[asterisk-users] About Asterisk with Digium TE405P PRI ISDN cad
Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks -- from longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possible Security issue with Kamailio - Asterisk Realtime integration
Hi I have an installation based on Daniel-Constantin Mierla's excellent Kamailio 3.3 / Asterisk 10 Realtime document ( http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) but have come across an issue which is a potential problem. In this installation all SIP clients register with Kamailio, and the registrations are forwarded to Asterisk. This means that all registered clients (stored in sipregs table) have the same IP address and Port: that of the Kamailio server. The secret which Asterisk reads is empty to avoid Asterisk issuing a challenge. I have discovered that if a client successfully registers with Kamailio, but for whatever reason this user is not in the database Asterisk is accessing - say for example if two MySQL slaves are out of sync - and then sends an INVITE, Asterisk ends up picking the first user in sipregs which shares the same IP and Port as the incoming request and treats this as the Caller. Of course in our scenario there will be many of these because all clients are registered from Kamailio's IP/Port. For example, here is the sequence of database queries Asterisk performs when a client with a From of 101864 attempts to make a call: SELECT * FROM ast_sipusers WHERE name = '101864' AND host = 'dynamic' SELECT * FROM ast_sipusers WHERE name = '101864' SELECT * FROM ast_sipusers WHERE host = '10.5.76.67' AND port = '5060' AND callbackextension = '014373500' OK, the above are fine. Asterisk looks for a user, and a callback exten. SELECT * FROM ast_sipusers WHERE host = '10.5.76.67' AND port = '5060' Now Asterisk looks for a peer. Still OK. SELECT * FROM ast_sipregs WHERE ipaddr = '10.5.76.67' AND port = '5060' Here Asterisk is checking sipregs for ANY entry with the IP and Port of Kamailio. In this case it finds the first such user, 485833 SELECT * FROM ast_sipusers WHERE name = '485833' Now Asterisk treats this call as if it was coming from 485833, which is totally wrong and very bad. Does anyone know what I would need to do in order to ensure that Asterisk rejects the call attempt if it does not find an exact username match? Thanks -Barry Flanagan -- _ -- 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] About Asterisk with Digium TE405P PRI ISDN cad
Hi, I think, you mean connecting the two boxes directly with a cable... not via PSTN, right? 1.) You need a special cross-over cable to connect one Port directly to another Port... (if you want to crimp it yourself, you can find the associated Pins via Google... ethernet crossover cables do not work as they have different links) 2.) configure one end as master (CPN) and the other asterisk as Network (CPN), otherwise you´ll get timing issues... thats all... regards, yves Am 11.02.2013 14:00, schrieb Shitian Long: Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks -- from longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Mediatrix Euro ISDN hangup problem
On 11/02/2013 5.18, Jean-Denis Girard wrote: The Mediatrix sends incoming calls from the PSTN to an Asterisk server via SIP: this works fine. But when the caller hangs up, the Mediatrix doesn't send Bye to Asterisk, so the call is not finished immediately from the Asterisk point of view: the delay is exactly 30 seconds. According to your logs the network sent a DISCONNECT message with progress indicator #8, so in this case is allowed not to release immediately but leave the user the opportunity to listen for tones/message. It's also allowed not to connect (or keep connected) the B channel then send a RELEASE then entering the release request state. May be this behaviour is configurable in the gateway? Then we took a trace on the gateway (see attached file). Line 345, ISDN Disconnect is received from the PSTN. According to the ISDN specs (and a trace I made on an Asterisk server connected via an ISDN card), I think the gateway should reply with a Release, then the network would reply with Release complete. But the Mediatrix never sends Release. After 30 sec, the*network* sends Release (line 445), then the Mediatrix immediately sends Release complete (line 448). Then, the Mediatrix sends SIP / BYE (line 475), and Asterisk immediately hangs up. I think the gateway is working in compliance to ETS 300 102-1 (5.3.4.1 Clearing when tones/announcements provided). 30 Sec. is the time assigned to T306 on the network side. The ETSI specification is available online for free. -- TeeBX VoIP communication platform (coming soon) http://code.google.com/p/teebx/ --- Lightweight++ Business Friendly++ Open++ -- _ -- 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] confbridge and talker
On 02/11/2013 01:20 AM, Dmitry Melekhov wrote: Hello! We use meetme, but, as I understand it will be soon removed from asterisk (already marked as deprecated), so I'm thinking about confbridge migration. Really, we use self-developed (really my ;-) ) web interface to control meetme. We use cli ( over manager ) command to get users list and status. Confbridge doesn't provide user name in list, but I just fixed this here: https://issues.asterisk.org/jira/browse/ASTERISK-18251 it was very easy. But we also rely on T option, which shows talkers in cli. Is there something like this in confbridge? Thank you! Dmitry: A few things: 1. When I looked at ASTERISK-18251, there doesn't appear to be a patch attached to the issue. In order for a patch to be considered for inclusion, it needs to be attached to the issue in question after a license contributor agreement has been signed. You can sign a contributor agreement directly in Jira using the button at the top of the page. 2. MeetMe is not actually deprecated. There was a short period of time in which it was marked as deprecated in Asterisk 10+, but as it provides SLA functionality that has no other alternative in Asterisk, it was instead marked as Extended support. Note that it is still 'core' supported in Asterisk 1.8. In short, it isn't going anywhere any time soon. 3. If you do want to move to ConfBridge, I would highly discourage using the CLI as a means to get run time information. That method is not encouraged for a variety of reasons: a) It requires polling Asterisk instead of responding to events b) If done over AMI, it requires configuring the account with the 'command' class authorization, which is akin to giving external applications control over Asterisk As such, patches that add CLI commands to ConfBridge that provide functionality that is better expressed over AMI are discouraged. In general, you should use AMI actions/events for interacting with ConfBridge. The existing actions/events include notification of talk detection and speaker change [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ConfbridgeTalking Hope this helps, Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quick start configuration sample for chan_dahdi.conf
I am really a beginner of PRI ISDN board, I am wondering if there is a quick start chan_dahdi.conf configuration I could use. I tried to install two FreePBX boxes follow the instructions from http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html; connected them between PRIs, It worked. And now if I refer the FreePBX chan_dahdi.conf it looks like http://pastebin.com/kfWWL6dm; and it seems there is no specific configuration in FreePBX chan_dahdi.conf. And now I tried to add [global] [3:33pm] #include dahdi-channels.conf into chan_dahdi.conf. and do a static-host*CLI dahdi restart still seems no progress… longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Asterisk with Digium TE405P PRI ISDN cad
On Monday 11 February 2013, Shitian Long wrote: Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks If you want to connect the two boxes together via the telephone network, then you will need appropriate NTEs (Network termination Equipment -- the boundary between where the telco's responsibility ends and yours begins) installed, and the telco should give you cables -- or at least advise on wiring. Connecting an Asterisk card to an NTE requires a straight-through cable. If you just want to connect the boxes directly (aot via the telephone network) then you will need to make up a special cable. Get CAT5 cable, plugs and crimping tool. (If you are especially lazy, you can even just cut the plug off one end of a pre-wired CAT5 cable, and crimp your own in place of where it used to be.) Now you need to swap over pin 1 (WHITE/orange) with pin 5 (WHITE/blue) and pin 2 (ORANGE/white) with pin 4 (BLUE/white). It won't do any harm leaving pins 3, 6, 7 and 8 connected, and it will make crimping up the plugs easier. One end: Standard wiring. 1: WHITE/orange 2: ORANGE/white 3: WHITE/green 4: BLUE/white 5: WHITE/blue 6: GREEN/white 7: WHITE/brown 8: BROWN/white Other end: Special wiring for ISDN crossover. 1: WHITE/blue 2: BLUE/white 3: WHITE/green 4: ORANGE/white 5: WHITE/orange 6: GREEN/white 7: WHITE/brown 8: BROWN/white Don't forget, one of the machines has to be told (in chan-dahdi.conf) to pretend it is an NTE rather than subscriber's equipment! -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] target number is busy after some calls
Hi, I used Asterisk 1.8 and I have a gsm modem with 8 port. When I called target number, gsm modem and asterisk show me one of these ports active. after hangup, the actived port is going to idl status and ready to use. but after some call from extension, when I want to call another number, asterisk gives me Busy status, however all ports are idle and ready to use. I think asterisk have to flashed my extension. please let me know what is your idea? -- _ -- 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] [OT] Mediatrix Euro ISDN hangup problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 11/02/2013 03:44, giovanni.v a ￩crit : I think the gateway is working in compliance to ETS 300 102-1 (5.3.4.1 Clearing when tones/announcements provided). 30 Sec. is the time assigned to T306 on the network side. Ok, thanks for your analysis. So the solution would be that the network does not send the progress indicator in the Disconnect message, or find a configuration parameter on the gateway so that it ignores the progress indicator, right? Thanks, - -- Jean-Denis Girard SysNux Syst￨mes Linux en Polyn￩sie franaise http://www.sysnux.pf/ T￩l: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlEZFccACgkQuu7Rv+oOo/g05ACfSCgt6+FVHSub4K1HMymJe6fx 7tkAnA/XWj9fC/xqNrv2+/THW+HfVt9T =2IzF -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Asterisk with Digium TE405P PRI ISDN cad
Thanks for your message. On Feb 11, 2013, at 2:34 PM, Yves A. yves...@gmx.de wrote: Hi, I think, you mean connecting the two boxes directly with a cable... not via PSTN, right? 1.) You need a special cross-over cable to connect one Port directly to another Port... (if you want to crimp it yourself, you can find the associated Pins via Google... ethernet crossover cables do not work as they have different links) Yes I made a special cross-over cable according to PRI cable pattern. 2.) configure one end as master (CPN) and the other asterisk as Network (CPN), otherwise you´ll get timing issues... I think I lack of some ISDN basic knowledge I am trying to follow the article from http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html to do this task. And I face some general question, for example : According to http://www.facebook.com/photo.php?fbid=10151387245397906set=a.10151387218792906.496748.736667905type=3theater It is a screen shot of one Span of a TE405P card from DAHDI tools. I am wondering if there are some document explain what different configurations means, for example, Current Alarms, Sync Source IRQ Misses, etc…. Thanks f thats all... regards, yves Am 11.02.2013 14:00, schrieb Shitian Long: Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks -- from longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] About Asterisk with Digium TE405P PRI ISDN cad
On Feb 11, 2013, at 4:31 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Monday 11 February 2013, Shitian Long wrote: Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks If you want to connect the two boxes together via the telephone network, then you will need appropriate NTEs (Network termination Equipment -- the boundary between where the telco's responsibility ends and yours begins) installed, and the telco should give you cables -- or at least advise on wiring. Connecting an Asterisk card to an NTE requires a straight-through cable. If you just want to connect the boxes directly (aot via the telephone network) then you will need to make up a special cable. Get CAT5 cable, plugs and crimping tool. (If you are especially lazy, you can even just cut the plug off one end of a pre-wired CAT5 cable, and crimp your own in place of where it used to be.) Now you need to swap over pin 1 (WHITE/orange) with pin 5 (WHITE/blue) and pin 2 (ORANGE/white) with pin 4 (BLUE/white). It won't do any harm leaving pins 3, 6, 7 and 8 connected, and it will make crimping up the plugs easier. One end: Standard wiring. 1: WHITE/orange 2: ORANGE/white 3: WHITE/green 4: BLUE/white 5: WHITE/blue 6: GREEN/white 7: WHITE/brown 8: BROWN/white Other end: Special wiring for ISDN crossover. 1: WHITE/blue 2: BLUE/white 3: WHITE/green 4: ORANGE/white 5: WHITE/orange 6: GREEN/white 7: WHITE/brown 8: BROWN/white Thanks for your message, at moment, I have an Asterisk with a TE405P Quad ports PRI ISDN card, Span 1 connect to a NT(network terminal) equipment, which is a GSM gateway with straight cable Span 2 connect to a TE(telecom equipment), which is an another asterisk installation with TE110P, with cross PRI rewiring cable. At moment, I think the cable are properly connected, since I check out TE405P card, actually two ports indicate green. TE110P indicate Green. And GSM Gateway LAY1 indicate Green, but LAY2 indicate blinking green. I am tying to process the following work according to http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html And may I have some general ISDN questions: according to dahdi_tools, I would be able to check configuration from TE405P card. It has following configurations : Current Alarms: No alarms. Sync Source:T4XXP (PCI) Card 0 Span 1 IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 31/ 31/ 0 112233 1234567890123456789012345678901 my question is what is meaning of each of configuration mentioned above? Thanks Don't forget, one of the machines has to be told (in chan-dahdi.conf) to pretend it is an NTE rather than subscriber's equipment! -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Can't detect remote answer
I forgot to add, cat /proc/dahdi/* yields: Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS (In use) (EC: MG2 - INACTIVE) 2 WCTDM/4/1 FXOKS (EC: MG2 - INACTIVE) 3 WCTDM/4/2 Reserved 4 WCTDM/4/3 Reserved I'm not sure if that (in use) is correct when I'm not actively in a call. This is a very sensitive setup, as a home installation it absolutely *must* pass the gruelling wife test, so I'm keen to see it up and running properly :) On 11 February 2013 16:50, Kevin Wright kev.lee.wri...@gmail.com wrote: I'm attempting to place an outgoing call over POTS/DAHDI, it dials without problem but the remote answer isn't tried. So far I've attempted: - Searching on google - Enabling full and verbose logging (including the debug option of the DAHDI module) - showing NO event at the time I answer on the remote phone a.k.a my mobile - Using another phone on the same line - it works - Receiving a call on that line - no problem - Logging DTMF - it shows digits dialled on my mobile, after I've answered, even whilst it seems to still be ringing locally - Looking on the wiki - Asking on IRC So far, I've found nothing that helps. A sample log output is here: http://pastebin.com/cprZSy9i And my chan_dahdi.conf: http://pastebin.com/L7mBJ66Y And dahdi system.conf: http://pastebin.com/6UQPVC9x also in modprobe.d/dahdi.conf: http://pastebin.com/5ZqtcZdj *any* advice/suggestions at this point would be very much appreciated! -- _ -- 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] Can't detect remote answer
Hey, Just quickly glanced over your data... one problem you have is that you're passing the 'r' flag in your Dial() statement in extensions.conf. That would definitely cause you to have never ending ringback from the analog line (since answer supervision is often not present). You might try removing that and retry your outbound call test. Hope that helps a bit. Matthew Fredrickson Digium, Inc. On 2/11/13 10:54 AM, Kevin Wright wrote: I forgot to add, cat /proc/dahdi/* yields: Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS (In use) (EC: MG2 - INACTIVE) 2 WCTDM/4/1 FXOKS (EC: MG2 - INACTIVE) 3 WCTDM/4/2 Reserved 4 WCTDM/4/3 Reserved I'm not sure if that (in use) is correct when I'm not actively in a call. This is a very sensitive setup, as a home installation it absolutely *must* pass the gruelling wife test, so I'm keen to see it up and running properly :) On 11 February 2013 16:50, Kevin Wright kev.lee.wri...@gmail.com mailto:kev.lee.wri...@gmail.com wrote: I'm attempting to place an outgoing call over POTS/DAHDI, it dials without problem but the remote answer isn't tried. So far I've attempted: * Searching on google * Enabling full and verbose logging (including the debug option of the DAHDI module) - showing NO event at the time I answer on the remote phone a.k.a my mobile * Using another phone on the same line - it works * Receiving a call on that line - no problem * Logging DTMF - it shows digits dialled on my mobile, after I've answered, even whilst it seems to still be ringing locally * Looking on the wiki * Asking on IRC So far, I've found nothing that helps. A sample log output is here: http://pastebin.com/cprZSy9i And my chan_dahdi.conf: http://pastebin.com/L7mBJ66Y And dahdi system.conf: http://pastebin.com/6UQPVC9x also in modprobe.d/dahdi.conf: http://pastebin.com/5ZqtcZdj *any* advice/suggestions at this point would be very much appreciated! -- _ -- 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] Can't detect remote answer
I finally got an answer on IRC, turns out the problem was callprogress=yes (a.k.a. screwupmycalls=yes). Changing it to no and dropping the r seemed to work. I'd like the feature to work properly, but it's more important that I'm able to actually make calls :) I *am* now stuck with a long pause before I hear the outgoing ringing though. Not sure if there's anything I can do to tackle that one. On 11 February 2013 21:32, Matthew Fredrickson cres...@digium.com wrote: Hey, Just quickly glanced over your data... one problem you have is that you're passing the 'r' flag in your Dial() statement in extensions.conf. That would definitely cause you to have never ending ringback from the analog line (since answer supervision is often not present). You might try removing that and retry your outbound call test. Hope that helps a bit. Matthew Fredrickson Digium, Inc. On 2/11/13 10:54 AM, Kevin Wright wrote: I forgot to add, cat /proc/dahdi/* yields: Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS (In use) (EC: MG2 - INACTIVE) 2 WCTDM/4/1 FXOKS (EC: MG2 - INACTIVE) 3 WCTDM/4/2 Reserved 4 WCTDM/4/3 Reserved I'm not sure if that (in use) is correct when I'm not actively in a call. This is a very sensitive setup, as a home installation it absolutely *must* pass the gruelling wife test, so I'm keen to see it up and running properly :) On 11 February 2013 16:50, Kevin Wright kev.lee.wri...@gmail.com mailto:kev.lee.wright@gmail.**com kev.lee.wri...@gmail.com wrote: I'm attempting to place an outgoing call over POTS/DAHDI, it dials without problem but the remote answer isn't tried. So far I've attempted: * Searching on google * Enabling full and verbose logging (including the debug option of the DAHDI module) - showing NO event at the time I answer on the remote phone a.k.a my mobile * Using another phone on the same line - it works * Receiving a call on that line - no problem * Logging DTMF - it shows digits dialled on my mobile, after I've answered, even whilst it seems to still be ringing locally * Looking on the wiki * Asking on IRC So far, I've found nothing that helps. A sample log output is here: http://pastebin.com/cprZSy9i And my chan_dahdi.conf: http://pastebin.com/L7mBJ66Y And dahdi system.conf: http://pastebin.com/6UQPVC9x also in modprobe.d/dahdi.conf: http://pastebin.com/5ZqtcZdj *any* advice/suggestions at this point would be very much appreciated! -- _ -- 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] Can't detect remote answer
I think the default DTMF tone duration is 100ms, if you are dialing 10 digits, that ends up being 1 second delay just to dial the DTMF, not including inter-digit delays. Try setting toneduration=50 in chan_dahdi.conf and see what happens. If you make it too low your telco will miss some digits, you'll need to experiment. You may need to increase it. If the telco switch is busy, like in the middle of the day your minimum workable tone duration may be higher than in the middle of the night. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Wright Sent: Monday, February 11, 2013 4:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't detect remote answer I finally got an answer on IRC, turns out the problem was callprogress=yes (a.k.a. screwupmycalls=yes). Changing it to no and dropping the r seemed to work. I'd like the feature to work properly, but it's more important that I'm able to actually make calls :) I *am* now stuck with a long pause before I hear the outgoing ringing though. Not sure if there's anything I can do to tackle that one. On 11 February 2013 21:32, Matthew Fredrickson cres...@digium.com wrote: Hey, Just quickly glanced over your data... one problem you have is that you're passing the 'r' flag in your Dial() statement in extensions.conf. That would definitely cause you to have never ending ringback from the analog line (since answer supervision is often not present). You might try removing that and retry your outbound call test. Hope that helps a bit. Matthew Fredrickson Digium, Inc. On 2/11/13 10:54 AM, Kevin Wright wrote: I forgot to add, cat /proc/dahdi/* yields: Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS (In use) (EC: MG2 - INACTIVE) 2 WCTDM/4/1 FXOKS (EC: MG2 - INACTIVE) 3 WCTDM/4/2 Reserved 4 WCTDM/4/3 Reserved I'm not sure if that (in use) is correct when I'm not actively in a call. This is a very sensitive setup, as a home installation it absolutely *must* pass the gruelling wife test, so I'm keen to see it up and running properly :) On 11 February 2013 16:50, Kevin Wright kev.lee.wri...@gmail.com mailto:kev.lee.wri...@gmail.com mailto:kev.lee.wri...@gmail.com wrote: I'm attempting to place an outgoing call over POTS/DAHDI, it dials without problem but the remote answer isn't tried. So far I've attempted: * Searching on google * Enabling full and verbose logging (including the debug option of the DAHDI module) - showing NO event at the time I answer on the remote phone a.k.a my mobile * Using another phone on the same line - it works * Receiving a call on that line - no problem * Logging DTMF - it shows digits dialled on my mobile, after I've answered, even whilst it seems to still be ringing locally * Looking on the wiki * Asking on IRC So far, I've found nothing that helps. A sample log output is here: http://pastebin.com/cprZSy9i And my chan_dahdi.conf: http://pastebin.com/L7mBJ66Y And dahdi system.conf: http://pastebin.com/6UQPVC9x also in modprobe.d/dahdi.conf: http://pastebin.com/5ZqtcZdj *any* advice/suggestions at this point would be very much appreciated! -- _ -- 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] Can't detect remote answer
On Mon, 11 Feb 2013, Kevin Wright wrote: I *am* now stuck with a long pause before I hear the outgoing ringing though. Not sure if there's anything I can do to tackle that one. Extension pattern matching (waiting for the digit timeout) can also induce perceived dialing delays. If you crank up console verbosity and debug, does that give you any clues where the delay is occurring? -- 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
Re: [asterisk-users] Can't detect remote answer
Press # after entering the number to see if it's an extension pattern matching delay --Don -Original Message- On Behalf Of Steve Edwards Sent: Monday, February 11, 2013 4:16 PM Extension pattern matching (waiting for the digit timeout) can also induce perceived dialing delays. -- _ -- 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] [OT] Mediatrix Euro ISDN hangup problem
On 11/02/2013 17.01, Jean-Denis Girard wrote: So the solution would be that the network does not send the progress indicator in the Disconnect message, or find a configuration parameter on the gateway so that it ignores the progress indicator, right? I believe the first one will be not a viable option at all, no telco will change any important protocol compliance rule on a per subscriber basis. Now forget your gateway for a moment and make a call on an imaginary phone connected to your PRI, after that call successfully answered let the called party hang up before you do. What you expect to hear? Sure, a disconnect tone... so you will put your phone on hook and the phone will send a disconnect immediately. Your pri-gateway-asterisk should work the same, even if the gateway does not send a disconnect immediately the user who started that call will hang up at least when hearing the disconnect tone (good feedback for humans, no?) and asterisk will send a bye to the sip gateway then that one shall initiate a disconnect on the user side. Check also if your gateway allow for customization to remap isdn/q.931 messages to sip. Sorry, hope you will be able to understand because English isn't my native language. -- TeeBX VoIP communication platform (coming soon) http://code.google.com/p/teebx/ --- Lightweight++ Business Friendly++ Open++ -- _ -- 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 join calls - not barge?
I'd like to have an extension join a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A - B channel? Or is there a more straight forward way to do this? sean -- _ -- 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 11 AGI
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten = s,1,Answer
exten = s,n,Set(ENCODED=${URIENCODE(${ARG1})})
exten = s,n,AGI(agi://localhost/url=${ENCODED})
exten = s,n,Hangup
Using asterisk 11 on the same host with the same config in extensions.conf:
-- Executing [s@VXML:1] Answer(SIP/143-0043, ) in new stack
-- Executing [s@VXML:2] Set(SIP/143-0043,
ENCODED=http%3A%2F%2Fexample.com%2Fcgi-bin%2Favxml) in new stack
-- Executing [s@VXML:3] AGI(SIP/143-0043,
agi://localhost/url=http%3A%2F%2Fexample.com%2Fcgi-bin%2Favxml) in new
stack
[Feb 11 16:28:45] ERROR[28501][C-0012]: utils.c:1187
ast_carefulwrite: write() returned error: Connection refused
[Feb 11 16:28:45] WARNING[28501][C-0012]: res_agi.c:1528
launch_netscript: Connect to
'agi://localhost/url=http%3A%2F%2Fexample.com%2Fcgi-bin%2Favxml' failed:
Connection refused
-- Executing [s@VXML:4] Hangup(SIP/143-0043, ) in new stack
== Spawn extension (VXML, s, 4) exited non-zero on 'SIP/143-0043'
however, my daemon listening on port 4573 never sees activity.
so i set up a super-simple server* on port 4573 and saw that Asterisk is
not attempting the connection.
can someone replicate this behavior ? Or is this just my config ?
* http://jeremy.kister.net/code/asterisk/simple_agid.pl
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] asterisk 11 AGI
On 2/11/2013 11:13 PM, Jeremy Kister wrote: [Feb 11 16:28:45] ERROR[28501][C-0012]: utils.c:1187 ast_carefulwrite: write() returned error: Connection refused [...] can someone replicate this behavior ? Or is this just my config ? opening issue in jira; this is a bug. https://issues.asterisk.org/jira/browse/ASTERISK-21065 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
