[asterisk-users] transmit_silence not properly recognized on 1.8 ?
Hello, I've got a problem at the moment, that setting transmit_silence = yes seems to have no effect on Asterisk 1.8-Certified. Although it's enabled and core show settings confirms, that it is really enabled, there are no RTP packets sent by Asterisk when waiting for DMTF input or when Wait() is called. Also, there seems to be a small gap of 2 or 3 not sent packets when playing several files one after another. At the moment I'm using Asterisk 1.8 Certified (Cert 5). In asterisk.conf I've set: highpriority = yes languageprefix = yes internal_timing = yes defaultlanguage = de transmit_silence = yes transcode_via_sln = yes documentation_language = en_US core show settings says: PBX Core settings - Version: 1.8.15-cert5 Build Options: DONT_OPTIMIZE, LOADABLE_MODULES, BUILD_NATIVE, G711_NEW_ALGORITHM, G711_REDUCED_BRANCHING, TEST_CODING_TABLES Maximum calls: Not set Maximum open file handles: Not set Verbosity: 8 Debug level: 0 Maximum load average:0.00 Minimum free memory: 0 MB Startup time:21:29:18 Last reload time:21:29:18 System: Linux/3.2.0-4-amd64 built by root on x86_64 2014-05-24 18:42:53 UTC System name: Entity ID: 00:25:xx:xx:xx:xx Default language:de Language prefix: Enabled User name and group: / xxx Executable includes: Disabled Transcode via SLIN: Enabled Internal timing: Enabled Transmit silence during rec: Enabled Generic PLC: Enabled Is there anything I could do for forcing transmitting silence without stopping sending RTP packets? Maybe this would solve the issue with the packet gaps between playbacks also. Greetings Max -- _ -- 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] Hack
Hi John, do you have a line in your sip.conf saying match_auth_username=yes ? It needs to be in the general context or (I think) inside the peer configuration. I need to use this with user based auth, don't know if it's mandatory for IP based auth also. Greetings, Max signature.asc Description: OpenPGP digital 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] Error 488 Not Acceptable Here
Hi, Maybe you have not allowed T.38 as acceptable codec ;-) You can try with allow=all in your sip.conf. Am 22.05.2013 16:39, schrieb Andrew Colin: Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax? -- _ -- 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 -- -- - Portunity GmbH - Werner-Seelenbinder-Str. 23 -- 42477 Radevormwald - Germany - - Portal: http://www.portunity.de - - General: Phone: +49 (0)202 - 69555 - 0 - eMail/SIP: i...@portunity.de - Fax: +49 (0)202 - 69555 - 190 - - Support: Phone: +49 (0)202 - 69555 - 300 - eMail/SIP: supp...@portunity.de - - Amtsgericht Koeln HRB 38162 - USt-Identnummer DE206277867 - Geschaeftsfuehrung: Bjoern Ruecker, Bernd Schnell -- -- _ -- 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] SetCallerPres questions
Hi, In 1.8 setting this flag it does not remove any caller id related data, it just sets an information on how to handle the data. If the direct dial target is a phone, it MAY show the callerid anyway, but dialling out on a PSTN/VoIP trunk also sends the callerid. But: If you place an emergency call the emergency service should always be able to see your callerid - even if you set SetCallerPres(Prohib). Max Am 16.05.2013 00:12, schrieb Adam Moffett: Does SetCallerPres(Prohib) remove the ANI data from a SIP call or does it simply set a flag telling other devices not to display the data? In other words, could another system override that and see the caller ID anyway? The answer may affect how I handle 911 calls, so I'm very curious. -- _ -- 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] Asterisk 1.8-cert and AGC
Hi, I'm trying to use AGC in combination with Asterisk 1.8 and an odd telephone which is very loud when used with a headset and more quiet when used normal. Regarding to the documentation, AGC should be available since * 1.6 - but every time I want to set it, the CLI tells me: -- Executing [0160xxx@intern:2] Set(SIP/intern-xxx-00d2, AGC(rx)=8000) in new stack [May 18 16:58:32] ERROR[23958]: pbx.c:3767 ast_func_write: Function AGC not registered What am I doing wrong? I pasted the example from the documentation into my dial plan, but this error message keeps occuring :-( Do I have to explicitly enable it for compiling? If yes - where could I find it in 'make menuconfig'? Thank you very much! Max -- _ -- 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] DAHDI: How to supress notification of changing CallerID on transfer?
Hi Richard, the macro you linked to did the trick for me - thank you! Greetings from Wuppertal Max Grobecker Am 24.01.2013 00:18, schrieb Richard Mudgett: - Original Message - Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 - IVR answers and puts caller to the chosen queue - Someone picks up the phone (Internal ext. 321) - CallerID shown on customers device changes to 020212345-321 Same when I park the call and pick it up on another phone. I don't want this to happen and can't figure out how to disable this on DAHDI or at least the current channel. I tried facilityenable=no in chan_dahdi.conf, but this only supresses signalling the on hold status. We are using a german ISDN Anlagenanschluss (bri_cpe) whith DDI served by the Deutsche Telekom, connected to a ISDN card which is used with DAHDI. Is there a hidden config flag or something to disable this for DAHDI? Or maybe a channel variable to temporarily disabling this on some channels? chan_dahdi in Asterisk 11 has this option to easily do what you want: ; Send ISDN conected line information. ; ; block: Do not send any connected line information. ; connect: Send connected line information on initial connect. ; update: Same as connect but also send any updates during a call. ; Updates happen if the call is transferred. (Default) ; ;colp_send=update You can also use the interception macros on v1.8. https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Richard -- _ -- 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 -- -- - Portunity GmbH - Werner-Seelenbinder-Str. 23 -- 42477 Radevormwald - Germany - - Portal: http://www.portunity.de - - General: Phone: +49 (0)202 - 69555 - 0 - eMail/SIP: i...@portunity.de - Fax: +49 (0)202 - 69555 - 190 - - Support: Phone: +49 (0)202 - 69555 - 300 - eMail/SIP: supp...@portunity.de - - Amtsgericht Koeln HRB 38162 - USt-Identnummer DE206277867 - Geschaeftsfuehrung: Bjoern Ruecker, Bernd Schnell -- -- _ -- 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] DAHDI: How to supress notification of changing CallerID on transfer?
Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 - IVR answers and puts caller to the chosen queue - Someone picks up the phone (Internal ext. 321) - CallerID shown on customers device changes to 020212345-321 Same when I park the call and pick it up on another phone. I don't want this to happen and can't figure out how to disable this on DAHDI or at least the current channel. I tried facilityenable=no in chan_dahdi.conf, but this only supresses signalling the on hold status. We are using a german ISDN Anlagenanschluss (bri_cpe) whith DDI served by the Deutsche Telekom, connected to a ISDN card which is used with DAHDI. Is there a hidden config flag or something to disable this for DAHDI? Or maybe a channel variable to temporarily disabling this on some channels? THANK YOU! Greetings from Wuppertal Max -- _ -- 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] Automatic Number Identification and anonymous calls
Hello, I know about the german phone system that the sense of an anonymous call is, that the called party has no way to get the caller's number in any way. The last switch honours the anonymous bit and removes the phone numbers before sending the call to the called party. In EURO-ISDN you have a feature called CLIRO which means Calling Line Identification Restriction Override and is only availiable to emergency services or special governmental services. With this line feature the last switch ignores the anonymous bit and you can see both caller id and ANI. The US phone system may work in another way, but if I call another person and want to hide my number I would appreciate if my phone number is not sent to the called party in any field. So this issue sounds to me more like feature than a bug ;-) Greetings from Wuppertal Max Grobecker Am 09.02.2012 11:12, schrieb Olivier: 2012/2/8, Kevin P. Fleming kpflem...@digium.com: On 02/08/2012 12:40 PM, Olivier wrote: 2012/2/8, Kevin P. Flemingkpflem...@digium.com: On 02/08/2012 10:06 AM, Carlos Alvarez wrote: On Wed, Feb 8, 2012 at 2:35 AM, Olivieroza_4...@yahoo.fr mailto:oza_4...@yahoo.fr wrote: I always thought that ANI (Automatic Number Identification) could not directly be set or changed by end users. In this experiment, it seems that if an end user calls anonymously, the ANI is also hidden to the receiving party. I never work with analog lines, but I do believe that anything less than a PRI would have the blocking done at the telco end before you get it. Mostly correct; ANI is only delivered over ISDN, SS7 and *some* EM trunk circuits. Analog circuits have no ability to transport ANI. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 Thanks for telling ANI feature can't come along analog lines. I did another test towards an ISDN BRI line and got the same results : ANI and CID are either both displayed or both hidden at the same time. You need to talk to your provider to see if your circuit is provisioned to send you ANI at all; if it's not sent by the provider, Asterisk just duplicates the CLID into the ANI channel variable. Then, may I say that I would prefer to have the ANI left blank, if the network doesn't provide this data, and let system administrators know about this. Most end-customer (retail) ISDN circuits are not configured to send ANI unless the customer asks for it. This matches what I had in mind : ANI being something dedicated to emergency services that may also be available to others requiring it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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] Early Media configuration doesn't seem to be working
Hi, on a similar setup I set in sip.conf: prematuremedia=no progressinband=never in the peers configuration. With this config you tell Asterisk not to handle inband information at all. But: Maybe you won't get any inband error messages also. Greetings from Wuppertal Max Grobecker Am 07.02.2012 11:53, schrieb Ishfaq Malik: Hi We are using asterisk 1.8.7.0 Our Sip provider is passing us ringing via Early Media, i.e. using a SIP 183 Session Progress, with session description message which is fine for the most part but some of our customers are terminating on an ISDN gateway which doesn't interpret this message and those customers get no ringing. After doing some reading on the subject I have tried the following set prematuremedia=yes in sip.conf set progressinband=never in sip.conf set progressinband=never in the peers configuration in question but the asterisk server still passes on the 183 message and RTP stream rather than converting it to a SIP 180 Ringing message. Is there a problem here or am I misunderstanding something? Thanks in Advance Ish -- _ -- 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 crashes on high IO load
Hi, the log files contained (sometimes) lines about refcount -1 in astobj.c. I also generated core dumps and analyzed them - but there were always errors in another module. Mabye I found the solution: Asterisk seems to crash when a required module cannot be loaded fast enough due to heavy disk usage. When I move the modules directory to another hard disk Asterisk runs fine. I'm using autoload=yes in modules.conf and have several noload lines in it. Is there a possibility to say asterisk to load all modules to RAM at start time and not on demand? Thanks and greetings Max Am 05.04.2011 09:45, schrieb Thorsten Göllner: Did you take a look at /var/log/syslog /var/log/asterisk/messages ? Using Debian? Take a look at iotop (apt-get install iotop). There you can see information about which process consumes high io load. Am 04.04.2011 17:23, schrieb Maximilian Grobecker: Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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
[asterisk-users] Asterisk crashes on high IO load
Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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 crashes on high IO load
Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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
