Re: [asterisk-users] Polycom UC 4.x Unreachable
Solved it! Turns out UCS Polycoms are quite picky about blank callerids, to the extant they ignore those packets completely. My global "callerid=" in sip.conf was intentionally blank. In ten years, in never caused a problem. By setting to 0, the Polycoms that didn't respond to SIP OPTIONS (nor the NOTIFY for waiting messages) now work fine. If anyone is curious, the problem is easily reproduced in the dialplan by setting the callerid there to blank, then the UCS polycom will ignore that INVITE as well. Set the callerid to anything else and it'll ring. On 23 August 2017 at 19:29, John Covici <cov...@ccs.covici.com> wrote: > I always set it to no, but set the registration time to 60 seconds, > and that has always worked for me. > > On Wed, 23 Aug 2017 17:23:38 -0400, > Gary Reuter wrote: >> >> Hello, >> We've had dozens of Polycom 3.x firmware phones deployed and working >> great for years. >> Now I've finally been charged with the long-overdue task of figuring >> out why newer Polycom devices with 4.x firmware register fine but do >> not respond to SIP OPTIONS request and therefore always become >> UNREACHABLE if the sip qualify setting is set to yes. >> >> To my dismay, searches for solutions from others who have encountered >> this problem have given zero results. >> >> >> Thanks! >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > cov...@ccs.covici.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom UC 4.x Unreachable
Hello, We've had dozens of Polycom 3.x firmware phones deployed and working great for years. Now I've finally been charged with the long-overdue task of figuring out why newer Polycom devices with 4.x firmware register fine but do not respond to SIP OPTIONS request and therefore always become UNREACHABLE if the sip qualify setting is set to yes. To my dismay, searches for solutions from others who have encountered this problem have given zero results. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium board considerations
I need to create an updated Asterisk install. I'm planning on using FreePBX. I have markings on an old Digium board TDM2400P rev A2 TDM2400P Rev B DIGCN01ATDM2400P Is there any reason I shouldn't use this board? Are there better board options that have been improved that I should consider? Thanks, Gary Kuznitz WPM$LEX5.PM$ Description: Mail message body -- _ -- 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] One Way Audio with WebRTC (with external asterisk)
Hi Amit, My rtp.conf has the stunaddr listed and icesupport set to yes. It looks like the issue is that the media isn't being sent from 192.168.3.150 to 192.168.3.131 (chrome browser to asteriskrtc.local). When using asteriskrtc.local to originate the call (make a call directly from sipml client to another number on asteriskrtc.local or to a number on another asterisk server) audio flows both ways with no issue, it's just when asteriskgary.local is originating the call that there is no audio flowing from chrome to asteriskrtc.local. I should probably rephrase the above though to say that on tshark I can actually see the packets flowing (tshark host 192.168.3.150): 2.384874 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.384925 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.385060 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175 2.385256 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021 2.394891 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.415195 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.434063 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.434121 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.434296 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175 2.434462 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021 2.435083 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.455310 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.475009 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 Thanks again for your time! Kind Regards, Gary Shergill - Original Message - From: Amit Patkar a...@avhan.com To: asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 4:55:57 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Please check rtp.conf Look for stunaddr setting. You can try with google STUN server stunaddr = stun.l.google.com:19302 Thanks Regards, Amit Patkar On 5/21/2014 9:13 PM, Gary Shergill wrote: Hi again, Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works... Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting). However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local). Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: Gary Shergill gsherg...@gltd.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix. I have directmedia=no set for the user on asteriskrtc.local. My exact setup/scenario is below: - asteriskgary.local has a route to dial extensions on my Elastix server. - asteriskgary.local has a route to dial extensions on asteriskrtc.local server. - The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server. What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000): - User on asteriskgary.local places a call to 1000, his number is 6901 - 6901 answers on the web browser and begins to dial 1000 - 1000 answers and the call is established correctly - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) - 6901 can NEVER hear 1000 key: 192.168.3.127 - asteriskgary.local 192.168.3.131 - asteriskrtc.local 192.168.3.150 - machine running chrome browser where 6901 is logged on 192.168.3.100 - phone where 1000 is logged on (1000 can hear 6901) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160
[asterisk-users] One Way Audio with WebRTC (with external asterisk)
Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows. When I try making a call from the other asterisk server (asteriskgary.local) to asteriskrtc.local (all routes are set up) I am seeing the following behaviour: - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901 - 6901 sees the call and has the option to answer - 6901 answers the call - 6901 can hear 1000 talking - 1000 can not hear 6901 The weird thing is, sometimes it works, sometimes it doesn't... I think it has something to do with the port destination changing when the call is answered but I'm not sure (wireshark suggests that, as it says Port Unreachable). Has anyone tried this before and seen this issue? Or knows why it is and how to debug it? I can provide any logs required, I have some logs from when it works and doesn't. Thank you for your help. Kind Regards, Gary Shergill -- _ -- 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] One Way Audio with WebRTC (with external asterisk)
=webrtc hasiax = no hassip = yes encryption = yes avpf = yes icesupport = yes videosupport=no directmedia=no canreinvite=no You can see from the trace packets that sometimes asteriskgary.local sees no packets from asteriskrtc.local, and at the same time the packets on asteriskrtc.local show half the number of records (there is no Probation passed - setting RTP source address to 192.168.3.127:15942 which causes twice the number of packets, no idea if this is relevant though). Please ask if you need anything else. I'm totally stumped with this issue... Note that on asteriskgary.local ICE is not configured, I wouldn't have though it would need it as it isn't talking with the webrtc client itself, it is just talking to an Asterisk server (and that asterisk server is the one which talks to the webrtc client). Thank you. Kind Regards, Gary Shergill - Original Message - From: Amit Patkar a...@avhan.com To: asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 04:41:50 AM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Gary You need to check if ICE / STUN is configured. How are these extensions configured? If you are in private network, you might have to disable DirectMedia / reInvite for calls going between 2 asterisk boxes. I hope this helps to resolve your issue. *Thanks Regards,* Amit Patkar On 5/21/2014 2:26 PM, Gary Shergill wrote: Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows. When I try making a call from the other asterisk server (asteriskgary.local) to asteriskrtc.local (all routes are set up) I am seeing the following behaviour: - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901 - 6901 sees the call and has the option to answer - 6901 answers the call - 6901 can hear 1000 talking - 1000 can not hear 6901 The weird thing is, sometimes it works, sometimes it doesn't... I think it has something to do with the port destination changing when the call is answered but I'm not sure (wireshark suggests that, as it says Port Unreachable). Has anyone tried this before and seen this issue? Or knows why it is and how to debug it? I can provide any logs required, I have some logs from when it works and doesn't. Thank you for your help. Kind Regards, Gary Shergill -- _ -- 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] One Way Audio with WebRTC (with external asterisk)
Hi again, Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works... Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting). However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local). Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: Gary Shergill gsherg...@gltd.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix. I have directmedia=no set for the user on asteriskrtc.local. My exact setup/scenario is below: - asteriskgary.local has a route to dial extensions on my Elastix server. - asteriskgary.local has a route to dial extensions on asteriskrtc.local server. - The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server. What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000): - User on asteriskgary.local places a call to 1000, his number is 6901 - 6901 answers on the web browser and begins to dial 1000 - 1000 answers and the call is established correctly - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) - 6901 can NEVER hear 1000 key: 192.168.3.127 - asteriskgary.local 192.168.3.131 - asteriskrtc.local 192.168.3.150 - machine running chrome browser where 6901 is logged on 192.168.3.100 - phone where 1000 is logged on (1000 can hear 6901) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054709, ts 2304496624, len 000160) 0x7fe73c021740 -- Probation passed - setting RTP source address to 192.168.3.127:15942 Got RTP packet from192.168.3.127:15942 (type 00, seq 003763, ts 000160, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047008, ts 000160, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014309, ts 2304496791, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054710, ts 2304496784, len 000160) Got RTP packet from192.168.3.150:55148 (type 00, seq 014310, ts 2304496951, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054711, ts 2304496944, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003764, ts 000320, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047009, ts 000320, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014311, ts 2304497111, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054712, ts 2304497104, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003765, ts 000480, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047010, ts 000480, len 4294967284) (1000 can hear 6901) RTP TRACE ON asteriskgary.local ... Got RTP packet from192.168.3.131:17836 (type 00, seq 055375, ts 2304603184, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004428, ts 106560, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055376, ts 2304603344, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004429, ts 106720, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055377, ts 2304603504, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004430, ts 106880, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055378, ts 2304603664, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004431, ts 107040, len 000160) ... (no audio) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.127:17796 (type 00, seq 035016, ts 000640, len 000160) Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060981, ts 000640, len 4294967284) Got RTP packet from192.168.3.127:17796 (type 00, seq 035017, ts 000800, len 000160) Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060982, ts 000800, len 4294967284) Got RTP packet from
[asterisk-users] Direct DAHDI documentation
Hello, I wanted to switch from using Dialogic/Eicon cards to using Digium's T-1 cards. When I purchased a sample card the salesperson assured me there was documentation specific to the DAHDI interface. Now that I'm digging in, I'm finding it's documented a lot like Linux -- one must read the fairly uncommented source code. I don't have a problem with this generally, but here I just don't understand the divisions of labor between Asterisk, DAHDI Hardware, DAHDI kernel modules and Userland (me). (BTW, I do not wish to use Asterisk as we have numerous projects based on Dialogic/Eicon spanning some 20 years. My intent is to write a replacement look-a-like driver which uses Digium's cards instead of Dialogic's.) My specific issues are: 1) HDLC. Does the hardware have an HDLC controller, or is it the user's job to hunt for flags, frame the data and calc the FCS? 2) ISDN/PRI. Does the kernel module load Q.921/931 implementation or is this user's responsibility? I know there's a LIBPRI product, which I may use, but I have my own PRI library which was confirmance tested with ATT years ago. Either way, I'm not sure how the D-channel data is flowing. 3) I got the idea that B-channel data is collected by the kernel module in 8 sample blocks (1 ms). Does this mean I need to be reading it out/writing it in at that rate? I saw some buffering code, but wasn't sure if that was voicefile type playback/record or if all audio is treated without regard to its source/destination. I guess I could lock onto it at 1ms using Linux's HPET timer, although that sounds clumsy. 4) I can certainly convert between ulaw/linear to sum for conferencing, but it seems the kernel module might support that as well? Or at the least it seems the kernel module can support chan-to-chan connections. 5) I found some DTMF (FIR goertzel) code somewhere in DAHDI, but also in Asterisk. While I have such code in own library, am I to understand DTMF can be detected within the kernel module? I guess I really would like to see a doc on the overall concept of DAHDI hardware and its kernel module. I don't care how it's laid out, I'd just like to get my mind around it. Does anyone know of an example telephony C file that might show: 1) initialization of DAHDI spans 2) waiting for inbound events 3) answering a call 4) sending a voice file, recording a voice file 5) disconnection of calls 6) de-initialization And perhaps showing how two channels are connected to create a conversation? Thanks in advance, Gary -- _ -- 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] POKE from command line
Is it possible to issue the POKE to a end point from the CLI? Our asterisk servers is not seeing some end points drop off and I would like to create a script to manually check end points. Thanks! Gary -- _ -- 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] DIDForSale spam
I received the same spam myself. Regards, Gary Carr List users, Did anyone else recently receive spam from DIDForSale with the subject DIDForSale 2012 achievements? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. 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] call extension play sound file then connect caller
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Gary Carr Sent: Wednesday, October 03, 2012 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call extension play sound file then connect caller I am trying to setup a context to take a inbound call, hold the call, connect to an external number, play a sound file to the external number, then connect the inbound caller to the external number. My thought was to accept the call and place them in a parking lot. Then call the external number, play the sound file and connect the inbound caller to the external number. Is this even possible and if so, is this the best approach? Thank you in advance. You might look into FollowMe, especially if you want the external number to have a choice of whether or not to accept the call. A very high level overview is here: http://answers.oreilly.com/topic/2705-asterisk-how-to-use-followme-to-call-a-series-of-phone-numbers/ (though that gave me enough to get started) Thanks for the reply. I tried using FollowMe as it seemed like the perfect solution, however I was unable to play the sound file then connect the caller. I would like to bypass the need to press the 1 to accept the call. Thanks Again! -- _ -- 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] call extension play sound file then connect caller
I am trying to setup a context to take a inbound call, hold the call, connect to an external number, play a sound file to the external number, then connect the inbound caller to the external number. My thought was to accept the call and place them in a parking lot. Then call the external number, play the sound file and connect the inbound caller to the external number. Is this even possible and if so, is this the best approach? Thank you in advance. -- _ -- 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] white noise on conference
I am trying to track down a white noise problem we are having in our conference rooms. If there are 3 or 4 users in the conference the quality is good. After we get more users in the conference we develop a white noise that gets louder as more users come online. I have tried both meetme and confbridge. I am running 1.8.16.0 compiled from source. Can anyone provide some insight on where to look or anything to tweak to resolve this?-- _ -- 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] confbridge command not found
Currently running version 1.8.16.0 and trying to manage confbridge rooms and users. When I try to use the confbridge cli command I get a command not found error. CLI confbridge No such command 'confbridge' (type 'core show help confbridge' for other possible commands) I've tried googling this but did not get anywhere. How can I enable the confbridge commands? Thanks!-- _ -- 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] Screening Mode Ghost
Hi, It seems there is random behavior that causes screening mode to be activated when a user calls and the line answered and then forwarded using a dial command such as: EXEC Dial SIP/13365551212@8x8|60SIP/13365541212@8x8 |60SIP/13365531212@8x8 |60|dgF(callFlo-in^3^1)M(record^39ff6274-c0f0-453d-aa05-402a7bd6d567^) -- AGI Script Executing Application: (Dial) Options: (SIP/13365551212@8x8|60SIP/13365541212@8x8|60SIP/13365531212@8x8 |60|dgF(callFlo-in^3^1)M(record^39ff65-402a7bd6d567^)) -- Privacy DB is '+18665551212', clid is '16095551212' Any ideas. How can I disable screening? Why is it firing? I saw a similar post from 2007 where the person had the same issue. http://forums.digium.com/viewtopic.php?p=60477sid=caa115851aab005f6e56a218a81618b9 Any help anyone can provide would be greatly appreciated. Gary -- _ -- 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 1.6 Questions
I have a couple of questions about asterisk 1.6: Can you change codecs mid-call upon re-invite? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Thanks in advance for any insight. Gary -- _ -- 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 1.6 Questions
Can you answer both? Can Asterisk be configured to _initiate_ such a change at some point, mid-call? and Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? On Tue, May 3, 2011 at 12:56 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/03/2011 12:43 PM, Gary Graves wrote: Can you change codecs mid-call upon re-invite? Do you mean: can Asterisk be configured to _initiate_ such a change at some point, mid-call? Or do you mean: Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Doesn't seem to. Looking at chan_sip.c in 1.6.2.13, there is no call to add_sdp() that is not made either in the context of 1) an initial INVITE request or 2) a re-INVITE or 3) the construction of a response. Nothing in the case of the production of an end-to-end ACK. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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
[asterisk-users] TDM800P not detecting answer fast enough
Hi All, I have a server running Fedora 14, kernel 2.6.31.14, Asterisk 1.6.2.17.2 and Dahdi 2.4.1. I have the wctdm24xxp+ loaded with a Wildcard TDM800P with 8 FXO ports When a call is placed extension to extension, there is no problem... When an extension is used to dial out, the called number answers, but the server only detects the answer about 3 seconds later .. The outgoing line is TDM on port 2 of the TDM800P. The same hardware was running Asterisk 1.4 recently and we didn't have this problem. Where should I look? Thanks Gary B -- _ -- 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] Music on Hold not working?
Well... Looks like he's trying to use a streaming MOH solution like an online radio station or something, so the files are irrelevant. Too bad the original post didn't specify that. I still think there is a different source selected for the call queue than for the rest of the system. Sorry for the top post... Blackberry won't do it any other way. Sent from my BlackBerry® smartphone -Original Message- From: Doug Lytle supp...@drdos.info Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 16 Jan 2011 10:50:48 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? Tzafrir Cohen wrote: Is that really an issue? open() and all others would normally just reduce '//' to a '/'. That, I really wouldn't know. I'm not a programmer. I noted the differences between mine and his. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Music on Hold not working?
On Sat, Jan 15, 2011 at 7:20 AM, James Miller paramedi...@gmail.com wrote: I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. the middle, and still can not get MOH to work. Did you create /var/lib/asterisk/mohmp3/stream/stream.mp3? Did you Google it and try the solution here: http://nerdvittles.com/index.php?p=92 ? -- _ -- 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] Music on Hold not working?
On Sun, Jan 16, 2011 at 11:41 AM, Warren Selby wcse...@selbytech.com wrote: MOH plays the default class unless specified by a channel variable to play a different one. In queues.conf you can specify the MOH class on a queue by queue basis, but that's the hold music for someone waiting to be answered. Once an agent answers, if they put someone on hold they'll be put into the default MOH class unless a channel variable is specified beforehand. Thanks, --Warren Selby, dCAP On Jan 16, 2011, at 11:55 AM, James M Miller paramedi...@gmail.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, 16 January, 2011 12:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 09:20:10AM -0500, James Miller wrote: I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. Well, why should it play something different? How have you configured it to play something different? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 [James M Miller] Well one would think that if you configure the Music on Hold feature by setting streams for it to pull from, it should play it no matter how the phone is dialed. Meaning if I dial another extension on the network, I should hear the MOH since I have it programmed with streams. However what is occurring is it is only playing when you are placed into a queue. Once someone picks up the line, it starts playing the default again if that person places the person on hold. One would think that it would play MOH no matter what if you have the streams programmed and override the defaults, at least that's what I'd like for it to do. Regards, 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 -- _ -- 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 Yep. musinonhold.conf has not had the default changed to streaming. -- _ -- 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] Music on Hold not working?
I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? -- _ -- 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] Music on Hold not working?
On Sat, Jan 15, 2011 at 9:36 AM, James Miller paramedi...@gmail.com wrote: Forgive me, but how do I do moh show files? Basically what is occurring is: If you enter a queue and are waiting to be answered, you will hear the streaming MOH If you call another extension on the system, you will only hear the default MOH. I want it to stream MOH for everything. Hopefully that makes sense. Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gary Allen *Sent:* Saturday, January 15, 2011 11:33 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Music on Hold not working? I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? Go into Asterisk CLI (asterisk -r) and issue the command moh show files. I don't see how you can have different MOH in a queue vs. being on hold unless you have specified a specific MOH group for your call queues. -- _ -- 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] Why are 4 ports used for a single call?
RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- 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
[asterisk-users] SetVar Warning
I had lines 3 and 4 and added line 1 and 2 to extensions.conf
exten = 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,2,Monitor(wav,${CALLFILENAME},m)
exten = 106,3,hint,SIP/106
exten = 106,4,Macro(stdexten,106,${HINT})
I received this warning:
WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for
extension (voicemenu-custom-4, 106, 1)
I'm running Asterisk/1.4.22.
Does anyone have any idea what I need to do to either make SetVar work or
replace it
with something else?
Thanks you,
Gary
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[asterisk-users] Call hung up?
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT})
When I called x106 this was logged:
-- Executing [106@voicemenu-custom-4:1] Set(DAHDI/7-1,
CALLFILENAME=_xxx) in new stack
-- Executing [106@voicemenu-custom-4:2] Monitor(DAHDI/7-1, wav|_xxx-xxx-
|m) in new stack
== Auto fallthrough, channel 'DAHDI/7-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/7-1'
When I don't have the first two lines this is in the log:
-- Executing [106@voicemenu-custom-4:1] Macro(DAHDI/7-1,
stdexten|106|SIP/106) in new stack
-- Executing [s@macro-stdexten:1] Set(DAHDI/7-1, __DYNAMIC_FEATURES=)
in new stack
-- Executing [s@macro-stdexten:2] GotoIf(DAHDI/7-1, 0?5:3) in new stack
-- Goto (macro-stdexten,s,3)
-- Executing [s@macro-stdexten:3] Dial(DAHDI/7-1, SIP/106|20|) in new
stack
What did I do wrong in adding the first two lines?
Thank you,
Gary
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Re: [asterisk-users] Using SIP stack within Asterisk to rebootphones - Possible?
What type of phones? Easy to do with Polycom and several others from Asterisk CLI. Sent from my BlackBerry® smartphone -Original Message- From: Nikhil d.nik...@cem-solutions.net Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 28 Dec 2010 08:42:22 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible? -- _ -- 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
[asterisk-users] How to block everyone outside of our lan
I'd like to find out how to block everyone outside of the our LAN. The following phone call got through: 1. accountcode: Blank 2. src: Caller*ID number Blank 3. dst: Destination extension 901185294464086 4. dcontext: Destination context DLPN_DialPlan1 5. clid: Caller*ID with text Blank 6. channel: Channel used SIP/xxx-088c48d8 7. dstchannel: Destination channel DAHDI/1-1 8. lastapp: Last application if appropriate Dial 9. lastdata: Last application data (arguments) Dahdi/g1/01185294464086 10. start: Start of call 2010-12-16 04:49:28 11. answer: Answer of call 2010-12-16 04:49:32 12. end: End of call 2010-12-16 04:49:52 13. duration: Total time in system, 24seconds 14. billsec: Total time call is up, 20seconds 15. disposition: What happened to the call: ANSWERED 16. amaflags: What flags to use: DOCUMENTATION In Sip.conf I have: deny=0.0.0.0/0.0.0.0 permit=192.168.1.201/255.255.255.255 All the other local phones here snip One WanIP address Thank you, Gary Kuznitz -- _ -- 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] Configuring Softphone
Thank you for the reply. On 10 Dec 2010 at 8:53, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com) commented about Re: [asterisk-users] Configuring Softphone: Hi Gary, I not using anything to create my dialplan. I'm trying to add a softphone to a dialplan that was created a couple years ago by someone that knew what they were doing. Everything else in the dialplan works. As you can see I don't understand how to create a dialplan and I'm seeing from doing a lot of reading on google that everyone is having a hard time figuring out the dialplan that works with softphones. The part I There is no secret in a dialplan for softphones. In fact Asterisk doesn't care if the SIP-device is a softphone, a hard-phone or even another Asterisk box. Perhaps you are over-complicating the issue? If you have a working dialplan for other phones then why are you trying to set it up differently? Have you tried just using the same settings as a working phone? That is a great suggestion. Yes I did try that. I might be having router issues with a SonicWall. I'm working with a port sniffer now to try to figure it out. When I'm done with making sure the router is forwarding everything correctly I'll try that again. Thank you, Gary Best regards, Jeroen Eeuwes -- _ -- 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] Configuring Softphone
On 10 Dec 2010 at 9:31, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com)
commented about Re: [asterisk-users] Configuring Softphone:
Hi Gary,
That is a great suggestion. Yes I did try that. I might be having router
issues with a
SonicWall. I'm working with a port sniffer now to try to figure it out.
When I'm
done with making sure the router is forwarding everything correctly I'll
try that
again.
If a router is blocking stuff it is bound not to work. Something else
you could try is to configure a softphone on a PC on the same LAN as
the Asterisk box. That way you are by-passing any router issues.
That's a great idea. Even though it's an hour drive for me I might try that
just to
prove it's defiantly not a router issue.
I believe I have proven the router is forwarding just fine now. I have put
back in the
same configuration we use for in house phones.
[gary-incomming]
exten = 120,hint,SIP/120
exten = 120,1,Macro(stdexten,120,${HINT})
When I make a call from the softphone it
1. Shows it registered.
2. Initiated sip call to: the correct phone number
3. Says call answered
4. A few seconds later the phone rings.
5. I answer it.
6. A few seconds later the phone call disconnects from the called phone.
7. The phone call doesn't disconnect from the softphone. I have to disconnect
it
manually.
8. It says Call has disconnected.
9. It says Overall Call Jitter = 0.98 ms
SIP Debug
--- SIP read from SoftPhoneIP:5060 ---
INVITE sip:91phone#cal...@asteriskip SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK103080
To: sip:91phone#cal...@asteriskip
From: gary sip:1...@asteriskip;tag=8826
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 880 INVITE
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Contact: sip:1...@softphoneip:5060
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 380
v=0
o=NCHSoftware-Talk 1291970611 1291970614 IN IP4 SoftPhoneIP
s=Express Talk Call
c=IN IP4 SoftPhoneIP
t=0 0
m=audio 8000 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=local:192.168.168.7 8000
a=domain:SoftPhoneIP
-
--- (13 headers 16 lines) ---
Sending to SoftPhoneIP : 5060 (NAT)
Using INVITE request as basis request - 1291970614-3080-gar...@softphoneip
UbuntuAsterisk*CLI
--- Reliably Transmitting (no NAT) to SoftPhoneIP:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK103080;received=SoftPhoneIP;rport=5060
From: gary sip:1...@asteriskip;tag=8826
To: sip:91phone#cal...@asteriskip;tag=as361b6138
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 880 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0486b332
Content-Length: 0
Scheduling destruction of SIP dialog '1291970614-3080-gar...@softphoneip' in
32000 ms (Method: INVITE)
Found user '120'
--- SIP read from SoftPhoneIP:5060 ---
ACK sip:91phone#cal...@asteriskip SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK103080
To: sip:91phone#cal...@asteriskip;tag=as361b6138
From: gary sip:1...@asteriskip;tag=8826
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 880 ACK
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Content-Length: 0
-
--- (9 headers 0 lines) ---
UbuntuAsterisk*CLI
--- SIP read from SoftPhoneIP:5060 ---
INVITE sip:91phone#cal...@asteriskip SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK113080
To: sip:91phone#cal...@asteriskip
From: gary sip:1...@asteriskip;tag=8826
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 881 INVITE
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Contact: sip:1...@softphoneip:5060
Proxy-Authorization: Digest
username=120,realm=asterisk,nonce=0486b332,uri=sip:91phone#cal...@asteris
kIP,response=fba7a6cc66cf0238dfcc486a5c4f6c73,opaque=,algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 380
v=0
o=NCHSoftware-Talk 1291970611 1291970614 IN IP4 SoftPhoneIP
s=Express Talk Call
c=IN IP4 SoftPhoneIP
t=0 0
m=audio 8000 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=local:192.168.168.7 8000
a=domain:SoftPhoneIP
-
--- (14 headers 16 lines) ---
Sending to SoftPhoneIP : 5060 (NAT)
Using INVITE request as basis request - 1291970614-3080-gar...@softphoneip
Found user '120'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 3
Found RTP audio format 13
Found RTP audio format 101
Peer audio RTP is at port
Re: [asterisk-users] Asterisk SIP attacks and sshguard
I'm not sure if this is the log entry you are looking for. I had many of these last night. [Dec 9 06:47:51] NOTICE[5630]: chan_sip.c:15593 handle_request_register: Registration from '106 sip:1...@mywanaddress' failed for '121.11.158.174' - Wrong password If you need more information from this Asterisk box let me know. I need to find a way to block these also. Gary On 9 Dec 2010 at 7:57, Joe (Joe Greco asterisk-users@lists.digium.com) commented about [asterisk-users] Asterisk SIP attac: Hello, We had been seeing SIP-guessing attacks on our Asterisk server here. While it wasn't that hard to write a once-a-minute cron job to spank the lusers, that runs once a minute and creates little spikes in the usage and I/O graphs, and is slower to respond than I'd really prefer. I felt that it'd be much cooler to get something more comprehensive put together. We don't use fail2ban because I don't like having to install python. sshguard is a high-performance compiled C application that can run off a log file or a pipe from syslogd to sshguard, meaning that it can respond a lot more quickly than once a minute, and works with very modest overhead on the host system. It also has features such as touchiness, so that it can get tougher on a miscreant as time goes on; my own shell script is naive in that once it passes a threshold, there's just a permanent rule generated. This worries me if I ever have a situation where a legitimate remote client gets messed up and tries the wrong password or something like that; sshguard does a much nicer job in this regard. In any case, my initial attempts to create rules for sshguard didn't work right, quite possibly because I don't often work in LEX/YACC. I submitted a request to the sshguard guys suggesting new rules. http://www.sshguard.net/support/submission/detail/49ce7182028d8b6f3e3d/ and on their mailing list, a little more: http://sourceforge.net/mailarchive/forum.php?thread_name=F4E10075-5D93-43B4-B73A-1FD217BE130D%40sshguard.netforum_name=sshguard-users In particular, they're looking for log examples of some of those messages, but I have no idea how to generate the conditions that would cause these messages. I'm also not sure if there's a way to disable color codes in the Asterisk log files; we log indirectly via BSD's logger # asterisk -vvv 21 | logger -t asterisk so it may be thinking that the console is color-capable. We use this method because this forces them through the syslog mechanism; we need that for centralized logging, and it's handy for things like sshguard too. Specifically looking for examples of (or how to generate) 1).*No registration for peer '.*' (from HOST) 2).*Host HOST failed MD5 authentication for '.*' (.*) 3).*Failed to authenticate user .*@HOST.* If anyone who is more familiar with the attacks or how to generate these messages would give me some assistance, or chime in on the sshguard-users list, that'd be most appreciated. Thanks. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- 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
[asterisk-users] (Fwd) Re: Configuring Softphone
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
Sent: Wednesday, December 08, 2010 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Configuring Softphone
The phone is finally registering. That's great.
I'm trying to understand what all these lines in Extensions.conf are
defining.
I can't make heads or tails them. I have been reading the manual
AsteriskManualTheFutureOfTelephony2ndEdition.
I'm currently getting an error when placing a call on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to
extension '91AreaCodePhone#' rejected because extension not found.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = password
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
Frustrated,
Gary
Without any other comment, you need
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
in the gary-incomming context.
As defined now, Gary can
#1 answer a call
#2 call IAX/gogh using 1001
I entered the exten line you suggested:
[gary-incomming]
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
I removed all other lines in [gary-incomming]
When I place a call I get on the cmd line:
-- Executing [916618579...@gary-incomming:1] Dial(SIP/Gary-08941b20,
DAHDI/g1/916618579191) in new stack
-- Called g1/916618579191
-- DAHDI/1-1 answered SIP/Gary-08941b20
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 1291829914-5076-gar...@192.168.168.7 for seqno 669
(Critical Response) -- See doc/sip-retransmit.txt.
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1980 retrans_pkt: Hanging up call
1291829914-5076-gar...@192.168.168.7 - no reply to our critical packet (see
doc/sip-retransmit.txt).
-- Hungup 'DAHDI/1-1'
== Spawn extension (gary-incomming, 916618579191, 1) exited non-zero on
'SIP/Gary-08941b20'
Do you have any ideas? Would you like to see what is in extensions.conf for a
local
extension?
Thank you,
Gary
--- End of forwarded message ---
WPM$44FF.PM$
Description: Mail message body
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Re: [asterisk-users] (Fwd) Re: Configuring Softphone
On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz docf...@theoffice.la) commented
about
[asterisk-users] (Fwd) Re: Configuring Softphone:
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
Sent: Wednesday, December 08, 2010 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Configuring Softphone
The phone is finally registering. That's great.
I'm trying to understand what all these lines in Extensions.conf are
defining.
I can't make heads or tails them. I have been reading the manual
AsteriskManualTheFutureOfTelephony2ndEdition.
I'm currently getting an error when placing a call on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to
extension '91AreaCodePhone#' rejected because extension not found.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = password
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
Frustrated,
Gary
Without any other comment, you need
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
in the gary-incomming context.
As defined now, Gary can
#1 answer a call
#2 call IAX/gogh using 1001
I entered the exten line you suggested:
[gary-incomming]
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
I removed all other lines in [gary-incomming]
When I place a call I get on the cmd line:
-- Executing [916618579...@gary-incomming:1] Dial(SIP/Gary-08941b20,
DAHDI/g1/916618579191) in new stack
-- Called g1/916618579191
-- DAHDI/1-1 answered SIP/Gary-08941b20
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 1291829914-5076-gar...@192.168.168.7 for seqno 669
(Critical Response) -- See doc/sip-retransmit.txt.
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1980 retrans_pkt: Hanging up call
1291829914-5076-gar...@192.168.168.7 - no reply to our critical packet (see
doc/sip-retransmit.txt).
-- Hungup 'DAHDI/1-1'
== Spawn extension (gary-incomming, 916618579191, 1) exited non-zero on
'SIP/Gary-08941b20'
Do you have any ideas? Would you like to see what is in extensions.conf for
a local
extension?
Thank you,
Gary
I'm getting closer. Express Talk is now making the call.
I'm getting an error on the cmd line:
-- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro(SIP/120-
b6003810, trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|) in
new
stack
-- Executing [...@macro-trunkdial-failover-0.3:1]
GotoIf(SIP/120-b6003810, 0?1-
fmsetcid|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:2]
GotoIf(SIP/120-b6003810, 0?1-
setgbobname|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:3] Set(SIP/120-b6003810,
CALLERID(num)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:4]
GotoIf(SIP/120-b6003810, 0?1-
dial|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:5] Set(SIP/120-b6003810,
CALLERID(all)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:6] Goto(SIP/120-b6003810,
1-
dial|1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-d...@macro-trunkdial-failover-0.3:1]
Dial(SIP/120-b6003810,
Dahdi/g1/1MyAreaCodePhone#) in new stack
-- Called g1/1MyAreaCodePhone#
-- DAHDI/1-1 answered SIP/120-b6003810
-- Hungup 'DAHDI/1-1'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero
on
'SIP/120-b6003810' in macro 'trunkdial-failover-0.3'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero
on
'SIP/120-b6003810'
[Dec 9 18:40:39] WARNING[5806]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 1291829922-5076-gar...@192.168.168.7 for seqno 287
(Critical Response) -- See doc/sip-retransmit.txt.
I don't know if this has anything to do with Express Talk using Local RTP ports
to
listen 8000-8020 and Asterisk using 1 and up. I
Re: [asterisk-users] Configuring Softphone
Thanks for the reply.
On 9 Dec 2010 at 20:56, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] (Fwd) Re: Configuring Softp:
On Thu, 9 Dec 2010, Gary Kuznitz wrote:
I'm getting closer. Express Talk is now making the call.
I'm getting an error on the cmd line:
-- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro(SIP/120-
b6003810, trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|)
in new
stack
-- Executing [...@macro-trunkdial-failover-0.3:1]
GotoIf(SIP/120-b6003810, 0?1-
fmsetcid|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:2]
GotoIf(SIP/120-b6003810, 0?1-
setgbobname|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:3] Set(SIP/120-b6003810,
CALLERID(num)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:4]
GotoIf(SIP/120-b6003810, 0?1-
dial|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:5] Set(SIP/120-b6003810,
CALLERID(all)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:6]
Goto(SIP/120-b6003810, 1-
dial|1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-d...@macro-trunkdial-failover-0.3:1]
Dial(SIP/120-b6003810,
Dahdi/g1/1MyAreaCodePhone#) in new stack
-- Called g1/1MyAreaCodePhone#
-- DAHDI/1-1 answered SIP/120-b6003810
-- Hungup 'DAHDI/1-1'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited
non-zero on
'SIP/120-b6003810' in macro 'trunkdial-failover-0.3'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited
non-zero on
'SIP/120-b6003810'
[Dec 9 18:40:39] WARNING[5806]: chan_sip.c:1958 retrans_pkt: Maximum
retries
exceeded on transmission 1291829922-5076-gar...@192.168.168.7 for seqno 287
(Critical Response) -- See doc/sip-retransmit.txt.
I currently have in extensions.conf:
[gary-incomming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERIDNAME})
exten = s,n,Wait(4)
exten = s,n,Playback(tt-weasels)
exten = s,n,Voicemail(11...@vm-test)
exten = s,n,Wait(2)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Wait(2)
exten = s,n,HandUp()
exten = 120,1,Dial(SIP/gary)
exten = gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
Does it seem odd that your console output does not match your dialplan?
I would suggest discarding PIAF or Elastix or whatever created your
dialplan and start from scratch.
I not using anything to create my dialplan. I'm trying to add a softphone to a
dialplan
that was created a couple years ago by someone that knew what they were doing.
Everything else in the dialplan works. As you can see I don't understand how
to
create a dialplan and I'm seeing from doing a lot of reading on google that
everyone is
having a hard time figuring out the dialplan that works with softphones. The
part I
don't understand is why I'm not getting better answers on this list. I know
there are
lots of experts on this list. I'd be happy to hear from someone that gives me
a
private reply that says something like, I'd be happy to help you resolve your
issue if
you are willing to pay me for my time. I don't know what other secrete there
may be
to get help to resolve this issue.
Once you master the concepts and interaction between sip.conf and
extensions.conf you will be in a better place to evaluate the merits of
using a GUI to create your dialplan or continue growing your own.
I'm not using a GUI. It would probably do a much better job than I am. The
entries
I am trying are all found on Google. I'm amazed with all the experts in the
world that
there aren't lots of examples that work. With my trial and error I'm not
having a lot
of luck. Either finding examples that work or finding rules to create a
dialplan.
Thanks for your input,
Gary
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Audio ports
I see in sip debug it says Audio is at port 10342 Express Talk suggests Audio at UDP 8000-8020 I've tried changing Express Talk to 1 and forwarded 1-10400. Is there a possibility Express Talk won't work in the 1 range? Is it possible to limit Asterisk to 8000-8020? Thank you, Gary -- _ -- 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] Audio ports
On 9 Dec 2010 at 22:32, Gary (Gary Kuznitz docf...@theoffice.la) commented about [asterisk-users] Audio ports: I see in sip debug it says Audio is at port 10342 Express Talk suggests Audio at UDP 8000-8020 I've tried changing Express Talk to 1 and forwarded 1-10400. Is there a possibility Express Talk won't work in the 1 range? Is it possible to limit Asterisk to 8000-8020? I see in rpt.conf rtpstart = 8000 rtpend = 8020 Is Audio port 10342 in sip debug not related to rtp ports? It sounds like Express Talk should be configured for 8000-8020 Thanks, Gary Thank you, Gary -- _ -- 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
[asterisk-users] Configuring Softphone
The phone is finally registering. That's great.
I'm trying to understand what all these lines in Extensions.conf are defining.
I can't make heads or tails them. I have been reading the manual
AsteriskManualTheFutureOfTelephony2ndEdition.
I'm currently getting an error when placing a call on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to
extension '91AreaCodePhone#' rejected because extension not found.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = password
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
Frustrated,
Gary
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[asterisk-users] Configuring Softphone
Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Softphone
I have no idea the correct way to configure this software phone.
It's called Express Talk
The Asterisk box is at IP = WanLocation
Software phone is at IP = WanSoftware
They are not on the same LAN.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = 5351
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
When I reload the dialplan I get an error from Asterisk saying:
[Dec 7 22:01:48] NOTICE[5630]: chan_sip.c:15593 handle_request_register:
Registration from 'sip:g...@wanlocation' failed for 'WanSoftware' - No
matching
peer found
The Softphone SipTrace log says:
17:25:35 UDP Packet Received from WanLocation:5060
SIP/2.0 404 Not found
Via: SIP/2.0/UDP
192.168.168.7:5060;branch=z9hG4bK03856;received=WanSoftware;rport=16699
From: sip:g...@wanlocation;tag=1424
To: sip:g...@wanlocation;tag=as214040c6
Call-ID: 1291771532-3856-gar...@localip
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Any ideas on how to configure it better are welcome.
Thank you,
Gary
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[asterisk-users] Asterisk ports
Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary -- _ -- 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 ports
On 2 Dec 2010 at 15:11, Danny (Danny Nicholas da...@debsinc.com) commented about RE: [asterisk-users] Asterisk ports: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Thursday, December 02, 2010 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk ports Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary Try netstat -anp|grep ast This will show you all of the ports and addresses asterisk is using (if it is running). Thank you for the reply. Does this look correct? I don't know what port the sip phones are supposed to be communicating on. tcp0 0 0.0.0.0:50380.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:80880.0.0.0:* LISTEN 5382/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 5382/asterisk unix 2 [ ACC ] STREAM LISTENING 180595382/asterisk /var/run/asterisk.ctl unix 2 [ ACC ] STREAM LISTENING 205225768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 2 [ ] DGRAM325885382/asterisk unix 3 [ ] STREAM CONNECTED 207295768/fast-user-swit unix 3 [ ] STREAM CONNECTED 207285768/fast-user-swit unix 3 [ ] STREAM CONNECTED 207275768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205395768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205265768/fast-user-swit unix 3 [ ] STREAM CONNECTED 205255768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205205768/fast-user-swit unix 3 [ ] STREAM CONNECTED 205085768/fast-user-swit Thank you, Gary -- _ -- 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 ports
Thank you for the reply. On 2 Dec 2010 at 16:23, Jeff (Jeff LaCoursiere j...@sunfone.com) commented about Re: [asterisk-users] Asterisk ports: On Thu, 2 Dec 2010, Gary Kuznitz wrote: Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary You probably see it as: udp0 0 *:sip *:* I don't see this. That could certainly be why the phones are connecting. Why wouldn't that port be listening? Thank you, Gary 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] Asterisk ports
Thanks for the reply. On 2 Dec 2010 at 14:11, Steve (Steve Edwards asterisk-users@lists.digium.com) commented about Re: [asterisk-users] Asterisk ports: On Behalf Of Gary Kuznitz Shouldn't Asterisk be listening on UDP port 5060? Yes. Unless configured otherwise, that's the SIP port. It's set in sip.conf. What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 In sip.conf bindport = 5060 'Sip show settings' doesn't work in 1.4.22 I have re-booted this machine. What else could I look for as to why UDP 5060 isn't listening? Thanks, Gary unless you know what you're doing. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 ports
Thanks for the reply. On 2 Dec 2010 at 15:57, Steve (Steve Edwards asterisk-users@lists.digium.com) commented about Re: [asterisk-users] Asterisk ports: On Thu, 2 Dec 2010, Steve Edwards wrote: What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 On Thu, 2 Dec 2010, Gary Kuznitz wrote: In sip.conf bindport = 5060 'Sip show settings' doesn't work in 1.4.22 I don't have access to a '1.4' instance right now, but 'sip show settings' works in 1.2 and 1.6 so I'm guessing it should work in 1.4 as well. You may have an error that prevents the SIP channel driver from loading. What do you get with 'unload chan_sip.so' followed by 'load chan_sip.so'? You get extra points today. I think you found where the problem is. It found /etc/asterisk/sip.conf Warning parse error: No category context for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf. This is what is in sip.conf. [authentication] [general] context = default allowoverlap = no bindport = 5060 bindaddr = 0.0.0.0 srvlookup = yes limitonpeers = yes allow = all allowguest=yes What doesn't it like? Thanks, Gary -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 ports
Thank you very much for the reply.
On 2 Dec 2010 at 17:06, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] Asterisk ports:
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
You get extra points today. I think you found where the problem is. It
found /etc/asterisk/sip.conf Warning parse error: No category context
for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf.
This is what is in sip.conf.
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=yes
Running out of clues here :)
I can load the above fine in my 1.2 instance. Any chance the file was
edited on Windows and needs to be 'unixfied?'
What does 'hexdump -C sip.conf' look like?
Does commenting (';') out line 1 change anything?
This fixed the problem. There was some garbage in line 1.
You are great. Thank you very much.
Gary
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Trying to configure a SIP software phone
I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . I'd like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files at http://pastebin.com/ajp62wqF and see what I did wrong? Thank you, Gary -- _ -- 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] Trying to configure a SIP software phone
Thank you for the reply. Comments below... On 30 Nov 2010 at 20:21, Warren (Warren Selby wcse...@selbytech.com) commented about Re: [asterisk-users] Trying to configure a SIP so: On Tue, Nov 30, 2010 at 8:04 PM, Gary Kuznitz docf...@theoffice.la wrote: I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . I'd like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files at http://pastebin.com/ajp62wqF and see what I did wrong? Thank you, Gary That pastebin shows a lot of things that seem wrong. From the error message at the bottom of the pastebin, it looks like you've configured your softphone to register using the username 120, however you've configured your sip peer in sip.conf as Gary for the username. You'll need to match those up for starters. The extensions.conf snippet has a lot of odd logic to it as well, but before we begin to tackle that, let's get the phone registered first. I changed the user in the softphone to Gary. This is the new log. 20:07:48 SIP Public IP is: 75.xxx.xxx.xxx:4582 20:07:48 SIP Number: g...@75.xxx.xxx.xxx:4582 20:07:48 Attempting to register sip:g...@208.xxx.xxx.xxx 20:08:30 Server 208.xxx.xxx.xxx did not respond to register (user sip:g...@208.xxx.xxx.xxx) 20:08:30 Check server details for that line -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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] Someone has hacked into our system
On 23 Nov 2010 at 16:54, Joseph (Joseph syscon...@gmail.com) commented about Re: [asterisk-users] Someone has hacked into our : On 11/23/10 14:18, Gary Kuznitz wrote: Thank you for the reply... Comments below... On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk- us...@lists.digium.com) commented about Re: [asterisk-users] Someone has hacked into our : On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote: I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. allowguest=yes in sip.conf, with a context= in the [general] section that is permitted to make outbound calls? I'm trying to understand exactly what this means. I found a sip.conf in /etc/asterisk I have a [general] section. I don't have allowguest=yes. Is that good or am I supposed to have it? Look for allowguest default is yes I change it to allowguest=no In addition you might want to restrict some countries in your dial-plan, here is my list: This would be great. Can I put this anyplace in extensions.conf? Or does it need to go after [DLPN_DialPlanl] ? Thanks, Gary Kuznitz [blocked-numbers] ;block bahamas, etc exten = _91900.,1,congestion; N11 exten = _91XXX976.,1,congestion ; N11 exten = _91XXX555.,1,congestion ; N11 exten = _91X11.,1,congestion; N11 exten = _91867.,1,congestion; Yukon (sorry mike) ;exten = _1NPA Country exten = _91232.,1,congestion; Sierra Leone exten = _91242.,1,congestion; BAHAMAS exten = _91246.,1,congestion; BARBADOS exten = _91264.,1,congestion; ANGUILLA exten = _91268.,1,congestion; ANTIGUA/BARBUDA exten = _91284.,1,congestion; BRITISH VIRGIN ISLANDS exten = _91345.,1,congestion; CAYMAN ISLANDS exten = _91441.,1,congestion; BERMUDA exten = _91473.,1,congestion; GRENADA exten = _91649.,1,congestion; TURKS CAICOS ISLANDS exten = _91664.,1,congestion; MONTSERRAT exten = _91758.,1,congestion; ST. LUCIA exten = _91767.,1,congestion; DOMINICA exten = _91784.,1,congestion; ST. VINCENT GRENADINES exten = _91809.,1,congestion; DOMINICAN REPUBLIC exten = _91829.,1,congestion; DOMINICAN REPUBLIC exten = _91868.,1,congestion; TRINIDAD AND TOBAGO exten = _91869.,1,congestion; ST. KITTS AND NEVIS exten = _91876.,1,congestion; JAMAICA -- Joseph -- _ -- 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] Someone has hacked into our system
Thank you for the reply. On 23 Nov 2010 at 18:51, John (John Novack jnov...@stromberg-carlson.org) commented about Re: [asterisk-users] Someone has hacked into our : Gary Kuznitz wrote: Thank you for the reply... Comments below... On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesherasterisk- us...@lists.digium.com) commented about Re: [asterisk-users] Someone has hacked into our : On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote: I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. allowguest=yes in sip.conf, with a context= in the [general] section that is permitted to make outbound calls? I'm trying to understand exactly what this means. I found a sip.conf in /etc/asterisk I have a [general] section. I don't have allowguest=yes. Is that good or am I supposed to have it? I believe what you SHOULD have is; allowguest=no Not sure if that is the default behavior or not If I'm supposed to have it can it go any place in the [general] section? I have in the [general] section a line with: context = default Is this where I would remove default and enter the IP addresses that are allowed to make calls? Your default context in extensions.conf should basiclly lead nowhere. I have mine set up to play an insane laugh then hangup Probably safe to say NEVER use context default for any outbound calling I don't have any context in extensions.conf I do have context = default in sip.conf Should I remove that line? Could you give me an example of what you have in your extensions.conf? Thank you, Gary Kuznitz You should also have, in general: alwaysauthreject=yes This seems pretty effective in stopping some hacking These are simple fixes. I will let others comment on other more detailed firewalling John Novack What would a line with IP address look like? Could you give me an example? If that isn't where the IP address that are allowed supposed to be where would I put them? Thank you, Gary Kuznitz Just a guess, but there have been more than a few such discussions on the list about that configuration, plus a README-SERIOUSLY.bestpractices.txt in the root directory of every Asterisk source tree. You DID read that file, right? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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 -- Dog is my Co-pilot -- _ -- 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] Someone has hacked into our system
Thank you for the reply... Comments below... On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk- us...@lists.digium.com) commented about Re: [asterisk-users] Someone has hacked into our : On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote: I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. allowguest=yes in sip.conf, with a context= in the [general] section that is permitted to make outbound calls? I'm trying to understand exactly what this means. I found a sip.conf in /etc/asterisk I have a [general] section. I don't have allowguest=yes. Is that good or am I supposed to have it? If I'm supposed to have it can it go any place in the [general] section? I have in the [general] section a line with: context = default Is this where I would remove default and enter the IP addresses that are allowed to make calls? What would a line with IP address look like? Could you give me an example? If that isn't where the IP address that are allowed supposed to be where would I put them? Thank you, Gary Kuznitz Just a guess, but there have been more than a few such discussions on the list about that configuration, plus a README-SERIOUSLY.bestpractices.txt in the root directory of every Asterisk source tree. You DID read that file, right? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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 -- _ -- 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] Someone has hacked into our system
Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -- _ -- 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] Someone has hacked into our system
Thank you very much for help in finding the log. I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. Thanks you, Gary Kuznitz On 22 Nov 2010 at 13:11, Danny (Danny Nicholas da...@debsinc.com) commented about RE: [asterisk-users] Someone has hacked into our : From: Gary Kuznitz [mailto:docf...@theoffice.la] Sent: Monday, November 22, 2010 12:20 PM To: Danny Nicholas Subject: Re: [asterisk-users] Someone has hacked into our system Thank you for the quick response. Comments below... I am not familiar with navigating Asterisk. Would you please help me understand how to see the CDR? Thank you, Gary Kuznitz By default, Asterisk keeps the CDR as a flat-file in /var/log/asterisk/cdr-csv/Master.csv which you can open in Excel for easy viewing. If you have a custom cdr (see /etc/asterisk/cdr.conf or /etc/asterisk/cdr_custom.conf for more information), your CDR might be stored in a MYSQL table or some other place.I would start under the assumption that you have the flat file available.Once you have it open, use this link as a guide http://www.voip-info.org/wiki/view/Asterisk+cdr+csv Fields * accountcode: What account number to use: Asterisk billing account, (string, 20 characters) * src: Caller*ID number (string, 80 characters) * dst: Destination extension (string, 80 characters) * dcontext: Destination context (string, 80 characters) * clid: Caller*ID with text (80 characters) * channel: Channel used (80 characters) * dstchannel: Destination channel if appropriate (80 characters) * lastapp: Last application if appropriate (80 characters) * lastdata: Last application data (arguments) (80 characters) * start: Start of call (date/time) * answer: Answer of call (date/time) * end: End of call (date/time) * duration: Total time in system, in seconds (integer) * billsec: Total time call is up, in seconds (integer) * disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY, FAILED * amaflags: What flags to use: see amaflags::DOCUMENTATION, BILL, IGNORE etc, specified on a per channel basis like accountcode. You will want to see if there are any peculiar src fields on your international calls (dst). WPM$68B7.PM$ Description: Mail message body -- _ -- 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 sound between extensions
I have remote access to the server so I checked the canreinvite .. they are all set to no. I can't try the call from here, I will get back to you. Gary Baribault On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote: Do you agree something is blocking the audio in one direction? Can you do a 'rtp debug' and then initiate a SIP call and see if there is two way audio traffic. Also make sure these extensions have 'canreinvite=no'. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 7:02 PM, Gary Baribault g...@baribault.net mailto:g...@baribault.net wrote: As I stated, the incoming calls are on TDM DS0s connected to the Digium card, and the extensions are on the same local network as the Asterisk server. There is currently no NAT anywhere. Gary Baribault On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote: Output of 'iptables -L -n' would also be helpfu... -- _ -- 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] no sound between extensions
I have checked, the users have ulaw, then alaw, the phones are set to 711u then 711a which is the same thing (I think). Gary Baribault On 06/02/2010 08:32 AM, taimur hasan wrote: Also check the codecs as if you are using g729 or g723, there is a chance that they are not available in codecs directory ( /usr/lib/asterisk/modules). *-THQ- !!!ONE* Date: Tue, 1 Jun 2010 19:24:41 -0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] no sound between extensions Do you agree something is blocking the audio in one direction? Can you do a 'rtp debug' and then initiate a SIP call and see if there is two way audio traffic. Also make sure these extensions have 'canreinvite=no'. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 7:02 PM, Gary Baribault g...@baribault.net mailto:g...@baribault.net wrote: As I stated, the incoming calls are on TDM DS0s connected to the Digium card, and the extensions are on the same local network as the Asterisk server. There is currently no NAT anywhere. Gary Baribault On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote: Output of 'iptables -L -n' would also be helpfu... -- _ -- 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 Hotmail: Trusted email with powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- _ -- 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 sound between extensions
I don't know if this makes any difference, I created a lot of this configuration with the Asterisk-GUI (SVN-branch-2.0-r4980) and when I edit the users.conf file, there are two entries 'type = peer' for each extension and they are highlighted in red! Gary Baribault On 06/02/2010 08:32 AM, taimur hasan wrote: Also check the codecs as if you are using g729 or g723, there is a chance that they are not available in codecs directory ( /usr/lib/asterisk/modules). *-THQ- !!!ONE* Date: Tue, 1 Jun 2010 19:24:41 -0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] no sound between extensions Do you agree something is blocking the audio in one direction? Can you do a 'rtp debug' and then initiate a SIP call and see if there is two way audio traffic. Also make sure these extensions have 'canreinvite=no'. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 7:02 PM, Gary Baribault g...@baribault.net mailto:g...@baribault.net wrote: As I stated, the incoming calls are on TDM DS0s connected to the Digium card, and the extensions are on the same local network as the Asterisk server. There is currently no NAT anywhere. Gary Baribault On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote: Output of 'iptables -L -n' would also be helpfu... -- _ -- 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 Hotmail: Trusted email with powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- _ -- 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 sound between extensions
Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP, and are on the same flat non-routed network. There is no NAT involved. If I call from extension 6000 to extension 6001, or vice-versa both are SPA-921s. The 6001 rings, but when the phone is picked up, I have no sound. I have the same problem between any phones in the system, but this is the simplest example. Incoming calls and outgoing calls work fine, sound is correct. Voice mail works fine as well, the IVR works great. Any ideas? Gary Baribault -- _ -- 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 sound between extensions
Incomming calls are on TDM lines connected to the Digium card. Calls between extentions are on the LAN for SIP registered users/ip phones. Gary Baribault On 06/01/2010 03:32 PM, Zeeshan Zakaria wrote: Incoming and outgoing calls are on SIP or on ZAP? Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 3:28 PM, Gary Baribault g...@baribault.net mailto:g...@baribault.net wrote: Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP, and are on the same flat non-routed network. There is no NAT involved. If I call from extension 6000 to extension 6001, or vice-versa both are SPA-921s. The 6001 rings, but when the phone is picked up, I have no sound. I have the same problem between any phones in the system, but this is the simplest example. Incoming calls and outgoing calls work fine, sound is correct. Voice mail works fine as well, the IVR works great. Any ideas? Gary Baribault -- _ -- 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] no sound between extensions
This is done while the calls are active? I just issued the command and got nothing, but there where no active calls. Gary Baribault On 06/01/2010 03:45 PM, Danny Nicholas wrote: My assumption is that inbound/outbound calls are DAHDI and that internal calls are SIP. Can OP post core show channels from working and non-working calls? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Baribault Sent: Tuesday, June 01, 2010 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] no sound between extensions Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP, and are on the same flat non-routed network. There is no NAT involved. If I call from extension 6000 to extension 6001, or vice-versa both are SPA-921s. The 6001 rings, but when the phone is picked up, I have no sound. I have the same problem between any phones in the system, but this is the simplest example. Incoming calls and outgoing calls work fine, sound is correct. Voice mail works fine as well, the IVR works great. Any ideas? Gary Baribault -- _ -- 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 sound between extensions
As I stated, the incoming calls are on TDM DS0s connected to the Digium card, and the extensions are on the same local network as the Asterisk server. There is currently no NAT anywhere. Gary Baribault On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote: Output of 'iptables -L -n' would also be helpful. I am sure its a NAT issue if incoming and ougoing calls are on ZAP channels. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 3:53 PM, Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com wrote: My assumption is that inbound/outbound calls are DAHDI and that internal calls are SIP. Can OP post core show channels from working and non-working calls? -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-bou. mailto:asterisk-users-bou... -- _ -- 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 / Trixbox 2.6 Streaming MOH Problems
I've tried a number of solutions, but I've been unable to get Asterisk working with streaming MOH without running into the buffer issue. I've tried using various combinations madplay, mpg123, mpg321. I've also tried streamplayer by itself, and in combination with play-fifo ( http://www.freeswitch.org/asterisk_stuff/play-fifo.c ) to try and eliminate the issue. For those that are unaware of the problem, what happens when you use a streaming music source with asterisk is you have a process that is running all the time that pipes MOH into stdout, which is then read by asterisk. When a caller is on hold, asterisk starts reading from stdin, and you get your music on hold. When the caller hangs up, asterisk stops reading from stdin (and the pipe becomes blocking), and a buffer is created (I'm not sure where the buffer resides, although I suspect it's probably the system fifo pipe buffer). The problem becomes, when the next caller comes in, and is put on hold, you will hear that buffer (usually about 20-30 seconds), and then it will jump to the current position in the stream, so you hear an ugly jump between the middle of two songs. There was a magic version of mpg123 that was supposed to solve this problem (0.59r, I believe), but I've been unable to get this to work. For those interested, I'm streaming music off of a Barix Instreamer, attached to a satellite radio source (and yes, I'm paying the proper licence fees). The only thing I've found that works so far is a pretty ugly (although ingenious) hack as seen here (http://www.mail-archive.com/asterisk-users@lists.digium.com/msg197299.html), which creates its own host of problems (such as not being able to do a restart when convienent since it generates a call on its own (that's always running), so it's never convienent for asterisk to restart. Also, when I do restart asterisk, I have to restart the call, so I'd prefer having to go this route if at all possible. Another solution would be if asterisk could spawn a new process for every MOH caller. Is this possible? Does anyone have a successful deployment of streaming music on hold that they'd care to share? I'm using Asterisk 1.4 as part of Trixbox 2.6 -- _ -- 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] Off Topic
Please forgive this off-topic post... I've been on this list since 2005 (over 45k messages in my archive) and this is obviously really not something I normally do. If you have a minute and are feeling generous, please visit http://bailout.chipin.com/ and consider helping me out. Sorry if I've offended or wasted your time, but believe me that you don't feel as bad as I do. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Reporting
Hi Folks, sorry for the delay ... I found that the documentation was rather iffy .. I finally found the defines.php in the lib subdirectory and figured out how to give the MySQL port with the host and it all works fine now. Gary Baribault Courriel: g...@baribault.net GPG Key: 0xFA812835 GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835 On 09/11/2009 01:09 AM, Matt Riddell wrote: On 11/09/09 7:11 AM, Gary Baribault wrote: Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page.. http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Has anyone got that it running? My server is OpenSuSE 11.2 with Apache 2 and PHP5, which is probably the problem.. the software probably needs PHP4. Yeah we use it from time to time. What do you mean it wasn't working? Did you get some errors or something? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Reporting
Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page.. http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Has anyone got that it running? My server is OpenSuSE 11.2 with Apache 2 and PHP5, which is probably the problem.. the software probably needs PHP4. Any other solutions? I'm looking for open source, the server is not commercial, and I have very little budget. Thanks Gary B -- Gary Baribault Courriel: g...@baribault.net GPG Key: 0xFA812835 GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] requirecalltoken and Realtime
Hi, I've just had to enable the requirecalltoken=no option in iax.conf for one of my IAX2 trunks, and I don't think it works properly in the realtime version. I've created the requirecalltoken field in my (Postgres via ODBC) database, type text, and have variously tried it with 'yes', 'no' and 'auto' in the field. But the setting never seems to be used and thus calls fail down the trunk. If I try the same thing using iax.conf flat file, the requirecalltoken parameter works fine, so I was wondering if anyone else has seen this and wonder if I've tripped over a bug? All this was tested using 1.6.1 SVN, r216266. Gary H -- Gary Hawkins MBCS gary.hawk...@garyhawkins.me.uk OpenPGP Key ID: 0x9A1037BB Web: http://www.garyhawkins.me.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] requirecalltoken and Realtime
Tilghman Lesher wrote: On Friday 04 September 2009 12:08:26 Gary Hawkins wrote: I've just had to enable the requirecalltoken=no option in iax.conf for one of my IAX2 trunks, and I don't think it works properly in the realtime version. [snip] Please try the attached patch. I've just tried the patch - but it doesn't seem to have made any difference - iax.conf entries still work though exactly as before. Gary H -- Gary Hawkins MBCS gary.hawk...@garyhawkins.me.uk OpenPGP Key ID: 0x9A1037BB Web: http://www.garyhawkins.me.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskGUI Create VoiceMenu SNAFU
Hi All, I'm new to Asterisk, but am a relatively accomplished Linux guy (RH5.1) .. I'm using the Asterisk-GUI to try and create a menu for incoming calls on an Analog Trunk. I have recorded some .WAV files for the menu, but when I try to upload the files, I get an AG101 message. So I copied the files to /var/lib/asterisk/sounds/record .. when I go to the Voice Menu Prompts selection down the left side of the Asterisk-GUI, I see my four files with the options to record again, play and delete. If I then go to the Voice Menu option to configure a Voice Menu, and click on the Create New VoiceMenu, I get nothing .. nada .. kaput .. how the heck do I create a menu for an incoming call on a Trunk? When I started this project I knew it would be fun .. I would learn a lot! The problem is that one of our administrators is absolutely a newbi to Linux, so I have to make this work with the GUI .. any help or suggestions would be appreciated! Thanks Gary B ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Create VoiceMenu SNAFU
Hi All, I'm new to Asterisk, but am a relatively accomplished Linux guy (RH5.1) .. I'm using the Asterisk-GUI to try and create a menu for incoming calls on an Analog Trunk. I have recorded some .WAV files for the menu, but when I try to upload the files, I get an AG101 message. So I copied the files to /var/lib/asterisk/sounds/record .. when I go to the Voice Menu Prompts selection down the left side of the Asterisk-GUI, I see my four files with the options to record again, play and delete. If I then go to the Voice Menu option to configure a Voice Menu, and click on the Create New VoiceMenu, I get nothing .. nada .. kaput .. how the heck do I create a menu for an incoming call on a Trunk? When I started this project I knew it would be fun .. I would learn a lot! The problem is that one of our administrators is absolutely a newbi to Linux, so I have to make this work with the GUI .. any help or suggestions would be appreciated! Thanks Gary B ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-gui 2.0 Asterisk 1.4.26-RC6 Analog trunks
I'm new to asterisk, but have been at Linux since 1997 .. so kind of a newbie .. I have a good buddy who is supposed to be helping me with this darn install, but you all know how that works. I have a new mid-tower, AMD 64 x2, 4 gigs of memory and spoftware mirror. OpenSuSE 11.2 .. as I said Asterisk 1.4.26-rc6 and Asterisk-GUI 2.0 .. I have an 8 port TDM800P card from Digium. basic install is ok. I have the card defined, and everything seems hunky dorie .. but when I try and create an Analog Trunk, I add the trunk, select the first channel and the GUI says I have to reboot the server for it to work. I check hardware and the 8 port card is found .. I go to trunks .. none defined, so I create an analog trunk, select port 1, call it default, and accept all defaults. Hit add, and everything seems ok .. I click apply changes and get a warning that I will have to reboot for changes to take effect. When I reboot the server, the trunk is gone! WTF??? -- Gary Baribault Courriel: g...@baribault.net GPG Key: 0xFA812835 GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835 ___ -- 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
[asterisk-users] About Asterisk 1.6 web GUI
Hi, I had some experience on Asterisk 1.0.7 and 1.2.0. Now, I want to do something on the New Release of Asterisk 1.6.xx. I want to know wheather there are already web GUI for use now in the release. Or still nedd integrate some other third part GUI? Any advice will be appreciated. Thanks ahead, Best Regards, Gary ___ -- 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
[asterisk-users] Remote Connection to Asterisk
Hello all - This is basically an updated re-posting of one I've posted a few days ago. Thanks to the kind help provided but I still can't make it work. But I'm moving a little further down the line (thanks to you folks). Basically, I've got an Asterisk server in a LAB ENVIRONMENT on my home LAN. The server has a Wildcard TDM400 installed but has no POTs lines/phones connected. The LAN is talking to the Internet via a Linksys WRTG-54 router connected to my cable-modem. - On the home LAN, I've also got a few Cisco phones (7940's/7960's one 7905) that work fine talking back and forth with each other via Asterisk. Actually, one of the 7960's is connected to my Broadvoice account so I KNOW sip works over my local LAN connection. My home (real world) IP address is static. On the router, I've turned on DMZ to point it to my Asterisk box's static IP address. I know this is probably overkill, but at this point I'm trying anything. I've also set up port forwarding on the router for port '5060' and all the 'RTP' ports asterisk needs to point to the Asterisk box. I've also set in Asterisk's conf file for 'nat=yes' for all extensions registering with the server. The Problem: Basically - Remote users can register make calls. The phone on the other end rings, but there is no voice traffic. - Just silence. Remote users have been Cisco 79xx phones, xLite softphones (I think that name's correct), and Zoiper (we've tried both IAX and SIP on Zoiper). Any Ideas? - If anybody wants work with me over the phone, I'll be happy to call them. E-mail me: gguthary-at-jtech.net - With my Broadvoice phone, I can call anywhere in 35 countries. I'll set you up with an extension and you can test. Much (VERY MUCH) thanks in advance. This is very frustrating to say the least. Gary G. ___ -- 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
[asterisk-users] Remote connection to an Asterisk server
I've been reading this forum for over the past 4 years and have gained a wealth of knowledge. - Thanks to all! I don't post very often but I've just ran into a problem/condition that I simply can't figure out. - Hopefully some kind soul will help me. I've got an Asterisk server in a lab environment on my home LAN. - The LAN is talking to the Internet via a Linksys WRTG-54 router connected to my cable-modem. - On the home LAN, I've also got a couple of Cisco phones (7940's) and they work fine talking back and forth with each other via Asterisk. In fact, I've also got a Broadvoice account with a 7960 logged in all the time and it works fine. So I know SIP works through my router. On the router, I've turned on DMZ to point to my Asterisk box's static IP address. My home (real world) IP address is static. The Problem: When I grab one of my Cisco 7940's and take it to my office, it does not see or register with my home Asterisk server after I change it's proxy to point to my home IP address. Any Ideas? - Is this a router issue (sure seems like it)? - Much thanks in advance. Gary G. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gosubs broken since r160626 (1.6.0 SVN) ?
Mark Michelson wrote: In a fit of wild curiosity, I decided to double-check to be sure that the problem was an AEL parser issue and not one of my own. I actually discovered a bug introduced by my changes. I have fixed this bug in revision 161494 of the 1.6.0 branch. I suspect this will fix the problem you were seeing, too. I've just tested with this revision and all seems to be well again. Thanks for finding and fixing the bug! Gary H ___ -- 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
[asterisk-users] Gosubs broken since r160626 (1.6.0 SVN) ?
Hi all,
I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs
have stopped working.
This is from the verbose logs:
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/aaisp-3802,
1?5:7) in new stack
-- Goto (incoming-aaisp,0407271,5)
-- Executing [EMAIL PROTECTED]:5] Gosub(IAX2/aaisp-3802,
macro-announcement,s,1(anonymous_call_rejection,22)) in new stack
== Spawn extension (incoming-aaisp, 0407271, 6) exited non-zero on
'IAX2/aaisp-3802'
-- Hungup 'IAX2/aaisp-3802'
This was the original AEL2 code:
0407271 = {
Verbose(We got here);
AGI(caller_id_rewriter/caller_id_rewriter.py);
Set(CALLERID(name)=1 ${CALLERID(name)});
if (${WITHHELD} = yes) {
macro-announcement(anonymous_call_rejection,22);
Hangup(22);
}
Dial(${ALLPHONES},20);
if (${DIALSTATUS} = BUSY) {
VoiceMail(201,b);
}
else
{
VoiceMail(201,u);
}
Hangup(${HANGUPCAUSE});
}
This was working on 1.6.0 SVN before r160626 and I have not changed any
of the code. The Gosubs were generated by the AEL parser. In the AEL2
dialplan I am calling
macro-announcement(anonymous_call_rejection,22);
Has anyone seen similar problems to this?
Thanks
Gary H
--
Gary Hawkins [EMAIL PROTECTED]
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[asterisk-users] Need help for debuging
I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. # thread apply all bt Thread 6 (process 20135): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb7469b80, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb7469b4c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xb22bf9a8, c1=0xa2ae648, config=0xb746a7a0, fo=0xb7469c40, rc=0xb7469c44) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xb22bf9a8, peer=0xa2ae648, config=0xb746a7a0) at res_features.c:1365 #5 0x005a40ba in dial_exec_full (chan=0xb22bf9a8, data=Variable data is not available. ) at app_dial.c:1633 #6 0x005a6a33 in dial_exec (chan=0xfffc, data=0x7fff) at app_dial.c:1680 #7 0x08090bad in pbx_extension_helper (c=0xb22bf9a8, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xb22bf9a8) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 5 (process 11504): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb51e2e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb51e2e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xa2db720, c1=0xa12acf8, config=0xb51e37c0, fo=0xb51e2f50, rc=0xb51e2f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xa2db720, peer=0xa12acf8, config=0xb51e37c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb51e3ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0xa2db720, data=0xb51e8010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0xa2db720, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xa2db720) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 ---Type return to continue, or q return to quit--- #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 4 (process 24033): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb6c56e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb6c56e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0x9e9cb80, c1=0xaf0ce2f8, config=0xb6c577c0, fo=0xb6c56f50, rc=0xb6c56f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0x9e9cb80, peer=0xaf0ce2f8, config=0xb6c577c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb6c57ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0x9e9cb80, data=0xb6c5c010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0x9e9cb80, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0x9e9cb80) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 3 (process 30070): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x00fa9e27 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 #4 0x080668f5 in ast_read (chan=0x2f6a9e) at channel.c:1945 #5 0x08069d98 in ast_channel_bridge (c0=0xb015fea0, c1=0xa0b5c88, config=0xb4e937a0, fo=0xb4e92c40, rc=0xb4e92c44) at channel.c:3399 #6 0x00548f65 in ast_bridge_call (chan=0xb015fea0, peer=0xa0b5c88, config=0xb4e937a0) at res_features.c:1365 #7 0x005a40ba in dial_exec_full (chan=0xb015fea0, data=Variable data is not available. ) at app_dial.c:1633 #8 0x005a6a33 in dial_exec (chan=0xfffc, data=0x348ff4) at app_dial.c:1680 #9 0x08090bad in pbx_extension_helper (c=0xb015fea0, con=Variable con is not available. ) at pbx.c:574 #10 0x08091e86 in __ast_pbx_run (c=0xb015fea0) at pbx.c:2250 #11 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #12 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #13 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 2 (process 21752): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x0028bb61 in strcasecmp () from
Re: [asterisk-users] Need help for debuging
- Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 11:54 AM Subject: Re: [asterisk-users] Need help for debuging On Monday 13 October 2008 10:29:17 gary wrote: I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. The definition of insanity is doing the same thing over and over again, expecting a different outcome. I told you after your previous post how to find the problem. If you aren't willing to follow those instructions, then there is nobody who can help you. -- Tilghman For some reason, I never received your reply nor my original post. That is why I repost again. Can you repost your reply here? gary ___ -- 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
[asterisk-users] Help need for debuging the core file.
I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. # thread apply all bt Thread 6 (process 20135): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb7469b80, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb7469b4c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xb22bf9a8, c1=0xa2ae648, config=0xb746a7a0, fo=0xb7469c40, rc=0xb7469c44) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xb22bf9a8, peer=0xa2ae648, config=0xb746a7a0) at res_features.c:1365 #5 0x005a40ba in dial_exec_full (chan=0xb22bf9a8, data=Variable data is not available. ) at app_dial.c:1633 #6 0x005a6a33 in dial_exec (chan=0xfffc, data=0x7fff) at app_dial.c:1680 #7 0x08090bad in pbx_extension_helper (c=0xb22bf9a8, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xb22bf9a8) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 5 (process 11504): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb51e2e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb51e2e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xa2db720, c1=0xa12acf8, config=0xb51e37c0, fo=0xb51e2f50, rc=0xb51e2f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xa2db720, peer=0xa12acf8, config=0xb51e37c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb51e3ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0xa2db720, data=0xb51e8010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0xa2db720, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xa2db720) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 ---Type return to continue, or q return to quit--- #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 4 (process 24033): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb6c56e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb6c56e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0x9e9cb80, c1=0xaf0ce2f8, config=0xb6c577c0, fo=0xb6c56f50, rc=0xb6c56f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0x9e9cb80, peer=0xaf0ce2f8, config=0xb6c577c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb6c57ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0x9e9cb80, data=0xb6c5c010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0x9e9cb80, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0x9e9cb80) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 3 (process 30070): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x00fa9e27 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 #4 0x080668f5 in ast_read (chan=0x2f6a9e) at channel.c:1945 #5 0x08069d98 in ast_channel_bridge (c0=0xb015fea0, c1=0xa0b5c88, config=0xb4e937a0, fo=0xb4e92c40, rc=0xb4e92c44) at channel.c:3399 #6 0x00548f65 in ast_bridge_call (chan=0xb015fea0, peer=0xa0b5c88, config=0xb4e937a0) at res_features.c:1365 #7 0x005a40ba in dial_exec_full (chan=0xb015fea0, data=Variable data is not available. ) at app_dial.c:1633 #8 0x005a6a33 in dial_exec (chan=0xfffc, data=0x348ff4) at app_dial.c:1680 #9 0x08090bad in pbx_extension_helper (c=0xb015fea0, con=Variable con is not available. ) at pbx.c:574 #10 0x08091e86 in __ast_pbx_run (c=0xb015fea0) at pbx.c:2250 #11 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #12 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #13 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 2 (process 21752): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x0028bb61 in strcasecmp () from
[asterisk-users] Help! - Double NAT issue
Hi folks. Please don't flame me but I've been googling around for days, read a tremendous amount, tried everything, and still no go. This is most definitely a typical newbie question. - I sure hope there's somebody(s) out there who'll humble themselves to help me out. I've set up an 'out of the box' basic Asterisk server running on Slackware Linux. - It basically works fine. - The wife and I are having lots of fun playing around with all the VOIP phones I'm using to talk to the thing. Now. - I want to try to take a phone to my office and try to connect from there. - But I can't. - Sound familiar? Here's my setup some scenarios: |---Home-||-OFFICE---| Asterisk box Linksys WRT54G---Internet---Linksys WRT54G At home... Router: Linksys WRT54G Public IP address: 61.25.172.48 (static) Public Netmask: 255.255.255.128 DNS1: 220.152.38.233 DNS2: 220.152.38.201 Internal IP range: 10.0.0.xxx Internal IP Netmask:255.255.255.0 Router's internal IP: 10.0.0.1 DMZ Enabled, points to: 10.0.0.12 (Asterisk server) (see below) DHCP Enabled, pool starts: 10.0.0.100 Asterisk Server IP address: 10.0.0.12 Netmask:255.255.255.0 DNS:Same as router's. Changes made to sip.conf externip=61.25.172.48 localnet=10.0.0.0/255.255.255.0 nat=yes FYI - No other changes made to ANY of Asterisk's .conf files. - It's a basic 'vanilla' test box. At the office... Router:Linksys WRT54G (out of the box config) Scenarios: I have a Sipura SPA-1001, Cisco-7940, Cisco-7905, and X-Lite running on my home PC. Although I've got DNS servers assigned, I'm not using server.domain names (IP addresses only). - So I believe DNS is not an issue. Scenario A. - When the devices are 'pointing' to the Asterisk server's 'internal' IP (10.0.0.12), they all register and work fine. Scenario B. - If I configurer a phone to use (as a proxy) the home's 'public IP' (61.25.172.48), it works fine. - This tells me (I believe) that the phone is going to the router's 'public IP' but since DMZ is turned on, all the ports are forwarded to the Asterisk box's 'internal' IP (10.0.0.12). Scenario C. - The problem... If I take a device to my office (i.e. the Sipura) and connect it. - It is configured to 'talk' to my home's 'public IP'. - This thing doesn't even REGISTER with the Asterisk server. - So I can't even try to make a call. This is verified (from the office) by being telnet(ted) into my home Asterisk box and watching it's console. Anybody have any clue? If you want to try for yourself, set up a device and try to connect to my box's 'public IP' (above) and use a username of '60' with a password of '1234'. - If that works, try extension '1000' and see if you get the Asterisk box's 'congratulations' message. I'd be very interested in your results. Also, if anybody wants to take this off-forum and discuss/help me out, I'll be greatly thankful. - I have a Broadvoice account and we can even establish a phonecon. Thanks VERY MUCH in advance. Gary Guthary ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie IVR: How to read() before playback() isfinished?
- Original Message - From: Lee, John (Sydney) [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 19, 2008 11:48 PM Subject: [asterisk-users] Newbie IVR: How to read() before playback() isfinished? I am working on a menu to accept input from a caller like as follows: Exten = 100,1,Answer() Exten = 100,n,Playback(LONG-MESSAGE) Exten = 100,n,Read(OPTION,,2) ... When I tested it, I noticed if I start pressing a key before the Playback() is finished, the input is not buffered (simply ignored) and I have to listen to the whole message before I could re-enter again. Is there a way that I could press a key and it will be Read() before the Playback is finished? It seems like a lot of IVR system in the market can doing that and I am wondering if I have missed something in Asterisk. Any thoughts? Use Read( ) app to play your LONG-MESSAGE ___ -- 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
[asterisk-users] How to get call back during attendant transfer?
Asterisk 1.2.26.2 On an ACD call, I can press 0 to do attendant transfer. After talking to the transfered party, I want to cancel the transfer and get back to the original party. If I press *, it will disconnect me and complete the transfer. How can I set it up so I can press * and get the call back? I notice that after the first attendant transfer, the transfered party can do another attendant transfer and this time * key behave differently. If he press * , he get the transfered call back. Why it works for the second transfer not for the first transfer? Gary___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
I will be out of the office on Wednesday, January 9, 2008. If this is an emergency, please call Customer Service at (877) 791-7700. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call-limit in database
I will be out of the office until Wednesday, January 2, 2008. If this is an emergency, please call Customer Service at (877) 791-7700. Thank you have a great holiday season! ___ --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
[asterisk-users] How to play Asterisk .raw file
I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary ___ --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
[asterisk-users] Configure one call per line on Cisco 7941/7961
Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961
David, Yes, I'm aware of that, but unfortunately it does two calls on each line appearance (button), so the first two calls go on line 1, and the third will appear on line 2. I'd like to limit it to 1 call per line. Any ideas? Gary On 9/25/07, David Cook [EMAIL PROTECTED] wrote: Gary, if you register multiple lines with the same SIP credentials the phone will do rollover and take care of it. (2nd call comes in on L2, etc.) - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-25-07 6:37 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [on-asterisk] Configure one call per line on Cisco 7941/7961 Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backports to 1.2.14 of 1.4.0 app_queue features.
Sorry to drag up an old thread, but the backport of ringinuse is a godsend for those of use stuck using asterisk 1.2 (trixbox 2.2). Many thanks, Gavin GTG On 1/21/07, Gavin Hamill [EMAIL PROTECTED] wrote: Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] Dead SIP channels
I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean? Here is the output from 'sip show channels': Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 195.7.123.234 +180924402 3c3c4cee419 00102/0 alaw No Tx: ACK 9.9.94.9 6478517573 2752611-195 00101/1 ulaw No Rx: ACK 136.59.30.19 8787041796 76775e35788 00102/0 ulaw No Tx: ACK 9.9.95.13 9057047798 2752419-199 00101/1 ulaw No Rx: ACK 195.7.123.234 +011503733 25afde8070b 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011503733 71688696061 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011503733 1700ab8b2ae 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011578435 0ecb33f75bb 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 71eac20715c 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 01b9eacf6de 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 744e7a3f501 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 0080443e6ad 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 6f3745a266d 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011221693 3b705a03141 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 4ab469132b7 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 0b2dcf2332b 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 583bd73d09a 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011593222 4d237ba325e 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011639103 33f84238290 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011526778 72bd7b5f080 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011527693 0ffa93c642d 00102/2 unkn No (d) Rx: BYE gary___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 37, Issue 88
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users@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 [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. Re: 99 bottles of beer (David Cook) 2. DUNDi, So Easy A Caveman Could Do It! (JR Richardson) 3. Polycom behind NAT won't register to * server behind ALG (Matthew Warren) 4. Re: Polycom behind NAT won't register to * server behind ALG (Alex Balashov) 5. Re: Polycom and NAT (Darryl Dunkin) 6. Re: Polycom behind NAT won't register to * server behind ALG (Henry L.Coleman) 7. Re: Polycom behind NAT won't register to *serverbehind ALG (Marty Mastera) 8. rfc3680, reginfo+xml (Olivier) 9. How to re-read values from database in Trixbox (Edgar Guadamuz) 10. Re: How to re-read values from database in Trixbox (Diego Iastrubni) 11. Re: Saftware RAID1 or Hardware RAID1 with Asterisk (Richard Scobie) 12. How do I configure asterisk? (fateme fatah) 13. Which interface? (fateme fatah) 14. Re: rfc3680, reginfo+xml (Raj Jain) 15. Cisco firmwares 3.6.3 vs 3.8.6 (Adrian Marsh) 16. Re: compatibility of PRI Two B channel transfers TBTC/2BTC (Matt Florell) 17. Re: DUNDi, So Easy A Caveman Could Do It! (Lenz) 18. Re: Cisco firmwares 3.6.3 vs 3.8.6 (Arnaud Ligot) 19. Re: rfc3680, reginfo+xml (Olivier) 20. asterisk with FAX problem (satish patel) 21. Re: Polycom and NAT (Klaverstyn, David C) 22. Re: How do I configure asterisk? (Atis) 23. Re: Polycom behind NAT won't register to * server behind ALG (Eric ManxPower Wieling) 24. Re: 99 bottles of beer (Russell Handorf) 25. Re: Saftware RAID1 or Hardware RAID1 with Asterisk (Steven) -- Message: 1 Date: Tue, 21 Aug 2007 21:01:50 -0400 From: David Cook [EMAIL PROTECTED] Subject: Re: [asterisk-users] 99 bottles of beer To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote: To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... Before this thread I already had a Firecracker on the server, a fair assortment of lights and the sprinklers are on an X10Pro Irrigation Controller. Damn, now I'm gonna be up all night. - dbc. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070821/291de3 99/attachment-0001.htm -- Message: 2 Date: Tue, 21 Aug 2007 20:51:51 -0500 From: JR Richardson [EMAIL PROTECTED] Subject: [asterisk-users] DUNDi, So Easy A Caveman Could Do It! To: asterisk-users@lists.digium.com, [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- JR Richardson Engineering for the Masses -- Message: 3 Date: Tue, 21 Aug 2007 22:03:30 -0400 From: Matthew Warren [EMAIL PROTECTED] Subject: [asterisk-users] Polycom behind NAT won't register to * server behind ALG To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Polycom's were simply not originally built for multi location VoIP. There is no NAT support in the Polycom's. We have several networks, being an ISP, and have found that when transversing one network say 192.168.2.x with the * box on a 192.168.1.x the polycoms were able to communicate however sustained a lot of one way audio problems. Moving thim onto the same network is the only thing we have been able to reliable do. According to Polycom Support this is what they are intended for and no definitive answer as to whether they would support Stun or another method in the future. At least as of 6 months ago. Matt -- Message: 4 Date: Tue, 21 Aug 2007 22:17:17 -0400 (EDT) From: Alex Balashov [EMAIL PROTECTED] Subject: Re: [asterisk-users] Polycom behind NAT won't register to * server behind ALG
[asterisk-users] Problem with H option of Dial()
I just upgrade my test server to Asterisk 1.4.7.1 and having problem using H option in Dial() app. When press * during the call from caller side, Asterisk does not disconnect the call. The * just pass through. Here is my test dial plan: exten = 8111001001,1,Answer() exten = 8111001001,n,Wait(2) exten = 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|HL(12:61000:3)) exten = 8111001001,n,Hangup() It worked with previous version 1.2.18 on this box. I am new to 1.4. Do I miss something? Or is it just a bug? Gary Chen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with H option of Dial()
I also tried blind transfer with t option and it did not work. I added following into my dial plan contest: include = featuremap exten = 8111001001,1,Answer() exten = 8111001001,n,Wait(2) exten = 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|tHL(12:61000:3)) exten = 8111001001,n,Hangup() It still does not work. I issue show features in CLI it show this: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# #1 Attended Transfer One Touch Monitor Disconnect Call * * Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 What else do I need to do to make the features work? Gary Chen - Original Message - From: Gary Chen To: asterisk-users@lists.digium.com Sent: Tuesday, July 17, 2007 8:24 AM Subject: [asterisk-users] Problem with H option of Dial() I just upgrade my test server to Asterisk 1.4.7.1 and having problem using H option in Dial() app. When press * during the call from caller side, Asterisk does not disconnect the call. The * just pass through. Here is my test dial plan: exten = 8111001001,1,Answer() exten = 8111001001,n,Wait(2) exten = 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|HL(12:61000:3)) exten = 8111001001,n,Hangup() It worked with previous version 1.2.18 on this box. I am new to 1.4. Do I miss something? Or is it just a bug? Gary Chen -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Edit ulaw file
I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen___ --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
[asterisk-users] SIP / STUN / Network - Help!!
Hi Everyone. I'm in a quandry don't know which way to go. - Obviously I'm an Asterisk newbie although I've been watching this list for over 2 years now. I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On the same LAN I've got a Cisco 7940, 7960, and Sipura SPA-1001 (obviously, all using SIP). - They all work fine. - They can call each other, leave retrieve voicemail, etc. - It's a VERY basic setup. - The box also has a Digium TDM-400P card with one each FXO FXS ports but I haven't gotten that far in my testing. What I want to do is take one of my SIP devices to my office (which is ALSO behind another NAT) and try to connect with my home Asterisk box with it. I've read in the VOIP WIKI that if both server SIP device are behind (separate, non-co-located) NATs, you need both port forwarding (at the Asterisk server side) AND the use of STUN (I'm guessing STUN is for RTP traffic). - Is this correct? For port forwarding, my AsteriskNOW box has a static IP on the inside of my NAT and I've configured the LinkSys router to port-forward ports 5060 (TCP UDP) and all the RTP port range used (UDP only) to the static IP of the AsteriskNOW box. - Was this the right thing to do? Although my home IP is supposed to be 'dynamic', it hasn't changed in 4 years! (shhh! Don't tell anyone, okay) - My LinkSys router DHCP's it's 'real-world-IP-address' DNS server, etc., from my cable-modem. So I set up yet another Sipura SPA-1001, pointed it to my 'real-world' IP, etc., took it to my my office, and it didn't work. - Naturally. - My luck. Is it because I need a STUN server to go through? - Or what? The reason I chose the Sipura over the Cisco hardphone is I've read that Sipura works well via STUN. I know Digium developed IAX to overcome this problem, but none of my devices support IAX. I've read that the STUN server CANNOT be behind a NAT. - But there's free ones we can use. - My problem is that all the free STUN servers are in North America. - I live in Japan. - About 30 miles north of Tokyo. - And my office is in downtown Tokyo. - If I were to use a N.A. STUN server, I'm afraid I'll run into all kinds of latency problems. I have no clue how on how to build a STUN server. - And would like to avoid this if possible. But I've also read that if the Asterisk box has a 'real-world-IP' (plugging my Asterisk box directly into my cable modem), port forwarding STUN are not needed on the devices. - For me, this would mean also making my Asterisk box also a router so all the other stuff I have here at home would still work. - Something I've never done but am willing to give it a shot. If anybody wants to take me by the hand and lead me to a solution, I'll be truly gratefull! - If you want to take it off-line (off-list), please e-mail me: gary at guthary dot com Oh Yeah! - Whatever I learn from this adventure will be fully documented an made freely available on my website for the next newbie who runs into a similar situation. Thanks in advance I sincerely apologize if this posting is not appropriate for this list. Gary Guthary ___ --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
[asterisk-users] MOH question w/Cisco 79xx phones
Hi Everyone Got a newbie type question regarding MOH Cisco phones. I'm still new to Asterisk (very new in fact) built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems. My problem is with MOH (Music On Hold). - The 7940/7960 are somewhat okay, but the 7905 is another story. When a is call from a 7940/7960 is placed on 'hold' (by the calling party), MOH starts up on the 7940/7960, plays for about a second or two, then drops out for about a second or so, then continues. - After that, it continues to play okay. But when a call from the 7905 is placed on 'hold' (by the calling party), MOH starts up on the 7905, plays for a second or two, drops out for a sec, starts again for a sec or so, drops out, starts back up, drops out, etc., etc., etc Just up and down. - Kinda' like a Yo-Yo. Also - When the call from the 7905 is placed on hold, I see the following warning at the Asterisk CLI: [Jun 29 22:18:28] NOTICE[3376]: rtp.c783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.0.0.110 I don't see this warning when the 7940/7960 is playing MOH. I'm using basic default settings for just about everything. - Could this be with the RTP config? - The 7905 Audio settings? Anybody have a clue? Thanks in advance. Gary Guthary ___ --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
[asterisk-users] Test Message
Sorry to clutter up the mailiing list, but I've been unable to post to this list for the past 2 WEEKS! My ISP's blocking SMPT from other than his own servers. I think I've worked around it. - But if I see this message in the digest then I know I'm okay. Again. - Sorry for any inconvenience. Gary Guthary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
- Original Message - From: Darrick Hartman (lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, June 24, 2007 11:25 AM Subject: Re: [asterisk-users] inband DTMF for g729 Gang Chen wrote: On 6/22/07, Gary Chen [EMAIL PROTECTED] wrote: We are using Level 3. At this point, changing carrier is not an option. Gary, I use Level(3) with G729a and RFC2833. No problems, no requirement for inband G729. -- Kristian Kielhofner I can connect to Asterisk IVR using a SIP phone and send RFC2833 with g729. It works fine. But when test call from PSTN to Asterisk, if I set dtmf=auto with g729, I got warning saying something like * does not support inband for g729 and sutomaticlly switch to rfc2833. If I set dtmf=g729, there is no warning but I have the same problem. This tells me that Level3 does use inband for g729 or maybe I am doing something wrong . Gary Gary, I'll restate what Kristian just said above. You do NOT need inband for Level 3. Set dtmf=RFC2833. Do you have the correct g729 codec licenses installed? This may be more of a transcoding issue than anything else. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com We have not yet purchase the g729 codec licenses. I want to test it out first before we buy any license. I download g729 from Internet. I did set dtmfmode=rfc2833. It worked if I use an SIP phone connect to Asterisk using g729 and send dtmf tone using rfc2833. But not from PSTN through Level 3 . Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] international numbers...
This is the required dial plan: 0+61|XXX. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall Sent: Friday, June 22, 2007 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] international numbers... Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +61242110 to something like 02422110 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XX but that still seems wrong to me. it could match other numbers. anyone had to do this in the past ? thanks. ___ --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
[asterisk-users] POTS - Incoming Voice or Fax - How to tell?
Hi Folks - This may sound weird - but here goes: I live in Japan and on my home POTS line I have a Fax/Phone machine. If I receive a fax, the thing automatically switches to 'fax mode' and prints the fax. If the call is a 'voice call', it sits there rings until answered. The above is very reliable and works okay. Of course signalling differs in each country (and even by Telco supplier) but my question is: Basically, how does the machine know if the incoming call is a fax or voice call? If there's a way to tell.. Is there a way (for example) to plug the POTS line into a FXS port then plug the fax machine into the FXO port... AND... If the incoming call is a fax, let Asterisk route it to the FXO port to print the fax. If the incoming call is voice, have Asterisk send the call to one of the SIP hardphones. Of course, Asterisk would have to figure out what type of incoming call this is. Just thinking. - Is this do-able? Thanks in advance Gary Guthary ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] searching for compatible servers
Everyone is going to have their sacred cow on this one so suspect you might have opened a can of worms ... I can tell you that I have very good results using a number of different Intel based SuperMicro servers ... these seem to be very mundane and extremely well behaved ... I have used both Digium and Sangoma cards in them (TDM only, have not tried T1's or ISDN) ... my only beef with them is that they seem rather noisy (very loud cooling fans) ... I have also used a couple entry level Intel based Dell servers with good results and can tell you that these seem to be a good bit quieter than the SuperMicro ... however, the quality of construction and components used on the Dell seems inferior to the SuperMicro ... I have also used a couple mid range HP servers with good results ... the HP is very nicely made and seems to be a notch above the SuperMicro in terms of overall quality of construction and components used ... however, they are about 20% more expensive in similar configuration ... I have had good results using the new 300mb SATA Raid setup from Adaptec ... I normally use CentOS as my OS and the installation utility finds the controller and could not be any simpler ... would expect similar with most RedHat based Linux flavors ... in general, have always had good luck with Adaptec drive controllers ... just be careful to use SATA drives that are specifically intended for use in a RAID, not common workstation drives ... there is a difference and it can bite you in the hind quarters if you buy the wrong type of hard drives and try to use them in a RAID ... Did recently have some trouble with an Intel 1gb NIC ... this surprised me ... I have always favored Intel NIC's mainly because I am lazy and the OS just seemed to find them without having to jump through any hoops ... but this fancy new server class 1gb Intel NIC required that I hunt down and install a unique driver for a CentOS 4.x install ... but this was an odd ball ... most 10/100 and older 1gb Intel NIC's have worked without issue for me ... have had generally good experience with 3Com and Realtec also ... I think the only server class hardware that I recall giving me fits was an ancient Compaq server that someone gave me ... I messed with that one for a week or so on and off and never did get the darn thing to run Linux let alone Asterisk ... As far as I can tell, the only really temperamental aspect is TDM cards from Digium ... while the cards are generally of decent quality, they seem to be a bit picky about what kind of PCI slot they will work with ... so far, this has not been a major problem for me as the hardware I used is purposely very mundane ... but with the published compatibility list hopelessly out of date, you stand some risk of buying a server with a motherboard that the Digium TDM card will not take to ... I have NEVER heard of this problem with Sangoma cards ... Most of my installs these days are on embedded hardware ... I favor the Astlinux flavor of Asterisk and like my PBX's to be small, fanless, lean and mean ... for these I have tried a number of fanless type barebones systems and finally settled on the Lex Neo/Twister models as being my production standard ... these are VIA C3 1ghz machines that are similar to a Mini-Itx ... the Lex Twister model will handle a Digium TDM card nicely and still have room for a 2.5 in hard drive if you want one ... the Lex Neo has no card slot so is not suggested if you want a PCI card that supports connection to the PSTN, but will take a 2.5 in hard drive ... both models have 3+ Realtec NIC's built in which works well with Astlinux when used in router/firewall mode ... With Astlinux, I normally boot off a CF card and forego the moving parts associated with the hard drive but to each his own ... anyway, them's my 2 cents ... Regards G.Hendershot From: Hart Green [mailto:[EMAIL PROTECTED] Sent: Friday, June 22, 2007 11:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] searching for compatible servers Im trying to find the best hardware to run asterisk on. I see that the compatibility list is a little dated. Any recommendations out there? This is for a 19 phone system with 2 tdm cards. Thanks Hart Green -- Internal Virus Database is out-of-date. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.4/705 - Release Date: 2/27/2007 3:24 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] inband DTMF for g729
Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
We are using Level 3. At this point, changing carrier is not an option. - Original Message - From: Matthew Fredrickson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 22, 2007 3:20 PM Subject: Re: [asterisk-users] inband DTMF for g729 Sounds like you need a new SIP carrier. G.729 has a way of destroying inband DTMF tones. --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 22, 2007, at 1:20 PM, Gary Chen wrote: Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Transit problem
Hi! Hope someone can help me. I'm trying to pass SIP traffic from one asterisk to another through a third server. Here is the desired scenario: ServerA -- SIP -- ServerB -- SIP -- ServerC When a call is placed on a ServerA local, I can see that ServerB receives the call and dials ServerC. But ServerC says: Jun 8 09:38:32 NOTICE[3269] chan_sip.c: Failed to authenticate user asterisk sip:[EMAIL PROTECTED];tag=as15c8b5e0 However, when I change the configuration between ServerA and ServerB such that: ServerA -- IAX/2 -- ServerB -- SIP -- ServerC This works just fine. If I understand correctly, ServerA only needs to authenticate to ServerB. The fact that ServerB dials ServerC when both legs are SIP seems to indicate that there is no AUTH problem between A and B. And with the 2nd scenario, it proves that there is no auth issue between B and C. Am I missing something? Has anybody got a recipe for this? I'd appreciate any info. Thanks Jug Mensenares ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
