marlon araujo
2015-Jan-21 13:38 UTC
[asterisk-users] asterisk-users Digest, Vol 126, Issue 18 mtr
You could use MTR command. Its a trace route improved. Marlon Araujo> On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > 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 > asterisk-users-request at lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner at lists.digium.com > > 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. <tjrlist at live.com> > To: Asterisk-Users List <asterisk-users at lists.digium.com> > Subject: [asterisk-users] sip show channelstats reliable? > Message-ID: <BLU173-W265CCDC9CB89501E36210ECD4A0 at phx.gbl> > 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. <tjrlist at live.com> > To: Asterisk-Users List <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: <BLU173-W470794F737AECEA2FCD353CD4A0 at phx.gbl> > 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: tjrlist at live.com > To: asterisk-users at lists.digium.com > 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 <EWieling at nyigc.com> > To: "tjrlist at live.com" <tjrlist at live.com>, Asterisk Users Mailing List > - Non-Commercial Discussion <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: > <616B4ECE1290D441AD56124FEBB03D082F43F2E5E7 at 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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: tjrlist at live.com<mailto:tjrlist at live.com> > To: asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com> > 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. <tjrlist at live.com> > To: Eric Wieling <ewieling at nyigc.com>, Asterisk-Users List > <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: <BLU173-W166D750594A2E845C58840CD4A0 at phx.gbl> > 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: EWieling at nyigc.com > To: tjrlist at live.com; asterisk-users at lists.digium.com > 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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: tjrlist at live.com > To: asterisk-users at lists.digium.com > 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 <sgriepentrog at digium.com> > To: tjrlist at live.com, Asterisk Users Mailing List - Non-Commercial > Discussion <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] sip show channelstats reliable? > Message-ID: > <CACrpESbTXJXAuPLNdbBMTWUMyH4ksv_zRL0aSrM-QnjHrmOVUg at 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. <tjrlist at live.com> 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: EWieling at nyigc.com >> To: tjrlist at live.com; asterisk-users at lists.digium.com >> 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:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *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: tjrlist at live.com >> To: asterisk-users at lists.digium.com >> 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 <xserverlinux at gmail.com> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] SEMI-OFFTOPIC openvox > Message-ID: > <CAL_GE3Q=bF6sngOsS=5dUEK5oe5pH3p7=R=nyN=buNqeAc5Nbg at 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:984783842 at 50.X.X.X:5060> > Content-Length: 0 > > > <------------> > -- Executing [984783842 at 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%40 at 190.53.38.203 SIP/2.0 > Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport > Max-Forwards: 70 > From: "Operadora" <sip:101 at 50.X.X.X>;tag=as3708c762 > To: <sip:84783842%40 at 190.53.38.203> > Contact: <sip:101 at 50.X.X.X:5060> > Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7 at 50.X.X.X: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:101 at 50.X.X.X>;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:101 at 50.X.X.X>;tag=35721c1e3f767ceao4 > To: <sip:984783842 at 50.X.X.X>;tag=as77fb37e2 > Call-ID: 7f55e32e-e4c6e11a at 172.16.8.179 > 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:984783842 at 50.X.X.X: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:101 at 50.X.X.X>;tag=as3708c762 > To: <sip:84783842%40 at 190.53.38.203>;tag=as4bb74f30 > Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7 at 50.X.X.X: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%40 at 190.53.38.203 SIP/2.0 > Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport > Max-Forwards: 70 > From: "Operadora" <sip:101 at 50.X.X.X>;tag=as3708c762 > To: <sip:84783842%40 at 190.53.38.203>;tag=as4bb74f30 > Contact: <sip:101 at 50.X.X.X:5060> > Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7 at 50.X.X.X: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:101 at 50.X.X.X>;tag=as3708c762' > Scheduling destruction of SIP dialog > '0c9236b922c5a99f6a1a797c7c3f9eb7 at 50.X.X.X:5060' in 32000 ms (Method: > INVITE) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [984783842 at 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:101 at 50.X.X.X>;tag=35721c1e3f767ceao4 > To: <sip:984783842 at 50.X.X.X>;tag=as77fb37e2 > Call-ID: 7f55e32e-e4c6e11a at 172.16.8.179 > 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:984783842 at 50.X.X.X SIP/2.0 > Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-61b74f36 > From: "101" <sip:101 at 50.X.X.X>;tag=35721c1e3f767ceao4 > To: <sip:984783842 at 50.X.X.X>;tag=as30070ac7 > Call-ID: 7f55e32e-e4c6e11a at 172.16.8.179 > CSeq: 101 ACK > Max-Forwards: 70 > Contact: "101" <sip:101 at 190.X.X.1:41316> > 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:101 at 50.X.X.X>;tag=35721c1e3f767ceao4 > To: <sip:984783842 at 50.X.X.X>;tag=as77fb37e2 > Call-ID: 7f55e32e-e4c6e11a at 172.16.8.179 > 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 <xserverlinux at gmail.com>: >> 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 <asterisk_list at earthshod.co.uk> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] SEMI-OFFTOPIC openvox > Message-ID: <201501200939.58525.asterisk_list at earthshod.co.uk> > 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" <scott.haley at edwardjones.com> > To: "asterisk-users at lists.digium.com" > <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] MWI issue > Message-ID: <E0917100-D825-4336-8EF0-6961913A3C20 at edwardjones.com> > 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 messages at edwardjones.com 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 > ***********************************************
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