voip provider To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:18 PM According to what I know, you have to have 5060 open out and 1-2 open in (you can cut this to as small as 1-10004). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Wednesday, November 18, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't call through voip provider According to the provider he says he doesn't see anything coming in on their side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new connections. I thought when asterisk starts a communication with a remote server using an unprivate port to port 5060 theres already an ESTABLISHED communication. I don't know if I'm having problems with my firewall script or what but, since there isn't any new connections coming form outside I think I'm ok to accept only ESTABLISHED,RELATED coming in. I don't know but, I'm stuck with this problem and don't know what else to do. --- On Wed, 11/18/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:03 PM What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] can't call through voip provider
Thanks for replying. Here is the output of sip set debug peer voipprovider: -- Called 1829257x...@voipprovider Retransmitting #1 (NAT) to myextip:5060: INVITE sip:18292574...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77632...@190.80.152.7 Call-ID: 2908dd00500059761cc66bd81553e...@190.80.152.7 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 190.80.152.7 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 myextip t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Scheduling destruction of SIP dialog '2908dd00500059761cc66bd81553e...@myextip' in 32000 ms (Method: INVITE) By looking at this trace I dont see my provider's ip address anywhere. I guess I'm doing something wrong in my conf. --- On Mon, 11/16/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, November 16, 2009, 9:51 PM On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy landysacco...@yahoo.com wrote: snip I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763x...@default:1] Dial(SIP/102-b6a06a40, SIP/1xxx763x...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx763x...@voipprovider snip We could really use a little more of the CLI output of a failed call. Maybe increase your verbosity to at least 10. Also, what does the SIP debug of a call to the VOIP provider look like (from the cli, type sip set debug peer voipprovider)? -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows- ___ -- 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] can't call through voip provider
On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy landysacco...@yahoo.comwrote: snip I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763x...@default:1] Dial(SIP/102-b6a06a40, SIP/1xxx763x...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx763x...@voipprovider snip We could really use a little more of the CLI output of a failed call. Maybe increase your verbosity to at least 10. Also, what does the SIP debug of a call to the VOIP provider look like (from the cli, type sip set debug peer voipprovider)? -- Thanks, --Warren Selby http://www.selbytech.com ___ -- 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
