You could use MTR command. Its a trace route improved. Marlon Araujo
> On Jan 20, 2015, at 08:59, [email protected] wrote: > > Send asterisk-users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. sip show channelstats reliable? (Todd R.) > 2. Re: sip show channelstats reliable? (Todd R.) > 3. Re: sip show channelstats reliable? (Eric Wieling) > 4. Re: sip show channelstats reliable? (Todd R.) > 5. Re: sip show channelstats reliable? (Scott Griepentrog) > 6. Re: SEMI-OFFTOPIC openvox (ricky gutierrez) > 7. Re: SEMI-OFFTOPIC openvox (A J Stiles) > 8. Re: MWI issue (Haley,Scott A) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 19 Jan 2015 12:17:25 -0600 > From: Todd R. <[email protected]> > To: Asterisk-Users List <[email protected]> > Subject: [asterisk-users] sip show channelstats reliable? > Message-ID: <[email protected]> > Content-Type: text/plain; charset="iso-8859-1" > > I am seeing lots of lost packets when running the command sip show > channelstats at the CLI. > There are issues across multiple Asterisk servers I am trying to diagnose but > everything I read seems to point to this command being pretty unreliable. > Can I trust the info this command shows? > I am showing lots of lost packets in sip show channelstats but I can't see > any packet loss when pinging the same IP's to/from. > Since I don't 100% control the network my gear is on, I need something > outside of Asterisk to show the network engineer to convince here and myself > that there are network issues. > All I have is the loss that's shown from this command with no real network > stats to back it up. > Is there a magic command in CentOS anyone can recommend to diagnose and match > up the issues shown in Asterisk using this command? > Moving gear around on the network changes the info Asterisk shows a LOT. For > example, if I point traffic to the main physical gateway I get loss to a > particular customer's IP (their PBX), if I move it to another place on the > network (as a VM) their IP is good and other customers IP's start showing > loss using the channelstats info. > Driving me freakin' crazy. It does appear there are network issues causing my > troubles but I can't get help if I can't point to some hard and fast issues > outside of Asterisk. > The only thing I have right now is collissions showing on one of a few of our > pfSense devices but they are virtual running on XenServer, still this would > indicate a problem in my opinion. > Thanks in advance for any assistance on this issue. Stepping back from the > ledge now LOL > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150119/eecb2218/attachment-0001.html> > > ------------------------------ > > Message: 2 > Date: Mon, 19 Jan 2015 12:44:33 -0600 > From: Todd R. <[email protected]> > To: Asterisk-Users List <[email protected]> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: <[email protected]> > Content-Type: text/plain; charset="iso-8859-1" > > Additional info: > At the moment I am running 1.8.x but the other day I was getting the same > results on 11.x > Here is a sample from show channelstats. I do think this command is showing > that there is trouble between specific IP's and my Asterisk box but I don't > know if the numbers are accurate and reliable. > > > > > > > > > > > > > > > Peer > Call ID > Duration > Recv: Pack > Lost > ( %) > Jitter > Send: Pack > Lost > ( > %) > Jitter > > > x.x.x.x > 5531341d06b > 00:07:42 > 0000023123 > 0000063836 > (73.41%) > 0.0000 > 0000023102 > 0000000000 > ( > 0.00%) > 0.0007 > > Peer IP changed to protect the innocent :-) > > From: [email protected] > To: [email protected] > Date: Mon, 19 Jan 2015 12:17:25 -0600 > Subject: [asterisk-users] sip show channelstats reliable? > > > > > I am seeing lots of lost packets when running the command sip show > channelstats at the CLI. > There are issues across multiple Asterisk servers I am trying to diagnose but > everything I read seems to point to this command being pretty unreliable. > Can I trust the info this command shows? > I am showing lots of lost packets in sip show channelstats but I can't see > any packet loss when pinging the same IP's to/from. > Since I don't 100% control the network my gear is on, I need something > outside of Asterisk to show the network engineer to convince here and myself > that there are network issues. > All I have is the loss that's shown from this command with no real network > stats to back it up. > Is there a magic command in CentOS anyone can recommend to diagnose and match > up the issues shown in Asterisk using this command? > Moving gear around on the network changes the info Asterisk shows a LOT. For > example, if I point traffic to the main physical gateway I get loss to a > particular customer's IP (their PBX), if I move it to another place on the > network (as a VM) their IP is good and other customers IP's start showing > loss using the channelstats info. > Driving me freakin' crazy. It does appear there are network issues causing my > troubles but I can't get help if I can't point to some hard and fast issues > outside of Asterisk. > The only thing I have right now is collissions showing on one of a few of our > pfSense devices but they are virtual running on XenServer, still this would > indicate a problem in my opinion. > Thanks in advance for any assistance on this issue. Stepping back from the > ledge now LOL > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150119/632335cd/attachment-0001.html> > > ------------------------------ > > Message: 3 > Date: Mon, 19 Jan 2015 13:55:33 -0500 > From: Eric Wieling <[email protected]> > To: "[email protected]" <[email protected]>, Asterisk Users Mailing List > - Non-Commercial Discussion <[email protected]> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: > <616B4ECE1290D441AD56124FEBB03D082F43F2E5E7@mailserver2007.nyigc.globe> > > Content-Type: text/plain; charset="us-ascii" > > I've seen something similar with Adtran SIP gateways. When a re-invite > happens the Adtran gets all confused about call stats and marks the > pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC > reinvites happen when a codec changes or the channel switches to T.38. > > Also Adtran SIP gateways appear not to support OPTIONS packets when running > in SIP proxy mode, which is very annoying. At some point I'll try and > arrange a slugfest between Digium and Adtran and they can figure out why it > doesn't work. > > From: [email protected] > [mailto:[email protected]] On Behalf Of Todd R. > Sent: Monday, January 19, 2015 1:45 PM > To: Asterisk-Users List > Subject: Re: [asterisk-users] sip show channelstats reliable? > > Additional info: > > At the moment I am running 1.8.x but the other day I was getting the same > results on 11.x > > Here is a sample from show channelstats. I do think this command is showing > that there is trouble between specific IP's and my Asterisk box but I don't > know if the numbers are accurate and reliable. > > Peer > > Call ID > > Duration > > Recv: Pack > > Lost > > ( %) > > Jitter > > Send: Pack > > Lost > > ( > > %) > > Jitter > > x.x.x.x > > 5531341d06b > > 00:07:42 > > 0000023123 > > 0000063836 > > (73.41%) > > 0.0000 > > 0000023102 > > 0000000000 > > ( > > 0.00%) > > 0.0007 > > > Peer IP changed to protect the innocent :-) > > ________________________________ > From: [email protected]<mailto:[email protected]> > To: [email protected]<mailto:[email protected]> > Date: Mon, 19 Jan 2015 12:17:25 -0600 > Subject: [asterisk-users] sip show channelstats reliable? > I am seeing lots of lost packets when running the command sip show > channelstats at the CLI. > > There are issues across multiple Asterisk servers I am trying to diagnose but > everything I read seems to point to this command being pretty unreliable. > > Can I trust the info this command shows? > > I am showing lots of lost packets in sip show channelstats but I can't see > any packet loss when pinging the same IP's to/from. > > Since I don't 100% control the network my gear is on, I need something > outside of Asterisk to show the network engineer to convince here and myself > that there are network issues. > > All I have is the loss that's shown from this command with no real network > stats to back it up. > > Is there a magic command in CentOS anyone can recommend to diagnose and match > up the issues shown in Asterisk using this command? > > Moving gear around on the network changes the info Asterisk shows a LOT. For > example, if I point traffic to the main physical gateway I get loss to a > particular customer's IP (their PBX), if I move it to another place on the > network (as a VM) their IP is good and other customers IP's start showing > loss using the channelstats info. > > Driving me freakin' crazy. It does appear there are network issues causing my > troubles but I can't get help if I can't point to some hard and fast issues > outside of Asterisk. > > The only thing I have right now is collissions showing on one of a few of our > pfSense devices but they are virtual running on XenServer, still this would > indicate a problem in my opinion. > > Thanks in advance for any assistance on this issue. Stepping back from the > ledge now LOL > > > > -- _____________________________________________________________________ -- > Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or > update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150119/ff2ae758/attachment-0001.html> > > ------------------------------ > > Message: 4 > Date: Mon, 19 Jan 2015 13:00:37 -0600 > From: Todd R. <[email protected]> > To: Eric Wieling <[email protected]>, Asterisk-Users List > <[email protected]> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: <[email protected]> > Content-Type: text/plain; charset="windows-1252" > > Thanks but no Adtran here. > I do think these stats are indicating an issue, I just don't know how to > prove it outside Asterisk. > > From: [email protected] > To: [email protected]; [email protected] > Date: Mon, 19 Jan 2015 13:55:33 -0500 > Subject: RE: [asterisk-users] sip show channelstats reliable? > > I?ve seen something similar with Adtran SIP gateways. When a re-invite > happens the Adtran gets all confused about call stats and marks the > pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC > reinvites happen when a codec changes or the channel switches to T.38. Also > Adtran SIP gateways appear not to support OPTIONS packets when running in SIP > proxy mode, which is very annoying. At some point I?ll try and arrange a > slugfest between Digium and Adtran and they can figure out why it doesn?t > work. From: [email protected] > [mailto:[email protected]] On Behalf Of Todd R. > Sent: Monday, January 19, 2015 1:45 PM > To: Asterisk-Users List > Subject: Re: [asterisk-users] sip show channelstats reliable? Additional > info: At the moment I am running 1.8.x but the other day I was getting the > same results on 11.x Here is a sample from show channelstats. I do think this > command is showing that there is trouble between specific IP's and my > Asterisk box but I don't know if the numbers are accurate and reliable. > PeerCall IDDurationRecv: PackLost( %)JitterSend: > PackLost(%)Jitterx.x.x.x5531341d06b00:07:4200000231230000063836(73.41%)0.000000000231020000000000(0.00%)0.0007 > Peer IP changed to protect the innocent :-) From: [email protected] > To: [email protected] > Date: Mon, 19 Jan 2015 12:17:25 -0600 > Subject: [asterisk-users] sip show channelstats reliable?I am seeing lots of > lost packets when running the command sip show channelstats at the CLI. There > are issues across multiple Asterisk servers I am trying to diagnose but > everything I read seems to point to this command being pretty unreliable. Can > I trust the info this command shows? I am showing lots of lost packets in sip > show channelstats but I can't see any packet loss when pinging the same IP's > to/from. Since I don't 100% control the network my gear is on, I need > something outside of Asterisk to show the network engineer to convince here > and myself that there are network issues. All I have is the loss that's shown > from this command with no real network stats to back it up. Is there a magic > command in CentOS anyone can recommend to diagnose and match up the issues > shown in Asterisk using this command? Moving gear around on the network > changes the info Asterisk shows a LOT. For example, if I point traffic to the > main > physical gateway I get loss to a particular customer's IP (their PBX), if I > move it to another place on the network (as a VM) their IP is good and other > customers IP's start showing loss using the channelstats info. Driving me > freakin' crazy. It does appear there are network issues causing my troubles > but I can't get help if I can't point to some hard and fast issues outside of > Asterisk. The only thing I have right now is collissions showing on one of a > few of our pfSense devices but they are virtual running on XenServer, still > this would indicate a problem in my opinion. Thanks in advance for any > assistance on this issue. Stepping back from the ledge now LOL > -- _____________________________________________________________________ -- > Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or > update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150119/f83d1bb6/attachment-0001.html> > > ------------------------------ > > Message: 5 > Date: Mon, 19 Jan 2015 13:13:01 -0600 > From: Scott Griepentrog <[email protected]> > To: [email protected], Asterisk Users Mailing List - Non-Commercial > Discussion <[email protected]> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: > <cacrpesbtxjxauplndbbmtwumyh4ksv_zrl0asrm-qnjhrmo...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > I would recommend capturing traffic outside your Asterisk server with > Wireshark, then running the Telephony/Rtp/Analysize Streams option to > determine if you have packet loss at that point in the network. > >> On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <[email protected]> wrote: >> >> Thanks but no Adtran here. >> >> I do think these stats are indicating an issue, I just don't know how to >> prove it outside Asterisk. >> >> >> ------------------------------ >> From: [email protected] >> To: [email protected]; [email protected] >> Date: Mon, 19 Jan 2015 13:55:33 -0500 >> Subject: RE: [asterisk-users] sip show channelstats reliable? >> >> >> I?ve seen something similar with Adtran SIP gateways. When a re-invite >> happens the Adtran gets all confused about call stats and marks the >> pre-reinvite leg of the call as losing large numbers of packets. BTW, >> IIRC reinvites happen when a codec changes or the channel switches to T.38. >> >> >> >> Also Adtran SIP gateways appear not to support OPTIONS packets when >> running in SIP proxy mode, which is very annoying. At some point I?ll >> try and arrange a slugfest between Digium and Adtran and they can figure >> out why it doesn?t work. >> >> >> >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *Todd R. >> *Sent:* Monday, January 19, 2015 1:45 PM >> *To:* Asterisk-Users List >> *Subject:* Re: [asterisk-users] sip show channelstats reliable? >> >> >> >> Additional info: >> >> >> >> At the moment I am running 1.8.x but the other day I was getting the same >> results on 11.x >> >> >> >> Here is a sample from show channelstats. I do think this command is >> showing that there is trouble between specific IP's and my Asterisk box but >> I don't know if the numbers are accurate and reliable. >> >> >> >> Peer >> >> Call ID >> >> Duration >> >> Recv: Pack >> >> Lost >> >> ( %) >> >> Jitter >> >> Send: Pack >> >> Lost >> >> ( >> >> %) >> >> Jitter >> >> x.x.x.x >> >> 5531341d06b >> >> 00:07:42 >> >> 0000023123 >> >> 0000063836 >> >> (73.41%) >> >> 0.0000 >> >> 0000023102 >> >> 0000000000 >> >> ( >> >> 0.00%) >> >> 0.0007 >> >> >> >> Peer IP changed to protect the innocent :-) >> >> >> ------------------------------ >> >> From: [email protected] >> To: [email protected] >> Date: Mon, 19 Jan 2015 12:17:25 -0600 >> Subject: [asterisk-users] sip show channelstats reliable? >> >> I am seeing lots of lost packets when running the command sip show >> channelstats at the CLI. >> >> >> >> There are issues across multiple Asterisk servers I am trying to diagnose >> but everything I read seems to point to this command being pretty >> unreliable. >> >> >> >> Can I trust the info this command shows? >> >> >> >> I am showing lots of lost packets in sip show channelstats but I can't see >> any packet loss when pinging the same IP's to/from. >> >> >> >> Since I don't 100% control the network my gear is on, I need something >> outside of Asterisk to show the network engineer to convince here and >> myself that there are network issues. >> >> >> >> All I have is the loss that's shown from this command with no real network >> stats to back it up. >> >> >> >> Is there a magic command in CentOS anyone can recommend to diagnose and >> match up the issues shown in Asterisk using this command? >> >> >> >> Moving gear around on the network changes the info Asterisk shows a LOT. >> For example, if I point traffic to the main physical gateway I get loss to >> a particular customer's IP (their PBX), if I move it to another place on >> the network (as a VM) their IP is good and other customers IP's start >> showing loss using the channelstats info. >> >> >> >> Driving me freakin' crazy. It does appear there are network issues causing >> my troubles but I can't get help if I can't point to some hard and fast >> issues outside of Asterisk. >> >> >> >> The only thing I have right now is collissions showing on one of a few of >> our pfSense devices but they are virtual running on XenServer, still this >> would indicate a problem in my opinion. >> >> >> >> Thanks in advance for any assistance on this issue. Stepping back from the >> ledge now LOL >> >> >> >> >> >> >> -- _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New >> to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello 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 -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > [image: Digium logo] > Scott Griepentrog > Digium, Inc ? Software Developer > 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US > direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 > Check us out at: http://digium.com ? http://asterisk.org > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150119/dbbd5968/attachment-0001.html> > > ------------------------------ > > Message: 6 > Date: Mon, 19 Jan 2015 14:37:34 -0600 > From: ricky gutierrez <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] SEMI-OFFTOPIC openvox > Message-ID: > <CAL_GE3Q=bF6sngOsS=5dUEK5oe5pH3p7=R=nyN=bunqeac5...@mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Hi, when I make an outgoing call sends me a busy here, and no one is making > call > > Contact: <sip:[email protected]:5060> > Content-Length: 0 > > > <------------> > -- Executing [984783842@to_pstn:1] Dial("SIP/101-0000004e", > "SIP/5001/84783842@,40,rRT") in new stack > == Using SIP VIDEO TOS bits 136 > == Using SIP VIDEO CoS mark 6 > == Using SIP RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > Audio is at 13780 > Video is at 50.X.X.X:18488 > Adding codec 100003 (ulaw) to SDP > Adding codec 100004 (alaw) to SDP > Adding video codec 200004 (h264) to SDP > Adding video codec 200003 (h263p) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 190.53.38.203:5060: > INVITE sip:84783842%[email protected] SIP/2.0 > Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport > Max-Forwards: 70 > From: "Operadora" <sip:[email protected]>;tag=as3708c762 > To: <sip:84783842%[email protected]> > Contact: <sip:[email protected]:5060> > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > User-Agent: inmaconsa-Voice-Sip-ipbx > Date: Mon, 19 Jan 2015 20:17:52 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE > Supported: replaces, timer > Remote-Party-ID: "Operadora" > <sip:[email protected]>;party=calling;privacy=off;screen=no > Content-Type: application/sdp > Content-Length: 507 > > v=0 > o=root 541548714 541548714 IN IP4 50.X.X.X > s=inamaconsa-Voice-Sip-pbx > c=IN IP4 50.X.X.X > b=CT:384 > t=0 0 > m=audio 13780 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=ptime:20 > a=sendrecv > m=video 18488 RTP/AVP 99 98 > a=rtpmap:99 H264/90000 > a=fmtp:99 > redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 > a=rtpmap:98 H263-1998/90000 > a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 > a=sendrecv > > --- > -- Called SIP/5001/84783842@ > > <--- Transmitting (NAT) to 190.X.X.1:41316 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP > 190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316 > From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 > To: <sip:[email protected]>;tag=as77fb37e2 > Call-ID: [email protected] > CSeq: 102 INVITE > Server: inmaconsa-Voice-Sip-ipbx > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE > Supported: replaces, timer > Contact: <sip:[email protected]:5060> > Content-Length: 0 > > > <------------> > > <--- SIP read from UDP:190.53.38.203:5060 ---> > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > 50.X.X.X:5060;branch=z9hG4bK374c2247;received=50.X.X.X;rport=5060 > From: "Operadora" <sip:[email protected]>;tag=as3708c762 > To: <sip:84783842%[email protected]>;tag=as4bb74f30 > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > Server: VoxStack Wireless Gateway > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH > Supported: replaces, timer > Content-Length: 0 > > <-------------> > --- (10 headers 0 lines) --- > Transmitting (NAT) to 190.53.38.203:5060: > ACK sip:84783842%[email protected] SIP/2.0 > Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport > Max-Forwards: 70 > From: "Operadora" <sip:[email protected]>;tag=as3708c762 > To: <sip:84783842%[email protected]>;tag=as4bb74f30 > Contact: <sip:[email protected]:5060> > Call-ID: [email protected]:5060 > CSeq: 102 ACK > User-Agent: inmaconsa-Voice-Sip-ipbx > Content-Length: 0 > > > --- > [Jan 19 14:17:53] WARNING[11596][C-0000003d]: chan_sip.c:23037 > handle_response_invite: Received response: "Forbidden" from > '"Operadora" <sip:[email protected]>;tag=as3708c762' > Scheduling destruction of SIP dialog > '[email protected]:5060' in 32000 ms (Method: > INVITE) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [984783842@to_pstn:2] Busy("SIP/101-0000004e", "3") > in new stack > > <--- Reliably Transmitting (NAT) to 190.X.X.1:41316 ---> > SIP/2.0 486 Busy Here > Via: SIP/2.0/UDP > 190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316 > From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 > To: <sip:[email protected]>;tag=as77fb37e2 > Call-ID: [email protected] > CSeq: 102 INVITE > Server: inmaconsa-Voice-Sip-ipbx > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE > Supported: replaces, timer > X-Asterisk-HangupCause: Call Rejected > X-Asterisk-HangupCauseCode: 21 > Content-Length: 0 > > > <------------> > == Spawn extension (to_pstn, 984783842, 2) exited non-zero on > 'SIP/101-0000004e' > > <--- SIP read from UDP:190.X.X.1:41316 ---> > ACK sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-61b74f36 > From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 > To: <sip:[email protected]>;tag=as30070ac7 > Call-ID: [email protected] > CSeq: 101 ACK > Max-Forwards: 70 > Contact: "101" <sip:[email protected]:41316> > User-Agent: Cisco/SPA508G-7.5.6 > Content-Length: 0 > > <-------------> > --- (10 headers 0 lines) --- > Retransmitting #1 (NAT) to 190.X.X.1:41316: > SIP/2.0 486 Busy Here > Via: SIP/2.0/UDP > 190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316 > From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 > To: <sip:[email protected]>;tag=as77fb37e2 > Call-ID: [email protected] > CSeq: 102 INVITE > Server: inmaconsa-Voice-Sip-ipbx > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE > Supported: replaces, timer > X-Asterisk-HangupCause: Call Rejected > X-Asterisk-HangupCauseCode: 21 > Content-Length: 0 > > 2015-01-19 10:24 GMT-06:00 ricky gutierrez <[email protected]>: >> Hi list, I write on the list looking for help, buy a openvox gw gsm >> for four channels and I'm a little disappointed with the support >> openvox, for some reason , The call doesn?t get trough >> >> support tells me it was my asterisk server, but does not really work >> me and my internal calls are working perfectly, I tested with another >> sangoma FXO gateway and works perfectly. >> >> the problem is that support openvox is Chinese and the difference in >> time zone is high. >> >> my trunk is connected >> >> 5001/5001 X.X.X.X D Yes >> Yes 5060 >> >> Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline] >> >> I follow this guide , but not work >> >> http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf >> >> -- >> rickygm >> >> http://gnuforever.homelinux.com > > > > -- > rickygm > > http://gnuforever.homelinux.com > > > > ------------------------------ > > Message: 7 > Date: Tue, 20 Jan 2015 09:39:58 +0000 > From: A J Stiles <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Subject: Re: [asterisk-users] SEMI-OFFTOPIC openvox > Message-ID: <[email protected]> > Content-Type: Text/Plain; charset="utf-8" > >> On Monday 19 Jan 2015, ricky gutierrez wrote: >> Hi list, I write on the list looking for help, buy a openvox gw gsm >> for four channels and I'm a little disappointed with the support >> openvox, for some reason , The call doesn?t get trough >> >> support tells me it was my asterisk server, but does not really work >> me and my internal calls are working perfectly, I tested with another >> sangoma FXO gateway and works perfectly. >> >> the problem is that support openvox is Chinese and the difference in >> time zone is high. >> >> my trunk is connected >> >> 5001/5001 X.X.X.X D Yes >> Yes 5060 >> >> Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline] >> >> I follow this guide , but not work >> >> http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_ >> of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf > > I've had some experience with OpenVox GSM cards and chan_extra. Their > support > isn't great; they like if you can give them ssh access to your box, and you > will need to ask questions afterwards to find out what they did in there, but > they did manage to sort out an obscure problem for me and explained enough > for > me to work out what had been the matter in the first place. > > As far as I can work out, their GSM gateway appliances seem to be some kind > of > server motherboard with GSM cards and a pre-installed Linux, Asterisk and > chan_extra; but I've not had direct experience of them, having built my own > boxes using G400P and/or G400E cards in my favourite supplier's motherboards. > > Oh, and finally, if you're using any kind of GSM gateway, be careful! > Otherwise, you will end up incurring the wrath of your telco -- "unlimited" > often does not really mean unlimited, and the only way to find out what the > limit actually is is to exceed it. > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > > > ------------------------------ > > Message: 8 > Date: Tue, 20 Jan 2015 13:59:36 +0000 > From: "Haley,Scott A" <[email protected]> > To: "[email protected]" > <[email protected]> > Subject: Re: [asterisk-users] MWI issue > Message-ID: <[email protected]> > Content-Type: text/plain; charset="utf-8" > > I have a situation that I need help with. I have 2 phone systems, 1 Asterisk > and 1 Avaya. All voicemail is kept on the Avaya system. Whenever a call comes > into an extension that the Asterisk server owns, I re-direct it to a > different number that is owned by the Avaya System. If that Avaya extension > does not answer it, I send it to the voicemail on the Avaya Messaging system > for the extension that it came in on the Asterisk box. > > Once that happens, I need to send a MWI indicator to an application on the > desktop of the Avaya User that there is a voicemail for that mailbox. > > I see the SIP Notify come in from Avaya for the extension (I did this with a > tcpdump). My question is how do I configure Asterisk to act on that request > and call an agi program to do what I want. > > Any help would be appreciated. > > Thanks, > Scott Haley > > > > If you are not the intended recipient of this message (including > attachments), or if you have received this message in error, immediately > notify us and delete it and any attachments. > > If you do not wish to receive any email messages from us, excluding > administrative communications, please email this request to > [email protected] along with the email address you wish to unsubscribe. > > For important additional information related to this email, visit > www.edwardjones.com/US_email_disclosure<http://www.edwardjones.com/US_email_disclosure>. > Edward D. Jones & Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. > Louis, MO 63131 ? Edward Jones. All rights reserved. > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150120/4ea4765c/attachment.html> > > ------------------------------ > > _______________________________________________ > --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 > > End of asterisk-users Digest, Vol 126, Issue 18 > *********************************************** -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
