Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
My incoming BV has been intermittant for the last two days as well. It has gone down somewhere around 4:30 PM Eastern two days in a row, then been back up in the morning. In the 10:00 AM hour today, it was down for about ten minutes. Jason Schafer writes: I have been trying on and off for a couple of weeks to no avail... Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Janina SajkaPhone: +1.240.715.1272 Partner, Capital Accessibility LLC http://www.CapitalAccessibility.Com Marketing the Owasys 22C talking screenless cell phone in the U.S. and Canada--Go to http://www.ScreenlessPhone.Com to learn more. Chair, Accessibility Workgroup Free Standards Group (FSG) [EMAIL PROTECTED] http://a11y.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I'm not sure if it matters, but I am running Asterisk 1.0.9. I used the AAH distribution to do the build. Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
No incoming voice or incoming calls? Jason Schafer wrote: I'm not sure if it matters, but I am running Asterisk 1.0.9. I used the AAH distribution to do the build. Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I cannot send dial commands to my auto attendant or speak if I use a did to send the inbound calls to a specific extension Does asterisk says something in the verbose console? please post your sip.conf relevant entries for BroadVoice. I have just cancelled with BroadVoice (too much latency for the places i wanted to call), so i never used the incoming number. But im glad to help if i can. Best RegardsOn 9/26/05, Jason Schafer [EMAIL PROTECTED] wrote: Hi:I am running AAH and setup Broadvoice, but when I call in to the BVnumber I cannot send dial commands to my auto attendant or speak if Iuse a did to send the inbound calls to a specific extension.I'll gladly capture an SID debug and place a call, or post any necessary conffiles.TIAJason___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Does asterisk says something in the verbose console? I'm not sure what the verbose console is, but I can run sip debug and post the output when I make an inbound call. please post your sip.conf relevant entries for BroadVoice. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown pedantic=no ; added for Broadvoice support 8/3/05 EK externip=216.xxx.xxx.xxx localnet=172.xxx.xxx.0/255.255.255.0 I have just cancelled with BroadVoice (too much latency for the places i wanted to call), so i never used the incoming number. But im glad to help if i can. I have outbound setup on VOIPJet, my intent with the Broadvoice is to setup a forward on busy with my landline to roll over to the BV number. Here's the output from sip debug m=audio 14008 RTP/AVP 0 8 2 18 96 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:101 telephone-event/8000 13 headers, 12 lines Using latest request as basis request Sending to 147.135.0.128 : 5060 (non-NAT) Found no matching peer or user for '147.135.0.128:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 101 Peer audio RTP is at port 147.135.0.128:14008 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for s in from-sip-external list_route: hop: sip:[EMAIL PROTECTED]:5060;ep=147.135.0.129;transport=udp Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack -- Executing Goto(SIP/147.135.0.129-095da350, from-pstn|s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(SIP/147.135.0.129-095da350, ) in new stack We're at 216.xxx.xxx.xxx port x Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 238 v=0 o=root 1782 1782 IN IP4 216.xxx.xxx.xxx s=session c=IN IP4 216.xxx.xxx.xxx t=0 0 m=audio 14138 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 ACK Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Max-Forwards: 69 Content-Length: 0 9 headers, 0 lines -- Executing SetVar(SIP/147.135.0.129-095da350, intype=aa_2) in new stack -- Executing Cut(SIP/147.135.0.129-095da350, intype=intype|-|1) in new stack -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?7:9) in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?10:12) in new stack -- Goto (from-pstn-reghours,s,12) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?13:15) in new stack -- Goto
RE: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I am also a long time client, and have no incoming BV today. -Darren From: [EMAIL PROTECTED] on behalf of Jason Schafer Sent: Mon 9/26/2005 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice Does asterisk says something in the verbose console? I'm not sure what the verbose console is, but I can run sip debug and post the output when I make an inbound call. please post your sip.conf relevant entries for BroadVoice. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown pedantic=no ; added for Broadvoice support 8/3/05 EK externip=216.xxx.xxx.xxx localnet=172.xxx.xxx.0/255.255.255.0 I have just cancelled with BroadVoice (too much latency for the places i wanted to call), so i never used the incoming number. But im glad to help if i can. I have outbound setup on VOIPJet, my intent with the Broadvoice is to setup a forward on busy with my landline to roll over to the BV number. Here's the output from sip debug m=audio 14008 RTP/AVP 0 8 2 18 96 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:101 telephone-event/8000 13 headers, 12 lines Using latest request as basis request Sending to 147.135.0.128 : 5060 (non-NAT) Found no matching peer or user for '147.135.0.128:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 101 Peer audio RTP is at port 147.135.0.128:14008 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for s in from-sip-external list_route: hop: sip:[EMAIL PROTECTED]:5060;ep=147.135.0.129;transport=udp Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack -- Executing Goto(SIP/147.135.0.129-095da350, from-pstn|s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(SIP/147.135.0.129-095da350, ) in new stack We're at 216.xxx.xxx.xxx port x Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 238 v=0 o=root 1782 1782 IN IP4 216.xxx.xxx.xxx s=session c=IN IP4 216.xxx.xxx.xxx t=0 0 m=audio 14138 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 ACK Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Max-Forwards: 69 Content-Length: 0 9 headers, 0 lines -- Executing SetVar(SIP/147.135.0.129-095da350, intype=aa_2) in new stack -- Executing Cut(SIP/147.135.0.129-095da350, intype=intype|-|1) in new stack
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I have been trying on and off for a couple of weeks to no avail... Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren it works here today but they can be a bit unpredictable I use a cheap byod lite account mostly as a test tool. I figure if they grow up someday I might use them more. I have been wondering if they will meet the FCC deadlines or just fade away. At least some providers have been sending notices and collecting street addresses last few months. Others look like they are not really preparing to stay in the business when the deadlines hit. Maybe they are hoping another provider will buy the customer base and DID's? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
IMHO - you should not use price and quality in the same sentence for BV. On Mon, 2005-09-26 at 15:20 -0400, Jason Schafer wrote: I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
connect.voicepulse.com allows up to 4 calls at a time coming into an $11/month DID with choice of IAX or SIP there was discussion here and other places before about broadvoice allowing the calls but then charging 3.9c a minute for the extra channels used shop carefully Jason Schafer wrote: I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
This sounds like a winner, are you using voicepulse? Also, I downloaded the iso for AAH 1.5. Are there any noteworthy bugs that are fixed? I don't really like to upgrade unless I need to. Jason Paul wrote: connect.voicepulse.com allows up to 4 calls at a time coming into an $11/month DID with choice of IAX or SIP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Price is about the only good thing... quality? Jajajajaj reliable? Jajajajja -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Schafer Sent: Monday, September 26, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Using voicepulse retail account because I need the caller ID name. Maybe they have added that to connect accounts by now. Jason Schafer wrote: This sounds like a winner, are you using voicepulse? Also, I downloaded the iso for AAH 1.5. Are there any noteworthy bugs that are fixed? I don't really like to upgrade unless I need to. Jason Paul wrote: connect.voicepulse.com allows up to 4 calls at a time coming into an $11/month DID with choice of IAX or SIP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I'm having the same issue too. Just signed up, never had a problem before with a sipura box at a friends house configured as an extension off my Asterisk box. I'm behind NAT, and my firewall has the asterisk box configured as a DMZ, also I forwarded the sip and RTP ports just to be sure. Anyone solve this one yet? I sent an email off to broadvoice but they were less than helpful. Tim Jackson [EMAIL PROTECTED] [2004-10-24 00:29:02 -0500]: I'm having the same issue, and I'm not behind NAT. Maybe this is a BV issue? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry Evans Sent: Saturday, October 23, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router since someone had a problem with that, but still no dice. I've had my Asterisk server running fine for a few months, but this is the first time I've tried a VOIP service with it. I just downloaded and installed the lastest CVS and the problem is still there also. Here's some of my configuration information: sip.conf (I've tried with nat=no and it didn't help) [general] context=from-sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) maxexpirey=3600 defaultexpirey=120 callerid=No CallID tos=lowdelay; 0x18 ; reliabile before dtmfmode=inband srvlookup=yes ;progressinband=no nat=yes notifymimetype=text/plain [broadvoice] type=friend username=801527 (hid real number) fromuser=801527 (hid real number) secret= (hid real password) fromdomain=sip.broadvoice.com host=sip.broadvoice.com canreinvite=no dtmfmode=inband context=broadvoice-inbound nat=yes (tried nat=never also) disallow=all allow=ulaw insecure=very I have the following ports forwarded to my linux server (it's behind a NAT router): 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of those have both TCP and UDP forwarded for now. I've tried several different combinations from different posts, including splitting the broadvoice section up into parts for incoming and outgoing, but it still didn't work. Anyone have any ideas? Let me know if traces, etc. will help and I'll capture and post some. Thanks, Terry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I have it working finehere are my configs: On my NAT Firewall I have port 5060 UDP and 1-2 UDP open to my * box. In SIP.conf I have the following: [general]context=incomingport=5060bindaddr=192.168.234.111maxexpirey=180defaultexpirey=160tos=0x08nat=nosrvlookup=yesvideosupport=nodtmfmode=inbanddisallow=all ; Disallow all codecsallow=ulawlanguage=enexternip=no-ip.com_hostnamelocalnet=192.168.234.0/255.255.255.0 register=phone_number:password@sip.broadvoice.com/broadvoice [broadvoice]type=friendusername=phonenumberfromuser=phonenumbersecret=passwordhost=sip.broadvoice.commaxexpirey=15fromdomain=sip.broadvoice.comnat=yescanreinvite=noinsecure=veryqualify=yesdtmfmode=inbanddisallow=all ; Disallow all codecsallow=ulaw In the EXTENSIONS.conf I have an extension defined to forward the "broadvoice" extension to my Sipura SPA-2000 Line 1 extension. It works fine for me thru NAT. Hope this helps. -Jeff - I'm having the same issue too. Just signed up, never had a problem before with a sipura box at a friends house configured as an extension off my Asterisk box. I'm behind NAT, and my firewall has the asterisk box configured as a DMZ, also I forwarded the sip and RTP ports just to be sure. Anyone solve this one yet? I sent an email off to broadvoice but they were less than helpful. Tim Jackson [EMAIL PROTECTED] [2004-10-24 00:29:02 -0500]: I'm having the same issue, and I'm not behind NAT. Maybe this is a BV issue? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Terry Evans Sent: Saturday, October 23, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router since someone had a problem with that, but still no dice. I've had my Asterisk server running fine for a few months, but this is the first time I've tried a VOIP service with it. I just downloaded and installed the lastest CVS and the problem is still there also. Here's some of my configuration information: sip.conf (I've tried with nat=no and it didn't help) [general] context=from-sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) maxexpirey=3600 defaultexpirey=120 callerid=No CallID tos=lowdelay; 0x18 ; reliabile before dtmfmode=inband srvlookup=yes ;progressinband=no nat=yes notifymimetype=text/plain [broadvoice] type=friend username=801527 (hid real number) fromuser=801527 (hid real number) secret= (hid real password) fromdomain=sip.broadvoice.com host=sip.broadvoice.com canreinvite=no dtmfmode=inband context=broadvoice-inbound nat=yes (tried nat=never also) disallow=all allow=ulaw insecure=very I have the following ports forwarded to my linux server (it's behind a NAT router): 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of those have both TCP and UDP forwarded for now. I've tried several different combinations from different posts, including splitting the broadvoice section up into parts for incoming and outgoing, but it still didn't work. Anyone have any ideas? Let me know if traces, etc. will help and I'll capture and post some. Thanks, Terry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Jeff, I did a cut-n-paste of your configuration straight into my sip.conf, updated the username and password. Still getting the same result as before, audio in only one direction. Can can call between my local SIP extensions fine, so I know my sipura box is working and configured correctly. I've sent another email to broadvoice, but I'm about to give up on them. Kinda sucks, because they have some of the better rates for BYOD. [EMAIL PROTECTED] [EMAIL PROTECTED] [2004-10-26 13:30:29 +]: I have it working finehere are my configs: On my NAT Firewall I have port 5060 UDP and 1-2 UDP open to my * box. In SIP.conf I have the following: [general] context=incoming port=5060 bindaddr=192.168.234.111 maxexpirey=180 defaultexpirey=160 tos=0x08 nat=no srvlookup=yes videosupport=no dtmfmode=inband disallow=all ; Disallow all codecs allow=ulaw language=en externip=no-ip.com_hostname localnet=192.168.234.0/255.255.255.0 register=phone_number:password@sip.broadvoice.com/broadvoice [broadvoice] type=friend username=phonenumber fromuser=phonenumber secret=password host=sip.broadvoice.com maxexpirey=15 fromdomain=sip.broadvoice.com nat=yes canreinvite=no insecure=very qualify=yes dtmfmode=inband disallow=all ; Disallow all codecs allow=ulaw In the EXTENSIONS.conf I have an extension defined to forward the broadvoice extension to my Sipura SPA-2000 Line 1 extension. It works fine for me thru NAT. Hope this helps. -Jeff - I'm having the same issue too. Just signed up, never had a problem before with a sipura box at a friends house configured as an extension off my Asterisk box. I'm behind NAT, and my firewall has the asterisk box configured as a DMZ, also I forwarded the sip and RTP ports just to be sure. Anyone solve this one yet? I sent an email off to broadvoice but they were less than helpful. Tim Jackson [EMAIL PROTECTED] [2004-10-24 00:29:02 -0500]: I'm having the same issue, and I'm not behind NAT. Maybe this is a BV issue? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry Evans Sent: Saturday, October 23, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router since someone had a problem with that, but still no dice. I've had my Asterisk server running fine for a few months, but this is the first time I've tried a VOIP service with it. I just downloaded and installed the lastest CVS and the problem is still there also. Here's some of my configuration information: sip.conf (I've tried with nat=no and it didn't help) [general] context=from-sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) maxexpirey=3600 defaultexpirey=120 callerid=No CallID tos=lowdelay; 0x18 ; reliabile before dtmfmode=inband srvlookup=yes ;progressinband=no nat=yes notifymimetype=text/plain [broadvoice] type=friend username=801527 (hid real number) fromuser=801527 (hid real number) secret= (hid real password) fromdomain=sip.broadvoice.com host=sip.broadvoice.com canreinvite=no dtmfmode=inband context=broadvoice-inbound nat=yes (tried nat=never also) disallow=all allow=ulaw insecure=very I have the following ports forwarded to my linux server (it's behind a NAT router): 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of those have both TCP and UDP forwarded for now. I've tried several different combinations from different posts, including splitting the broadvoice section up into parts for incoming and outgoing, but it still didn't work. Anyone have any ideas? Let me know if traces, etc. will help and I'll capture and post some. Thanks, Terry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
On Sat, 23 Oct 2004, Terry Evans wrote: I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. (snip) I have the following ports forwarded to my linux server (it's behind a NAT router): 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of those have both TCP and UDP forwarded for now. It really sounds like a NAT problem to me.. If your NAT supports the notion of a DMZ host then give that a try. Or if the NAT has some sort of logging feature to let you know when the nat receives unexpected packets and discards them, then look through the log. It may be that BV isn't sending RTP in the 2-21000 port range, and that these packets are being dropped by the NAT. Outgoing RTP (voice) would work fine, of course, because the NAT is designed to work that direction. FYI, I just placed a call to my BV number and ran 'netstat -nupa'. UDP connections showed up on ports 14704, 14705, 19838, 19839. These disappeared when I hung up the call. While it might be a config issue, I'm inclined to believe that NAT is making life unpleasant for you. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
It also sounds like some type of NAT issue to me, but I can't figure out what's going wrong. I changed the RTP ports back to 1-2 and set the router up to forward those, but still no incoming voice. Kevin suggest I try the two inbound sections in the sip.conf, but I had already tried them prior to my previous post. I've tried lots of combinations of sip.conf files I could find in this mailing list, but none of them seem to work for me for some reason. Terry On Sat, 23 Oct 2004 17:39:12 -0600 (MDT), Greg Hill [EMAIL PROTECTED] wrote: It really sounds like a NAT problem to me.. If your NAT supports the notion of a DMZ host then give that a try. Or if the NAT has some sort of logging feature to let you know when the nat receives unexpected packets and discards them, then look through the log. It may be that BV isn't sending RTP in the 2-21000 port range, and that these packets are being dropped by the NAT. Outgoing RTP (voice) would work fine, of course, because the NAT is designed to work that direction. FYI, I just placed a call to my BV number and ran 'netstat -nupa'. UDP connections showed up on ports 14704, 14705, 19838, 19839. These disappeared when I hung up the call. While it might be a config issue, I'm inclined to believe that NAT is making life unpleasant for you. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I'm having the same issue, and I'm not behind NAT. Maybe this is a BV issue? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry Evans Sent: Saturday, October 23, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router since someone had a problem with that, but still no dice. I've had my Asterisk server running fine for a few months, but this is the first time I've tried a VOIP service with it. I just downloaded and installed the lastest CVS and the problem is still there also. Here's some of my configuration information: sip.conf (I've tried with nat=no and it didn't help) [general] context=from-sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) maxexpirey=3600 defaultexpirey=120 callerid=No CallID tos=lowdelay; 0x18 ; reliabile before dtmfmode=inband srvlookup=yes ;progressinband=no nat=yes notifymimetype=text/plain [broadvoice] type=friend username=801527 (hid real number) fromuser=801527 (hid real number) secret= (hid real password) fromdomain=sip.broadvoice.com host=sip.broadvoice.com canreinvite=no dtmfmode=inband context=broadvoice-inbound nat=yes (tried nat=never also) disallow=all allow=ulaw insecure=very I have the following ports forwarded to my linux server (it's behind a NAT router): 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of those have both TCP and UDP forwarded for now. I've tried several different combinations from different posts, including splitting the broadvoice section up into parts for incoming and outgoing, but it still didn't work. Anyone have any ideas? Let me know if traces, etc. will help and I'll capture and post some. Thanks, Terry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
