Re: [asterisk-users] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Hello Matthew, My version is Asterisk 1.6.2.9. Or have you seen NAT? I have no NAT on my network. Have you seen my little diagram above? Here it is: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) My Asterisk server has two NIC/interfaces. - 1 interface with public IP (109.69.217.6 to talk with SIP friends) - 1 interface with internal ip (10.4.0.1 to talk with SIP gateway's) SIP friend should not even know that the call is routed to the SIP/PSTN gateway. It could be a SIP trunk to a SIP provider Internet, the user does not have to know... Best regards, Mickael 2013/6/13 Matthew J. Roth mr...@imminc.com Mickael MONSIEUR wrote: I have a standard Asterisk configuration: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6 ;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Mickael, What version of Asterisk are you running? Is the Asterisk server outside and the SIP gateway to PSTN converter inside of a NAT? What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
Re: [asterisk-users] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
On Thursday 13 June 2013, Mickael MONSIEUR wrote: Hello Matthew, My version is Asterisk 1.6.2.9. Or have you seen NAT? I have no NAT on my network. Have you seen my little diagram above? Here it is: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) My Asterisk server has two NIC/interfaces. And it's obviously doing NAT, if anything plugged into one interface can see anything plugged into the other. The important question is: Does it work? Because if so, leave it alone. IP addresses are not secret. If anything in your network depends on someone on the outside not knowing one or more of your inside IP addresses, then you are doing it wrong. -- AJS Answers come *after* questions. -- _ -- 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
[asterisk-users] A quick question in terms of DAHDI channel
Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI core show channeltypes I would have response like: connect*CLI core show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- USTMUNISTIM Channel Driver no yes no Phone Standard Linux Telephony API Driver no yes no Console OSS Console Channel Driver no yes no Skinny Skinny Client Control Protocol (Skinny) yes yes no Local Local Proxy Channel Driver yes yes no SIP Session Initiation Protocol (SIP)yes yes yes Agent Call Agent Proxy Channel yes yes no MGCPMedia Gateway Control Protocol (MGCP)yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes MulticastR Multicast RTP Paging Channel Driver no no no Bridge Bridge Interaction Channel no no no -- 11 channel drivers registered. But right now, I am planing to connect a PRI trunk to this Asterisk. so I put in a PRI card install dahdi-linux-complete-2.6.2 and libpri-1.4.14. Afterward, dahdi_tool is able to find PRI board, and all channels. But my question is when I try to send call to DAHDI channel in the dial plan, CLI print out a warning saying [Jun 13 07:48:21] WARNING[2393]: channel.c:5606 ast_request: No channel type registered for 'DAHDI' According to my description above, it make sense, since my Asterisk does not install DAHDI channel before. Therefore my question is in my case, it is required to re-intall whole Asterisk, or there is some other way that I just could only install DAHDI channel. I did some google search. but I didn't find a proper answer. Thanks for your help. longst -- _ -- 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
Re: [asterisk-users] A quick question in terms of DAHDI channel
Do you see the channel driver chan_dahdi.so in /usr/lib/asterisk/modules? If not, go back to the * src dir, issue a ./configure, then make make install and check what * got this time. If you have played with menuselect you might have to check these settings, too. jg -- _ -- 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
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Hello Brian, Thank you so much On 6/12/13, Brian LaVallee b.laval...@globaltank.jp wrote: Hi Nick, Going from DS1 to OC-n is a multi-step process. Typically requiring a hardware device to handle each MUX step. But you can find hardware that handles multiple MUX steps together. The connection is coming into our premise on the OC-n transport. The question now is should we have it multiplexed as DS1 or VT1.5s to the DS3s. What is common today, I think DS1 VT1.5s mappings are more flexible? VT1.5 is just a raw OC-n channel containing a single DS1. An M13 device converts between DS3 and DS1. Understood!!! A DACS (DCS or DXC) provides M13 conversion, sometimes even capable of extracting the raw VT1.5 signal directly to DS1. The ILEC transport option you choose really depends on the terminating interface. Do you want to connect with a DS3 or OC-n? The transport is coming in as OC-n. What I am trying to figure out are the advantages of mapping the STS-1 using DS1s or VT1.5s. No matter what hardware you choose, you will need to convert to single copper pairs (DS1/T1) to connect to your Asterisk boxes. So an M13 or DCS will be necessary to reach the DS1 level. The device you choose depends on budget and growth expectations. Typically a DCS is an expensive investment, handling hundreds of DS3's. An M13 device is typically a small unit that handles one or two DS3's. This is almost understood. Is an M13 device basically a MUX (In our case STS-1-DS1)? From there we would plug the signaling into the Quad Digiums as you mentioned (this is where I get more comfortable). Could you kindly post a link to an entry level DCS with OC-n interfacing and M13s being used today. That way I can see what functionality each provides and determine which better suits our need. I am guessing, but hate to presume: M13: Adtran MX2800 DCS: Mediant 3000, Metaswitch 0610 etc.. The advantage comes when you add the 29th DS1. With VT1.5 it's just adding a single channel, DS3 will require another whole DS3 to get an additional DS1. This is why we are going SONET. It's a new transport layer for me compared to DS3s, and want to make sure I can put everything together at the network level. Sincerely, Brian LaVallee Thank you kindly, Nick from Montreal. -- _ -- 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
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
On 6/12/13, Don Kelly d...@donkelly.biz wrote: Is there an OC-n to SIP solution that makes sense? --Don Hello Don, what will be coming out of the network discussed above would be SIP. Kind Regards, Nick. -- _ -- 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
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Correction: I think VT1.5s mappings are more flexible? Sorry! N. -- _ -- 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
[asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
Hi All, I am looking for a way to troubleshoot issues with TDM (E1) trunks with a provider. Currently with SIP trunks I am using tcpdump to perform packet captures between our gateways and the SIP providers IPs, capturing traffic on all ports, to include both the SIP messages and the RTP stream. How can I achieve a similar result on TDM links connected to TDM cards in Asterisk servers, where by I can see the signalling (like the SIP message) and the audio stream (like the RTP stream) in my packet captures? If it helps, the end goal is to create something like the packet captures I am making so I can see the control signals and audio streams (in and out) for troubleshooting one way audio issues for example. So, am I sending audio to the TDM provider, are they sending it to me, have we both signalled correctly to start/stop sending audio, etc. Many thanks, James. -- _ -- 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
Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
On Thu, Jun 13, 2013 at 9:23 AM, James Bensley jwbens...@gmail.com wrote: Hi All, I am looking for a way to troubleshoot issues with TDM (E1) trunks with a provider. Currently with SIP trunks I am using tcpdump to perform packet captures between our gateways and the SIP providers IPs, capturing traffic on all ports, to include both the SIP messages and the RTP stream. How can I achieve a similar result on TDM links connected to TDM cards in Asterisk servers, where by I can see the signalling (like the SIP message) and the audio stream (like the RTP stream) in my packet captures? If it helps, the end goal is to create something like the packet captures I am making so I can see the control signals and audio streams (in and out) for troubleshooting one way audio issues for example. So, am I sending audio to the TDM provider, are they sending it to me, have we both signalled correctly to start/stop sending audio, etc. Many thanks, James. Is it PRI? You can see PRI debug info on the console. Extremely valuable in troubleshooting. http://www.voip-info.org/wiki/view/Asterisk+CLI Zap channel commands zap destroy channel: Destroy a channel zap show channels: Show active zapata channels zap show channel: Show information on a channel zap show status: lists all the Zaptel spans. A span will apear here whether or not its channels are configured with chan_zap. zap show cadences: Show the configured ring cadences (available e.g with Zap/1r2). zap set swgain(= 1.6): set the (software) gain for a hannel. Temporary equivalents of rxgain and txgain in zapata.conf. zap set hwgain(=1.6): set the hardware gain for channels that support it. zap set dnd(=1.6) set a channel's do-not-disturb mode on or off. The following commands are available if the channel is built with support for libpri: pri debug span: Enables PRI debugging on a span pri intense debug span: Enables REALLY INTENSE PRI debugging pri no debug span: Disables PRI debugging on a span pri show spans: List spans and their status. pri show span: Information about a span. pri show debug: show where debug is enabled. -- _ -- 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
Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
Hi! Depending which TDM board you are using there might already be tool to get a pcap trace. E.g. if you have a Sangoma board, the wanpipemon utility has a -pcap option. I don't know about other boards. Wireshark already comes with basic support for ISDN protocols, so now work is needed here. jg -- _ -- 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
Re: [asterisk-users] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Mickael MONSIEUR wrote: My version is Asterisk 1.6.2.9. Or have you seen NAT ? I have no NAT on my network . Have you seen my little diagram above ? Here it is: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) My Asterisk server has two NIC/interfaces. - 1 interface with public IP (109.69.217.6 to talk with SIP friends ) - 1 interface with internal ip ( 10.4.0.1 to talk with SIP gateway's) SIP friend should not even know that the call is routed to the SIP /PSTN gateway . It could be a SIP trunk to a SIP provider Internet , the user does not have to know. .. Mickael, It's hard to be certain without seeing a full SIP trace, but I think the INVITE with the internal IP is actually a re-INVITE that Asterisk is sending to establish a native bridge between the SIP friend and the SIP gateway to PSTN converter. This would allow the endpoints to send their media directly to one another, but in your case I'd expect it to cause one-way audio because the SIP friend shouldn't be able to send RTP packets to the internal IP. If it's a re-INVITE, start by reconfiguring Asterisk with directmedia=no in the [general] section of sip.conf and for all of the endpoints involved in the calls. That should completely eliminate the re-INVITEs at the expense of relaying all RTP through Asterisk, even for calls between two phones on the internal network. After you've confirmed that internal IPs are no longer being sent to external endpoints you can start fine-tuning the NAT SUPPORT and MEDIA HANDLING settings in sip.conf to only allow re-INVITEs when appropriate for your environment. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Verizon (NE ILEC) has SIP handoff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, June 13, 2013 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3 On 6/12/13, Don Kelly d...@donkelly.biz wrote: Is there an OC-n to SIP solution that makes sense? --Don Hello Don, what will be coming out of the network discussed above would be SIP. Kind Regards, Nick. -- _ -- 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
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
On 6/13/13, Eric Wieling ewiel...@nyigc.com wrote: Verizon (NE ILEC) has SIP handoff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, June 13, 2013 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3 Hello Eric, Thank you so much for your response. Is this an ISUP-IP interconnect (i.e., SS7IP), or are you referring to the traditional DID based VoIP. In either case, do you have a contact I can get a hold of. Kind Regards, Nick. -- _ -- 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
[asterisk-users] Problem with CEL logging and channel bridging
Hi, I've already post this to the forum three days ago, sorry if it's sounds like a crosspost, but I've got no replies, so I'm trying other channels :) This is the link to the forum post if someone prefer to reply here: http://forums.asterisk.org/viewtopic.php?f=1t=86985 I'm using Asterisk 1.8.20.0 (the freepbx build) with CEL logging activated. I'm using CEL because in our pbx we have different queues and trunks serving different customers (we are an inbound call center) and we need to detect when and how we have to bill our customers. I'm facing an issue with the call transfer, for example I have: - call entering a queue - operator answer the call - operator make an outgoing call to reach the customer - operator put in communication the ingoing call with the outgoing this result in various channel to be created/destroyed, and I'm using bridge events to detect what is going on with the call. In this case I have (I've hidden CHAN_START,ANSWER and HANGUP events because they have no useful information in this case): ++---+-+---+-+--+-+-+--+ | id | eventtype | eventtime | exten | context | channame | appname | appdata | peer | ++---+-+---+-+--+-+-+--+ | 965224 | BRIDGE_START | 2013-06-10 10:15:18 | 20| ext-queues | DAHDI/i1/96034296-30a3 | Queue | 20,t,, | Local/1004@from-queue-00019c34;1 | | 965226 | BRIDGE_START | 2013-06-10 10:15:18 | s | macro-dial-one | Local/1004@from-queue-00019c34;2 | Dial| SIP/1004,,trM(auto-blkvm) | SIP/1004-40ce| | 965340 | BRIDGE_UPDATE | 2013-06-10 10:16:08 | s | macro-dialout-trunk | Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| IAX2/issuegroup-17175| | 965513 | BRIDGE_END| 2013-06-10 10:18:15 | 20| ext-queues | DAHDI/i1/96034296-30a3 | Queue | 20,t,, | Local/1004@from-queue-00019c34;1 | | 965515 | BRIDGE_END| 2013-06-10 10:18:15 | s | macro-dialout-trunk | Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| IAX2/issuegroup-17175| ++---+-+---+-+--+-+-+--+ The first BRIDGE_START is the connection between the inbound call (DAHDI/i1/96034296-30a3) and the local phone (Local/1004@from-queue-00019c34;1), the second BRIDGE_START is the connection between the local phone (Local/1004@from-queue-00019c34;2) and the outgoing call (SIP/1004-40ce) that is going out by a IAX trunk. After that I have a BRIGDE_UPDATE event where no field make me know which channel is being updated, I only have the channame (Local/1004@from-queue-00019c34;2) that is the channel being bridged out and the outgoing channel (IAX2/issuegroup-17175), but I have no information that in fact the ingoing call (DAHDI/i1/96034296-30a3) is being bridged to the outgoing channel. I have no other event (TRANSFER or something like that) to know what is going on. In my cel.conf I have: apps=queue events=CHAN_START,CHAN_END, APP_START,APP_END, ANSWER,HANGUP, BRIDGE_START,BRIDGE_END,BRIDGE_UPDATE, BLINDTRANSFER,ATTENDEDTRANSFER,TRANSFER, PICKUP, FORWARD, PARK_START,PARK_END, LINKEDID_END Should I change something in my configuration or it's wrong to rely on bridges to follow a call? What kind of event should I follow to be sure to catch where the call is going? Thank you for any suggestion! -- _ -- 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
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
They offer standard SIP DIDs.I don't have a sales contact (others deal with that), but if Google should have some links. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, June 13, 2013 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3 On 6/13/13, Eric Wieling ewiel...@nyigc.com wrote: Verizon (NE ILEC) has SIP handoff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, June 13, 2013 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3 Hello Eric, Thank you so much for your response. Is this an ISUP-IP interconnect (i.e., SS7IP), or are you referring to the traditional DID based VoIP. In either case, do you have a contact I can get a hold of. Kind Regards, Nick. -- _ -- 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
Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
In article caawx_pvgza965ljxwrna_tkz2hkhqubokngesssje3bkkve...@mail.gmail.com, James Bensley jwbens...@gmail.com wrote: Hi All, I am looking for a way to troubleshoot issues with TDM (E1) trunks with a provider. Currently with SIP trunks I am using tcpdump to perform packet captures between our gateways and the SIP providers IPs, capturing traffic on all ports, to include both the SIP messages and the RTP stream. How can I achieve a similar result on TDM links connected to TDM cards in Asterisk servers, where by I can see the signalling (like the SIP message) and the audio stream (like the RTP stream) in my packet captures? If it helps, the end goal is to create something like the packet captures I am making so I can see the control signals and audio streams (in and out) for troubleshooting one way audio issues for example. So, am I sending audio to the TDM provider, are they sending it to me, have we both signalled correctly to start/stop sending audio, etc. A google for DAHDI pcap led me to the following page: http://forums.digium.com/viewtopic.php?f=1t=82833 That suggests that DAHDI 2.6 onwards can be compiled with pcap support, that presumably provides the drivers with the ability to capture to pcap files, and a tool to control it. Presumably a recent version of Wiresharl will then be able to interpret the captured files. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- 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
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Hello Eric, Thank your for your reponse. We are discussing interconnects at a different level. We are more interested in SS7 or ISUP-IP SS7IP type interconnects. There are many people that offer DIDs channels etc. over the internet. Including us. N. -- _ -- 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
Re: [asterisk-users] A quick question in terms of DAHDI channel
If you do have chan_dahdi.so already, make sure you have a valid chan_dahdi.conf file and try module load chan_dahdi in the CLI. -- Eric Cooper e c c @ c m u . e d u -- _ -- 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
[asterisk-users] asterisk fax in debian
Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to send to email the voicemails. i would like to get rid of the analog fax machine and use asterisk to send/receive faxes. I do have a PSTN line with a SPA3102 adapter to interface it to asterisk. The number of the PSTN line is dedicated to faxing only. So i would like to: -receive faxes to asterisk and then send it as PDFs to an email address -Send from my PC a fax directly. is there any guide on how to do that since i got lost with all of it? -- _ -- 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
[asterisk-users] No audio until you put call on hold...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have been struggling with an audio issue for a week now and have not been able to solve it. We have an Asterisk server (now running 11.4 but started with 1.8) with several sip phones on an internal network and a SIP trunk for external calls. We recently put several phones in service that connect via the Internet to the server. All NAT settings and port configurations were done and all phones register. The problem we have is that when external phones dial a pstn number they get no audio. We found that if you dial and put the call on hold for a couple second you then get audio on the call. I really do not know what else I can check in the configuration. Why would putting the call on hold get the audio flowing? Any ideas or recommendations? - -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.18 (Darwin) Comment: GPGTools - http://gpgtools.org Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iEYEARECAAYFAlG5+sUACgkQqmNh+MyHzx56+wCfWCLoqlm3Loviat2zJJWbKsL+ Om4AoKI+/db48174uetU+2DAjvcP1S2c =qmvr -END PGP SIGNATURE- -- _ -- 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
[asterisk-users] Codec Negotiation problem
Hi there I have asterisk 10.11.1 which seems to have problem negotiating codec. Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, h263p. I have tried similar combination of codecs and SIP phone but when making a video call, it report Peer doesn't provide video. It seems Asterisk is failing to set capability correct. Both codecs are enabled on the SIP Phones --- (12 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.10.129:53188 Peer doesn't provide video Here is a sip show peer output and log when making calls. localhost*CLI sip show peer 1003 * Name : 1003 Description : Secret : Set MD5Secret: Not set Remote Secret: Not set Context : video-users Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1003@device VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : device 1003 MaxCallBR: 384 kbps Expire : 3605 Insecure : no Force rport : Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.10.10.129:48464 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1003 SIP Options : (none) Codecs : (alaw|h263p) Codec Order : (alaw:20,h263p:0) Auto-Framing : No Status : OK (8 ms) Useragent: X-Lite release 4.5.2 stamp 70142 Reg. Contact : sip:1003@10.10.10.129:48464;rinstance=cf0c3558f05c89dc Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No localhost*CLI sip show peer 1004 * Name : 1004 Description : Secret : Set MD5Secret: Not set Remote Secret: Not set Context : video-users Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1004@device VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : device 1004 MaxCallBR: 384 kbps Expire : 893 Insecure : no Force rport : Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.10.10.107:21769 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1004 SIP Options : (none) Codecs : (alaw|h263p) Codec Order : (alaw:20,h263p:0) Auto-Framing : No Status : OK (2 ms) Useragent: Grandstream GXV3175v2 1.0.1.19 Reg. Contact : sip:1004@10.10.10.107:21769 Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No localhost*CLI - --- (8 headers 0 lines) --- --- SIP read from UDP:10.10.10.129:48464 --- INVITE sip:1004@10.10.10.105 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;rport Max-Forwards: 70 Contact: sip:1003@10.10.10.129:48464 To: sip:1004@10.10.10.105 From: SAMsip:1003@10.10.10.105;tag=0c90cc0c Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 4.5.2 stamp 70142 Authorization: Digest username=1003,realm=10.10.10.105,nonce=05e8af6e,uri=sip:1004@10.10.10.105,response=20e63a04aa86d6ec1d1e045c05159b39,algorithm=MD5 Content-Length: 418 v=0 o=- 13015615910543193 1 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2
Re: [asterisk-users] Codec Negotiation problem
On Thu, Jun 13, 2013 at 12:04 PM, resea...@businesstz.com wrote: Hi there I have asterisk 10.11.1 which seems to have problem negotiating codec. Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, h263p. I have tried similar combination of codecs and SIP phone but when making a video call, it report Peer doesn't provide video. It seems Asterisk is failing to set capability correct. Both codecs are enabled on the SIP Phones snip The 200 OK response from the called XLite phone is declining the video stream: --- SIP read from UDP:10.10.10.129:48464 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060 Contact: sip:1003@10.10.10.129:48464 To: SAMsip:1003@10.10.10.105;tag=0c90cc0c From: sip:1004@10.10.10.105;tag=as24914503 Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 234 v=0 o=- 13015615910543193 2 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 0 RTP/AVP 115 - --- (12 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Note that the port for the video stream is set to 0. Asterisk is doing the correct thing: it notes that the answer to its offer declined the video stream, so it disables video for the call between the two endpoints. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] blocking spammer by callerID name
I have a subroutine to block spammer by CALLERID(number)
exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1)
exten = 4,n,Set(goaway=${CALLERID(number):0:2})
exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)
but I just got another spammer (automated calls) who rotates his callerID number that starts with valid area code so blocking by prefix is not practical but it seems to
me he uses the
same (or few same) caller name like:
Brit. Columbia 16047726633
KHAN SHARON 16042984429
Brit. Columbia 16042231781
So I was thinking the same subroutine can be used to block by CALLERID(name),
isn't it:
exten = 4,n,Set(goaway2=${CALLERID(name):0:11})
exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1)
The spammer is soliciting lowering credit card interest charges etc. anybody
know who it is :-/
--
Joseph
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Re: [asterisk-users] blocking spammer by callerID name
Google the number and you can probably find other complaints and
possibly who it is. Not that it will matter, there's nothing you can
do but block it.
My approach to call filtering is:
Deny All
Allow Some
I have a whitelist of callers I always want to accept that may include
businesses outside my local area code. If my dialplan doesn't
recognize the incoming number I send them to a voicemail mail where
they have to press 5 to leave a message. That knocks out the robo
dialers. Then I google the number and if it's a spammer, I add them
to a blacklist where the call is dropped immediately. Really no point
in playing funny or cute messages to them or even telling them they
are blacklisted because it's usually an auto-dialer and a real person
doesn't hear it anyway.
On Thu, Jun 13, 2013 at 1:31 PM, Joseph syscon...@gmail.com wrote:
I have a subroutine to block spammer by CALLERID(number)
exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1)
exten = 4,n,Set(goaway=${CALLERID(number):0:2})
exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)
but I just got another spammer (automated calls) who rotates his callerID
number that starts with valid area code so blocking by prefix is not
practical but it seems to me he uses the same (or few same) caller name
like:
Brit. Columbia 16047726633
KHAN SHARON 16042984429
Brit. Columbia 16042231781
So I was thinking the same subroutine can be used to block by
CALLERID(name), isn't it:
exten = 4,n,Set(goaway2=${CALLERID(name):0:11})
exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1)
The spammer is soliciting lowering credit card interest charges etc. anybody
know who it is :-/
--
Joseph
--
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--
Chris
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Re: [asterisk-users] blocking spammer by callerID name
Thank you for input.
Good idea, I like your approach with press number to leave a message, this will definitely cut the robo-calls voice-mail.
Do you use database for white-list?
Can you post a section of your dial plan that deals with blocking?
This is a medical clinic so white-list, black-list is not a good solution but
it might be good for home use.
Thanks,
--
Joseph
On 06/13/13 14:30, Chris Gentle wrote:
Google the number and you can probably find other complaints and
possibly who it is. Not that it will matter, there's nothing you can
do but block it.
My approach to call filtering is:
Deny All
Allow Some
I have a whitelist of callers I always want to accept that may include
businesses outside my local area code. If my dialplan doesn't
recognize the incoming number I send them to a voicemail mail where
they have to press 5 to leave a message. That knocks out the robo
dialers. Then I google the number and if it's a spammer, I add them
to a blacklist where the call is dropped immediately. Really no point
in playing funny or cute messages to them or even telling them they
are blacklisted because it's usually an auto-dialer and a real person
doesn't hear it anyway.
On Thu, Jun 13, 2013 at 1:31 PM, Joseph syscon...@gmail.com wrote:
I have a subroutine to block spammer by CALLERID(number)
exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1)
exten = 4,n,Set(goaway=${CALLERID(number):0:2})
exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)
but I just got another spammer (automated calls) who rotates his callerID
number that starts with valid area code so blocking by prefix is not
practical but it seems to me he uses the same (or few same) caller name
like:
Brit. Columbia 16047726633
KHAN SHARON 16042984429
Brit. Columbia 16042231781
So I was thinking the same subroutine can be used to block by
CALLERID(name), isn't it:
exten = 4,n,Set(goaway2=${CALLERID(name):0:11})
exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1)
The spammer is soliciting lowering credit card interest charges etc. anybody
know who it is :-/
--
Joseph
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] asterisk fax in debian
Maybe this can help: http://ofps.oreilly.com/titles/9781449332426/asterisk-Fax.html Best. 2013/6/13 vortex binary.vor...@gmail.com Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to send to email the voicemails. i would like to get rid of the analog fax machine and use asterisk to send/receive faxes. I do have a PSTN line with a SPA3102 adapter to interface it to asterisk. The number of the PSTN line is dedicated to faxing only. So i would like to: -receive faxes to asterisk and then send it as PDFs to an email address -Send from my PC a fax directly. is there any guide on how to do that since i got lost with all of it? -- __**__**_ -- 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-**usershttp://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
[asterisk-users] voice recognition voicemail to email
I fuzzily recall someone posting a script that shuffled off voicemails to Google for conversion to text that could then be emailed. Anyone have any luck with that? Anything new out there? j -- _ -- 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
Re: [asterisk-users] blocking spammer by callerID name
On Thu, 2013-06-13 at 13:55 -0600, Joseph wrote: Good idea, I like your approach with press number to leave a message, this will definitely cut the robo-calls voice-mail. I do this, but without any white or black lists, and it works great. The greeting says press one for my wife, or two for me. That alone is enough to knock out virtually all the spammers (99% of them are robo-calls these days). Once 1 or 2 is pressed, the phones in the house will ring with a different ring code for each, and if there's no answer, the call goes to separate voice mail boxes for my wife and myself. Works great. That alone was worth the effort to install asterisk. Now, the cost of the telephony card, that's a different story. Never before have I had a PCI card that costs more than the rest of the machine combined. So I really wouldn't have done this if it weren't such a cool geek project :-) --Greg -- _ -- 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
Re: [asterisk-users] No audio until you put call on hold...
Carlos Chavez wrote: I have been struggling with an audio issue for a week now and have not been able to solve it. We have an Asterisk server (now running 11.4 but started with 1.8) with several sip phones on an internal network and a SIP trunk for external calls. We recently put several phones in service that connect via the Internet to the server. All NAT settings and port configurations were done and all phones register. The problem we have is that when external phones dial a pstn number they get no audio. We found that if you dial and put the call on hold for a couple second you then get audio on the call. I really do not know what else I can check in the configuration. Why would putting the call on hold get the audio flowing? Any ideas or recommendations? Carlos, Please provide SIP traces of both call legs (external phone to Asterisk and Asterisk to SIP trunk) annotated to show when the audio starts as well as the CLI output of 'sip show settings', 'sip show peer external phone', and 'sip show peer SIP trunk'. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
[asterisk-users] Light-weight voice recognition for IVR
Hello list, 'Just wondering if anyone can point to a very light-weight and easy to incorporate into Asterisk (v. 11.x) to handle a minimal set of responses, like: 0 - 9 yes no (maybe * and # for some people) The idea is that within an IVR menu, the caller could respond by speaking to the typical IVR options, like: For Archie, press or say 1 now For Veronica, press or say 2 now For Jughead, press or say 3 now (etc.) You have selected option 2 for Veronica, press 1 or say yes if this is correct. If a voice response was received (not a DTMF key press) indeterminate, some status would be useful (beyond just a timeout). It would be great if this was simple to code into the dialplan, much like like the current background/wait model for keypresses. Low cost or free would be nice too! Thanks for any suggestions. -- _ -- 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
Re: [asterisk-users] blocking spammer by callerID name
On Thu, Jun 13, 2013 at 02:32:22PM -0600, Greg Woods wrote: I do this, but without any white or black lists, and it works great. The greeting says press one for my wife, or two for me. That alone is enough to knock out virtually all the spammers (99% of them are robo-calls these days). Once 1 or 2 is pressed, the phones in the house will ring with a different ring code for each, and if there's no answer, the call goes to separate voice mail boxes for my wife and myself. [...] Greg, would you mind posting your dialplan? I think it would be useful and instructive for newbies like me. Thanks. -- Eric Cooper e c c @ c m u . e d u -- _ -- 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
Re: [asterisk-users] blocking spammer by callerID name
On Thu, 2013-06-13 at 18:14 -0400, Eric Cooper wrote: Greg, would you mind posting your dialplan? It may be a day or two before I can do that, as of course I will need to sanitize it (remove passwords, commented lines, etc.) --Greg -- _ -- 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
Re: [asterisk-users] blocking spammer by callerID name
When I play: exten = s,n,Background(welcome) and press extension 1 the system will not jump to this extension immediately, there is a few sec. pause. I think because I have an extensions 1 and 11 in my system. Is there a way to tell Background to execute the first match? I see there are two options: 's' and 'm' but none of them return immediately I put voicemail on exten '0' (nothing else starts with '0') but when the message is playing and I hit '0' there is a two or three seconds pause before the Voicemail box rings. Is there a way to jump to extension immediately? exten = 1,n,Background(T-to-leave-msg) exten = 1,n,Background(press-0) exten = 1,n,WaitExten(5) [voicemail11] exten = 0,1,Voicemail(11,b) exten = 0,n,Hangup() -- Joseph On 06/13/13 14:32, Greg Woods wrote: On Thu, 2013-06-13 at 13:55 -0600, Joseph wrote: Good idea, I like your approach with press number to leave a message, this will definitely cut the robo-calls voice-mail. I do this, but without any white or black lists, and it works great. The greeting says press one for my wife, or two for me. That alone is enough to knock out virtually all the spammers (99% of them are robo-calls these days). Once 1 or 2 is pressed, the phones in the house will ring with a different ring code for each, and if there's no answer, the call goes to separate voice mail boxes for my wife and myself. Works great. That alone was worth the effort to install asterisk. Now, the cost of the telephony card, that's a different story. Never before have I had a PCI card that costs more than the rest of the machine combined. So I really wouldn't have done this if it weren't such a cool geek project :-) --Greg -- _ -- 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
Re: [asterisk-users] blocking spammer by callerID name
Yeah, probably wouldn't work too well in a business environment where
you actually NEED to answer calls. I go to a lot of trouble to make
sure people can't get in touch with me. :)
I keep my blacklist and whitelist in AstDB. However, I maintain it in
a bash script so that I can update the script and then rebuild the
AstDB very quickly. If I lose my AstDB I can just rebuild it with the
script.
; Check the Asterisk database for blacklisted number
exten =
s,n,GotoIf(${DB_EXISTS(blacklisted/${CALLERID(num)})}?blacklisted,s,1)
Whitelist can be done the same way:
; Check the Asterisk database for whitelisted number
exten =
s,n,GotoIf(${DB_EXISTS(whitelisted/${CALLERID(num)})}?voicemail,abc,1)
I have a [screened] context that screens the calls and prompts for pressing 5
[screened] ;{{{
exten = s,1,Zapateller()
exten = s,n,Set(TSTAMP=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,n,NoOp(${TSTAMP})
exten = s,n,Monitor(wav,${TSTAMP}-${CALLERID(num)}-screened,m)
exten = s,n,Set(COUNT=1)
exten = s,n(loop),WaitExten(1)
exten = s,n,Background(privacy-screening-unidentified-calls)
exten = s,n,WaitExten(.5)
exten = s,n,Background(press-5)
exten = s,n,Background(T-to-leave-msg)
exten = s,n,WaitExten(3)
exten = s,n,Set(COUNT=$[${COUNT} + 1])
exten = s,n,GotoIf($[${COUNT} = 3]?loop)
exten = s,n,PlayBack(goodbye)
exten = s,n,StopMonitor()
exten = s,n,Hangup()
exten = 5,1,NoOp(Pressed 5)
exten = 5,n,PlayBack(tcg/pls-lv-msg-w-nam-phnnum)
exten = 5,n,StopMonitor()
exten = 5,n,GoSub(voicemail,tcg,1)
exten = 5,n,Hangup()
exten = i,1,Playback(option-is-invalid)
exten = i,n,Goto(99,msg)
;}}}
On Thu, Jun 13, 2013 at 2:55 PM, Joseph syscon...@gmail.com wrote:
Thank you for input.
Good idea, I like your approach with press number to leave a message,
this will definitely cut the robo-calls voice-mail. Do you use database for
white-list?
Can you post a section of your dial plan that deals with blocking?
This is a medical clinic so white-list, black-list is not a good solution
but it might be good for home use.
Thanks,
--
Joseph
On 06/13/13 14:30, Chris Gentle wrote:
Google the number and you can probably find other complaints and
possibly who it is. Not that it will matter, there's nothing you can
do but block it.
My approach to call filtering is:
Deny All
Allow Some
I have a whitelist of callers I always want to accept that may include
businesses outside my local area code. If my dialplan doesn't
recognize the incoming number I send them to a voicemail mail where
they have to press 5 to leave a message. That knocks out the robo
dialers. Then I google the number and if it's a spammer, I add them
to a blacklist where the call is dropped immediately. Really no point
in playing funny or cute messages to them or even telling them they
are blacklisted because it's usually an auto-dialer and a real person
doesn't hear it anyway.
On Thu, Jun 13, 2013 at 1:31 PM, Joseph syscon...@gmail.com wrote:
I have a subroutine to block spammer by CALLERID(number)
exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1)
exten = 4,n,Set(goaway=${CALLERID(number):0:2})
exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)
but I just got another spammer (automated calls) who rotates his callerID
number that starts with valid area code so blocking by prefix is not
practical but it seems to me he uses the same (or few same) caller name
like:
Brit. Columbia 16047726633
KHAN SHARON 16042984429
Brit. Columbia 16042231781
So I was thinking the same subroutine can be used to block by
CALLERID(name), isn't it:
exten = 4,n,Set(goaway2=${CALLERID(name):0:11})
exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1)
exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1)
The spammer is soliciting lowering credit card interest charges etc.
anybody
know who it is :-/
--
Joseph
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Re: [asterisk-users] A quick question in terms of DAHDI channel
hi:you have to install libpri,dahdi and asterisk for E1 cards. Best regards, James.zhuVega VOIP gateways, Sangoma Asterisk cards, SBC, NetBorder VOIP Gateway, LYNC-TDM, Transcodingwebsite: www.hiastar.com From: longst...@gmail.com Date: Thu, 13 Jun 2013 10:31:28 +0200 To: asterisk-users@lists.digium.com Subject: [asterisk-users] A quick question in terms of DAHDI channel Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI core show channeltypes I would have response like: connect*CLI core show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- USTMUNISTIM Channel Driver no yes no Phone Standard Linux Telephony API Driver no yes no Console OSS Console Channel Driver no yes no Skinny Skinny Client Control Protocol (Skinny) yes yes no Local Local Proxy Channel Driver yes yes no SIP Session Initiation Protocol (SIP)yes yes yes Agent Call Agent Proxy Channel yes yes no MGCPMedia Gateway Control Protocol (MGCP)yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes MulticastR Multicast RTP Paging Channel Driver no no no Bridge Bridge Interaction Channel no no no -- 11 channel drivers registered. But right now, I am planing to connect a PRI trunk to this Asterisk. so I put in a PRI card install dahdi-linux-complete-2.6.2 and libpri-1.4.14. Afterward, dahdi_tool is able to find PRI board, and all channels. But my question is when I try to send call to DAHDI channel in the dial plan, CLI print out a warning saying [Jun 13 07:48:21] WARNING[2393]: channel.c:5606 ast_request: No channel type registered for 'DAHDI' According to my description above, it make sense, since my Asterisk does not install DAHDI channel before. Therefore my question is in my case, it is required to re-intall whole Asterisk, or there is some other way that I just could only install DAHDI channel. I did some google search. but I didn't find a proper answer. Thanks for your help. longst -- _ -- 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
