[asterisk-users] Call ID of the second call leg
Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr table) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you Leandro -- _ -- 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 Realtime issue after registering withx-lite
Can you please show the database entry for that peer then? On Thu, 2012-07-26 at 23:20 +0530, virendra bhati wrote: My sip.conf don't have any entry related to sip pees. I have everything into database. for more details please check below url, which have good example of asterisk realtime http://bahjons.com/stuff/asterisk-realtime-installation-guide On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz motty.c...@gmail.com wrote: can you post your sip.conf for Exten. 1000? it does not seem like you have [1000] mailbox=1000@default Thanks, -motty __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Thursday, July 26, 2012 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Realtime issue after registering withx-lite Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE If anyone have any suggestion please reply to me. -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] callback on busy
Thanks a lot ! I will try the suggested solutions :) Cheers! pepesz On Thu, Jul 26, 2012 at 3:02 PM, pepesz pep...@gmail.com wrote: Dear all, I know the topic comes back like boomerang, but I did not find a nice solution. Does someone has/knows how to achieve call back on busy otherwise called camping? If one is calling the extension and it is busy, then caller should get something like Press 5 to request call back and after the previous call is finished the system should: 1) call caller 2) dial callee or something similar ;) This topic comes back so many times - I'm wonder if there is already a function for that implemented in asterisk (my current one is 10.5) Thanks in advance. pepesz -- _ -- 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 ID of the second call leg
Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr table) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you Leandro -- _ -- 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] MWI not working - Asterisk 1.8.9.2
Hiii, I am testing MWI on my grandstream and bria. Following is sip show peer 1001 * Name : 1001 Secret : Set MD5Secret: Not set Remote Secret: Not set Context : EXT_1001 Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 1001 VM Extension : default LastMsgsSent : 32767/65535 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : 1001 1001 MaxCallBR: 512 kbps Expire : 40 Insecure : no Force rport : No ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : Yes . Following is sip show subscription Peer User Call ID Extension Last state TypeMailboxExpiry 172.16.26.1711002 627149977@172.1 -- none mwi 1002 60 172.16.26.1271001 2068510560-4266 -- none mwi 1001 60 Following is show voicemail - i changed format of the same for general use Mbox User NewMsg 1002 1002 6 1003 1003 0 1004 1004 0 1005 1005 0 1001 1001 28 I can receive listen and also do all stuff using voicemailmain application. But no MWI on any client. is there any thing else i need to check ? can any one help to solve the problem Thanks in advance, Bharat Lalcheta -- _ -- 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] still got ReceiveFax() problem, how to properly setup asterisk fax?
On 07/26/2012 10:33 PM, Roi Stork wrote: I've posted my problem with ReceiveFax() a long time ago. Majority of the incoming faxes still end up with a T2 timeout or hangup (fax session hangup) errors. Our Setup: - we're using the Digium Free Fax module for Asterisk, all settings are default - incoming/outgoing faxes go through an E1 line - faxes are outgoing/received via Sangoma A104DE Card - fax .tiff image is converted to pdf and sent to email - clock source has already been set to NORMAL (from the E1 line), and hardware/software echo cancellation already disabled Why would you disable echo cancellers? That's a terrible idea. -- 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
Re: [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, July 27, 2012 10:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax? On 07/26/2012 10:33 PM, Roi Stork wrote: I've posted my problem with ReceiveFax() a long time ago. Majority of the incoming faxes still end up with a T2 timeout or hangup (fax session hangup) errors. Our Setup: - we're using the Digium Free Fax module for Asterisk, all settings are default - incoming/outgoing faxes go through an E1 line - faxes are outgoing/received via Sangoma A104DE Card - fax .tiff image is converted to pdf and sent to email - clock source has already been set to NORMAL (from the E1 line), and hardware/software echo cancellation already disabled Why would you disable echo cancellers? That's a terrible idea. -- 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 People seem to think that Asterisk won't disable the Echo Canceler when a fax tone is detected. Why they think that is a total mystery to me,. -- _ -- 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] still got ReceiveFax() problem, how to properly setup asterisk fax?
On 07/27/2012 09:53 AM, Eric Wieling wrote: People seem to think that Asterisk won't disable the Echo Canceler when a fax tone is detected. Why they think that is a total mystery to me,. Asterisk doesn't do this, the echo canceller itself does (or DAHDI does, in some cases). With modern ECs, as well, they don't even get disabled when a CED tone (from the answering FAX endpoint) is heard, they instead just turn off their non-linear processors, because it's been found through years of experience that leaving 'most' of the EC still in place makes FAX calls more reliable than if it was completely disabled. Since the OP has a Sangoma card with an Octasic hardware echo canceller on it, he should just leave it alone and let it do its job :-) Turning it off is probably making things worse. Sometimes I wish it was possible to selectively eradicate 'conventional wisdom' from the world! -- 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
Re: [asterisk-users] Video conferencing?
- Original Message - From: Dmitry Melekhov d...@belkam.com To: asterisk-users@lists.digium.com Sent: Thursday, July 26, 2012 10:40:57 PM Subject: Re: [asterisk-users] Video conferencing? 25.07.2012 22:24, Ken D'Ambrosio пишет: Hi, all. I'm 99% sure that Asterisk technically *supports* videoconferencing well, confbridge supports sort of videoconferences , but our users refused to use them because asterisk switches video in the middle of stream and this leads to broken picture. developers refuse to fix this, i.e. add switching on i-frame ... Hi Dmitry! So, our original conversation is here: http://lists.digium.com/pipermail/asterisk-video/2012-April/003621.html As I said in our previous conversation, we don't currently have plans to implement a re-transmission of a new source's I-Frame, or delay switching of the video source until an I-Frame from the new source arrives. I wouldn't characterize that as a refusal. I was simply informing you of the actual state of affairs: we do not have plans at this time to implement this feature. That being said, as I mentioned in our discussion, Asterisk should have some built in mechanisms currently to help alleviate this situation. It should send a SIP INFO request when a video source changes notifying the client that it is the new video source. The client can then transmit a new I-Frame, which alleviates the need for Asterisk to delay switching of the video source (or otherwise manipulate the video stream in some fashion itself). As I asked in the e-mail: 1. What clients are you using? Do they support RFC 5168 (XML Schema for Media Control)? 2. If you get a SIP trace or a packet capture, do you see Asterisk sending the SIP INFO messages when a video update occurs? Since I never got a concrete answer to those questions, I'll ask again: Do your clients support RFC 5168? If not, do you have significant problems with clients that do support RFC 5168? Now, if this is a significant issue for a large number of video clients, including those that support RFC 5168, we could consider implementing something else (such as only switching the video source on the reception of a new I-Frame). As it is, there hasn't been widespread request for this, and - while it would be a very useful and interesting improvement - falls more into the category of improvement or new feature then bug. As such, improvements and new features typically come from either planned development improvements, or from open source contributions from the Asterisk Developer community. Note that we plan the vast majority of our features with the community as AstriCon, which is coming up soon: http://www.astricon.net/2012/astridevcon.aspx Barring that, if this is a feature you would love to have, then you can either write it and submit the contribution to Asterisk, or you could work with developers in the Open Source community to write this feature. I hope this clarifies the refusal. Thanks! -- Matthew Jordan Digium, Inc. | Software Developer 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
Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite
strange last night my serve had this issue but when next morning i check with register 1000 sip account no issue has come thanks for your reply On Fri, Jul 27, 2012 at 1:30 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Can you please show the database entry for that peer then? On Thu, 2012-07-26 at 23:20 +0530, virendra bhati wrote: My sip.conf don't have any entry related to sip pees. I have everything into database. for more details please check below url, which have good example of asterisk realtime http://bahjons.com/stuff/asterisk-realtime-installation-guide On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz motty.c...@gmail.com wrote: can you post your sip.conf for Exten. 1000? it does not seem like you have [1000] mailbox=1000@default Thanks, -motty __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Thursday, July 26, 2012 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Realtime issue after registering withx-lite Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE If anyone have any suggestion please reply to me. -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and
Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0
- Original Message - On 07/26/2012 03:32 PM, Danny Nicholas wrote: Question 1 - I think asterisk only supports a limited set of statuses Asterisk does not *receive* presence updates from Polycom phones (or really, non-Digium phones) at all. Instead, the presence (status) updates you are seeing appear on your phones are the statuses that Asterisk itself generates based on the phones' activity. Ah, I was suspecting that to be the case. Thanks for the info! --Tim -- _ -- 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] CAS T1 - No Ringback
Another mystery for the list, hopefully someone has ideas on a fix... :) I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS, fractional 1-8). Outbound dialing works correctly, but while the call is in progress, there is no 'ringing' heard by the end user. So, on a SIP phone connected to this system, I dial a number, that call goes out DAHDI via the CAS T1, and the remote side is actually ringing (my cell phone for example), but the SIP phone remains silent. If I answer my cell phone, full 2-way audio is present. The telco has already enabled ringback on the circuit but that has not had any effect on the operation. Any thoughts on how to proceed? Here are the pertinent parts of the debug log showing the events on the circuit when dialing: [Jul 27 11:31:20] VERBOSE[14199] app_dial.c: -- Called DAHDI/g1/XXX [Jul 27 11:31:20] DEBUG[14149] devicestate.c: No provider found, checking channel drivers for DAHDI - 1 [Jul 27 11:31:20] DEBUG[14149] devicestate.c: Changing state for DAHDI/1 - state 2 (In use) [Jul 27 11:31:20] DEBUG[14149] devicestate.c: device 'DAHDI/1' state '2' [Jul 27 11:31:20] DEBUG[14190] app_queue.c: Device 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 27 11:31:20] DEBUG[14190] app_queue.c: Device 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_WINKFLASH(3) on channel 1 (index 0) [Jul 27 11:31:21] DEBUG[14199] chan_dahdi.c: Channel 1: Sending 'T355885' to DAHDI_DIAL. [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Sent deferred digit string on channel 1: TXXXYYY [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_HOOKCOMPLETE(9) on channel 1 (index 0) [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_DIALCOMPLETE(6) on channel 1 (index 0) [Jul 27 11:31:22] DEBUG[14199] chan_dahdi.c: Enabled echo cancellation on channel 1 [Jul 27 11:31:22] DEBUG[14199] chan_dahdi.c: Engaged echo training on channel 1 [Jul 27 11:31:22] DEBUG[14199] chan_dahdi.c: Channel 1: Sending 'wwwYw' to DAHDI_DIAL. [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_DIALCOMPLETE(6) on channel 1 (index 0) [Jul 27 11:31:24] DEBUG[14199] chan_dahdi.c: Echo cancellation already on [Jul 27 11:31:24] DEBUG[14149] devicestate.c: No provider found, checking channel drivers for DAHDI - 1 [Jul 27 11:31:24] DEBUG[14149] devicestate.c: Changing state for DAHDI/1 - state 6 (Ringing) [Jul 27 11:31:24] DEBUG[14149] devicestate.c: device 'DAHDI/1' state '6' The odd part is, you can see above the dialed number was XXX, but the actual sequence on the trunk as performed was to dial XXXYYY, then some 'waits', then the last digit Y. Is this normal? --Tim -- _ -- 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] CAS T1 - No Ringback
On Friday 27 Jul 2012, Tim Nelson wrote: Another mystery for the list, hopefully someone has ideas on a fix... :) I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS, fractional 1-8). Outbound dialing works correctly, but while the call is in progress, there is no 'ringing' heard by the end user. So, on a SIP phone connected to this system, I dial a number, that call goes out DAHDI via the CAS T1, and the remote side is actually ringing (my cell phone for example), but the SIP phone remains silent. If I answer my cell phone, full 2-way audio is present. Do you Answer() the SIP phone before dialling DAHDI? Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] CAS T1 - No Ringback
I think its not inbound call its outgoing, and during call progress the remote end events are not passing back to sip. Mitul On Jul 27, 2012 10:36 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: On Friday 27 Jul 2012, Tim Nelson wrote: Another mystery for the list, hopefully someone has ideas on a fix... :) I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS, fractional 1-8). Outbound dialing works correctly, but while the call is in progress, there is no 'ringing' heard by the end user. So, on a SIP phone connected to this system, I dial a number, that call goes out DAHDI via the CAS T1, and the remote side is actually ringing (my cell phone for example), but the SIP phone remains silent. If I answer my cell phone, full 2-way audio is present. Do you Answer() the SIP phone before dialling DAHDI? Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] still got ReceiveFax() problem, how to properly setup asterisk fax?
Ok, Im putting back echo cancellation, since there's no change at all in the fax receiving success rate. I'd like to focus back to the original topic. My incoming faxes end up usually as either timeout or hangup error: timeout - is this supposed to happen in an E1 line? Can the timeout threshold be changed? hangup - currently more frequently encountered. I noticed in the console that the remote channel hangup status 16 shows up in the middle of the fax session. According to sangoma, this is a normal hangup but asterisk fax detects this as error. I have already set it to use the e1 line clock source. On Friday, July 27, 2012, Kevin P. Fleming kpflem...@digium.com wrote: On 07/27/2012 09:53 AM, Eric Wieling wrote: People seem to think that Asterisk won't disable the Echo Canceler when a fax tone is detected. Why they think that is a total mystery to me,. Asterisk doesn't do this, the echo canceller itself does (or DAHDI does, in some cases). With modern ECs, as well, they don't even get disabled when a CED tone (from the answering FAX endpoint) is heard, they instead just turn off their non-linear processors, because it's been found through years of experience that leaving 'most' of the EC still in place makes FAX calls more reliable than if it was completely disabled. Since the OP has a Sangoma card with an Octasic hardware echo canceller on it, he should just leave it alone and let it do its job :-) Turning it off is probably making things worse. Sometimes I wish it was possible to selectively eradicate 'conventional wisdom' from the world! -- 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
[asterisk-users] Asterisk on Dynamic IP to a SIP extension
Verizon has put another good third party DSL supplier out of the DSL business. Their mindset is to kill the competition and then kill DSL and copper althogether in FIOS areas. So am soon losing my static IP and I need to prepare for the change. I currently have Asterisk running using, besides local extensions, a remote SIP extension in another state. In the new configuration both Asterisk and the remote extension will be behind dynamic IP. I will be running dyndns or equivalent and likely ddclient to update IP's. Will there be any issues running in this way? Will Asterisk ride through an IP change without a restart? If there is a definitive wiki topic on this please pass me the link. ddclient is configurable to do any restarts or changes that might be necessary should an IP address change. I am told that Comcast, which I am hoping to get, has sticky dynamic IP meaning the IP addresses rarely if ever change. If that is the case then this is pretty much a non issue. I think they use the router mac address to assign an IP address. Also the version of Asterisk I am running is old - 1.2.35 - yes I know it's old but it works and does what I need. Are there differences in versions on how the above would work? Thanks, Doug-- _ -- 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
