Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Excellent! I finally got it working, it was ODBC drivers issue actually. Installed the proper compatible version and its working. There are still few errors which i see on asterisk console: [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: Realtime table book...@meetme requires column 'members', but that column does not exist! [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:440 update_odbc: Key field 'members' does not exist in table 'book...@meetme'. Update will fail [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] -- SIP/callman02-0005 Playing 'conf-onlyperson.ulaw' (language 'en') Also when i try to click the conference to manage it realtime it gives me Error connection to the manager! Following are the database files which i used: /web-meetme/cbmysql/db-admin-user-create.txt /web-meetme/cbmysql/db-table-create-v6.txt /web-meetme/cbmysql/db-tables-v6.txt Am i missing something here now? On Tue, Jul 13, 2010 at 8:43 PM, cov...@ccs.covici.com wrote: cov...@ccs.covici.com wrote: Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. OK, if the argument to meetme is conference number,TcMsrm it does not crash, but if it is conference number, cMs then it dies -- asterisk dies. Is this enough for someone to figure out? -- 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 -- 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 Regards Manmohan Singh Jandu -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
On 19 July 2010 00:35, Anthony Messina amess...@messinet.com wrote: On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Using Aastra 4801 CT phones... [external-context] ; Calls entering from outside the system exten = 1234,1,SIPAddHeader(Alert-Info: info=Bellcore-dr2) ; Double Ring same = n,Dial(SIP/... [internal-context] ; Calls routed from within the system exten = 1234,1,Dial(SIP/... ; No special ring One of the problems with Distinctive Ring tones is that its not consistent, between different phones so if you have a mix of phone types you have a problem. Quite a lot seam to follow the Bellcore stand says the rhythmn of the ring tone, but not the tune, so Bellcore-dr2 might be long long short and bellcore-dr3 might be short short. A type or Morse code I guess... But its hard work to notice the difference in a hurry when you need to answer the phone, hence its not normally enough. In an ideal world you should be able to send the ring tone with the call so sending a URL or embedding it in the sip header, but I've not heard any method to do this. Peter. -- _ -- 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] T.30 fax receiving problem with app_fax
On 10:12 Mon 19 Jul , Nasir Iqbal wrote:
Try 3 second wait between Answer and ReceiveFAX
I'am added but this don't help.
; extensions.conf part with fax
exten = fax,1,Goto(543,1)
exten = 543,1,Answer()
exten = 543,n,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}.tif)
exten = 543,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 543,n,Wait(3)
exten = 543,n,ReceiveFAX(${FAXFILE})
log: http://pastebin.ca/1903440
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Re: [asterisk-users] Busy Lamp Fields
Rob Many thanks for the pointer - I was missing limitonpeers=yes in the general section - Sorry I didn't say version (1.4.33.1) etc forgot with frustration ;-) Paddy -- _ -- 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] Voip rates to Mali
Hello, I am looking for Voip providers for voip minutes to Mali(South Africa) Kindly provide the ratesheet for the same. Regards, Amit Mehta -- _ -- 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] Voip rates to Mali
amit mehta wrote: Hello, I am looking for Voip providers for voip minutes to Mali(South Africa) Kindly provide the ratesheet for the same. Regards, Amit Mehta AQL - 0.1816 GBP/min Magrathea high call volume rate - 0.126 GBP/min They are a couple of UK providers. If it is only that destination you are interested in then look for a provider in South Africa as it will be a lot cheaper since it will only be a local call for them. -- _ -- 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] Voip rates to Mali
Hi Gareth, Thanks for the swift reply. Kindly provide A-Z price list. Regards, Amit On Mon, Jul 19, 2010 at 3:42 PM, Gareth Blades list-aster...@skycomuk.comwrote: amit mehta wrote: Hello, I am looking for Voip providers for voip minutes to Mali(South Africa) Kindly provide the ratesheet for the same. Regards, Amit Mehta AQL - 0.1816 GBP/min Magrathea high call volume rate - 0.126 GBP/min They are a couple of UK providers. If it is only that destination you are interested in then look for a provider in South Africa as it will be a lot cheaper since it will only be a local call for them. -- _ -- 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] Voip rates to Mali
You will need to contact the companies directly if you want a complete price list. We have a very high call volume so the price you get might be different if you dont make many calls. amit mehta wrote: Hi Gareth, Thanks for the swift reply. Kindly provide A-Z price list. Regards, Amit On Mon, Jul 19, 2010 at 3:42 PM, Gareth Blades list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote: amit mehta wrote: Hello, I am looking for Voip providers for voip minutes to Mali(South Africa) Kindly provide the ratesheet for the same. Regards, Amit Mehta AQL - 0.1816 GBP/min Magrathea high call volume rate - 0.126 GBP/min They are a couple of UK providers. If it is only that destination you are interested in then look for a provider in South Africa as it will be a lot cheaper since it will only be a local call for them. -- _ -- 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] hi
Hi, the following configuration: The number 0 will be forwarded to the Ring-Group 25 in which the numbers are 71 and 73. If you call the 0 so the office is ringing at the 71 and 73 . At the terminal stations are Snom 320. In the evening the 71 to make call forwarding via web interface to the 99 (voice mail). Problem: When I ring the phonenumber 0 after call forwarding .the 73 ringing and the Voice Mail (99) didn?t take the call .. But when I call the 71 after call forwarding all works fine and the Voicemail take the Call How can the Number 99 prioritization so that they take the Call in any case ... even if the 73 are still ringing? Or there other options? Regards Beebob in Deutsch: (hoffe das lesen hier welche die Deutsch sprechen...da das oben nur Googletranslate war) Hallo, folgende Konfiguration: Die Rufnummer 0 wird auf die Ring Group 25 weitergeleitet in der sich die Rufnummern 71 und 73 befinden. Ruft man also die 0 an über Amt klingelt es bei der 71 und 73.. An den Endstellen befinden sich Snom 320. Abends soll die 71 per Weboberfläche eine Umleitung auf die 99 (Voicemail) machen. Problem: Wenn ich die 0 nach erfolgter Rufumleitung anrufe klingelt die 73 munter vor sich hin und die Voicemail (99)geht nicht ran.. Rufe ich aber die 71 über Amt an funktioniert die Rufumleitung auf die 99 wunderbar. Wie kann ich die 99 bevorrechtigen, so dass sie auf jeden Fall rangeht...auch wenn die 73 noch klingelt bzw gibt es noch andere Möglichkeiten? Gruß Beebob -- _ -- 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] Skype for Asterisk, Skype For SIP
On 07/18/2010 11:56 AM, Vieri wrote: I still don't see why one should pay for a channel when using a PBX but not when using a client such as Skype. OK, I know that the Skype network is proprietary and I have to accept whatever they say. Usage of the standard Skype client is not free; it involves acting as part of the peer-to-peer Skype network and helping to route calls and in some cases even helping to route media streams for calls. The Skype business solutions (including Skype For Asterisk) don't participate in the peer-to-peer network in this fashion, so every single user of these products does in fact increase the burden on Skype's own network resources. Their solution to this issue is to charge a nominal fee for access to the network. For Skype For Asterisk, calls are still free, and there is no per-channel charge, only a per-user charge (when it begins). This means that for a one-user cost per month, you can receive dozens of simultaneous calls from the Skype network into your Asterisk system. However, if a standard user can call and receive for free then there should be a way to do it from a PBX such as Asterisk. In fact, I came across this project: http://www.mhspot.com/sts/siptosis.html It seems to be a bit of a hack in that it integrates a SIP PBX with a standard Skype client (which doesn't necessarily have to be on the same machine or same OS...). In short, one can use a standard Skype account and not pay a cent for user-to-user calls. Any solution that uses a regular Skype client will be limited to one call at a time; the regular Skype client is not multi-user, and does not support multiple calls (calls can be placed on hold, but there cannot be more than one active call). If this suits your needs, you can certainly try it. There are other Skype gateway solutions that use a similar method, but they are not free. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Skype for Asterisk, Skype For SIP
On 07/18/2010 12:18 PM, Steve Kennedy wrote: On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote: As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. [lots snipped] With Skype's just released SkypeKit it should be possible to build any application with Skype support (costs $20 to register as a dev), they've now got libraries for Linux and now Windows and MacOS X. SkypeKit is basically a headless Skype client. SkypeKit is currently single-user and single-call, just like the regular Skype client. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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
[asterisk-users] Call Forwarding to Voicemail
Hi, the following configuration: The number 0 will be forwarded to the Ring-Group 25 in which the numbers are 71 and 73. If you call the 0 so the office is ringing at the 71 and 73 . At the terminal stations are Snom 320. In the evening the 71 to make call forwarding via web interface to the 99 (voice mail). Problem: When I ring the phonenumber 0 after call forwarding .the 73 ringing and the Voice Mail (99) didn?t take the call .. But when I call the 71 after call forwarding all works fine and the Voicemail take the Call How can the Number 99 prioritization so that they take the Call in any case ... even if the 73 are still ringing? Or there other options? Regards Beebob in Deutsch: (hoffe das lesen hier welche die Deutsch sprechen...da das oben nur Googletranslate war) Hallo, folgende Konfiguration: Die Rufnummer 0 wird auf die Ring Group 25 weitergeleitet in der sich die Rufnummern 71 und 73 befinden. Ruft man also die 0 an über Amt klingelt es bei der 71 und 73.. An den Endstellen befinden sich Snom 320. Abends soll die 71 per Weboberfläche eine Umleitung auf die 99 (Voicemail) machen. Problem: Wenn ich die 0 nach erfolgter Rufumleitung anrufe klingelt die 73 munter vor sich hin und die Voicemail (99)geht nicht ran.. Rufe ich aber die 71 über Amt an funktioniert die Rufumleitung auf die 99 wunderbar. Wie kann ich die 99 bevorrechtigen, so dass sie auf jeden Fall rangeht...auch wenn die 73 noch klingelt bzw gibt es noch andere Möglichkeiten? Gruß Beebob -- _ -- 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 Queue + Caller ID issue
Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption is right. How can I check if those calls are being sent with caller ID. Because all I see on console output for the phone call is this -- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27 instead) -- Nobody picked up in 1000 ms -- Hungup 'DAHDI/56-1' It doesn't show where it actually tried to dial or not. I know it works because if I sent it to the in house number it calls that number and if someone answers it they get the person who is on hold in the queue. It only fails on outside the building calls. So where do I check to see if it is or isn't attaching caller ID. Let's assume I'm right and the CID is the issue; What config and/or context do I need to change so that the when a queue tries to place a call to an agent there is caller ID? James Shigley -- _ -- 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 - Segmentation Fault
OK, now i added the column members in the table booking manually. and disabled selinux to have this working. Now i am struggling with recording option in webmeetme. Not sure on how to enable it, though m checking the checkbox while creating the conference. But where does this save and how to retrieve it? On Mon, Jul 19, 2010 at 9:57 AM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Excellent! I finally got it working, it was ODBC drivers issue actually. Installed the proper compatible version and its working. There are still few errors which i see on asterisk console: [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: Realtime table book...@meetme requires column 'members', but that column does not exist! [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:440 update_odbc: Key field 'members' does not exist in table 'book...@meetme'. Update will fail [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] -- SIP/callman02-0005 Playing 'conf-onlyperson.ulaw' (language 'en') Also when i try to click the conference to manage it realtime it gives me Error connection to the manager! Following are the database files which i used: /web-meetme/cbmysql/db-admin-user-create.txt /web-meetme/cbmysql/db-table-create-v6.txt /web-meetme/cbmysql/db-tables-v6.txt Am i missing something here now? On Tue, Jul 13, 2010 at 8:43 PM, cov...@ccs.covici.com wrote: cov...@ccs.covici.com wrote: Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. OK, if the argument to meetme is conference number,TcMsrm it does not crash, but if it is conference number, cMs then it dies -- asterisk dies. Is this enough for someone to figure out? -- 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 -- 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 Regards Manmohan Singh Jandu -- Thanks Regards Manmohan Singh Jandu --
[asterisk-users] rtsavesysname not working in 1.6.1.20
Hi, I am trying to write the regserver value into my database using ARA but the field keeps empty. Afaik all that needs to be done to make it work is having a db field called regserver, the var systemname set in asterisk.conf and rtsavesysname=yes in sip.conf. But the regserver is not getting updated in my database and I don't see any warnings or errors on CLI. Do I miss something or should it work this way? Thanks. rtsavesysname=yes in sip.conf. But the regserver is not getting updated in my database and I don't see any warnings or errors on CLI. Do I miss something or should it work this way? 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
Re: [asterisk-users] Skype for Asterisk, Skype For SIP
I have been using SiSky Enterprise Edition to integrate Skype with asterisk. You can even call saved skype users from your asterisk system, by creating speed dials in SiSky. Unfortunately it is not a free product but it is very reasonable. Thank you, Brad Finberg - Original Message - From: Alejandro Imass a...@p2ee.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Date: Sunday, July 18 2010 8:57 AM Subject: Re: [asterisk-users] Skype for Asterisk, Skype For SIP On Sun, Jul 18, 2010 at 7:48 AM, Vieri rentor...@yahoo.com wrote: Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype user. I do not need to call landlines via Skype. I think this is _explicitly_ not supported in the Skype for SIP docs. 2) allow Internet Skype users to call my Asterisk PBX Skype user and route the call to a specific Asterisk SIP extension. Here is how it goes from my experience with Skype: each SIP channel will cost you about $5 a month, regardless if you have a landline number with them or not. Your account will be assigned a special Skype number 99x . With that number a Skype user can call you and it will be free. You _cannot_ call Skype users from your PBX, as I stated above, this is an explicit no-no in the docs. If you want to make calls from your PBX to landlines you have to buy Skype credit just like you would with a regular skype client. If you want land-lines to call your PBX you need to purchase a skype number which about $60 a year. At first, I thought it would be simple and free. However, correct me if I'm wrong but the Skype user I can use within the Asterisk PBX cannot be the standard type (used by eg. desktop Skype applications) but needs to be created by the Skype User Manager for Business Solutions. I believe this has a price although Skype For SIP Open Beta seems to be free until Q4 2010. I think you can associate existing skype users to your Business Solutions manager but I still don't understand exactly how or why this is useful, and I don't think it has to do with you being able to call any of them from your PBX. Then again I haven't paid much attention to that and perhaps you have more insight into this. Has anyone found a way to make pure Internet user-to-user Skype/SIP calls via Asterisk (no PSTN involved) for free? As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. Best, Alejandro Imass Thanks, Vieri -- _ -- 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] Does Flash Operator Panel allow for dragging a call into a parking lot?
It's doable with a work around. Create a misc extension with followme set to ##70# which point to your parking lots and failed destination to Misc parking extension. Regards, Bruce On Sun, Jul 18, 2010 at 3:38 PM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: Hi Everyone, If I receive a call on a ZAP line and pickup the call and drag and drop it (by mouse) into a Parking Lot through FOP, it just hangs up. Is this feature supported by FOP? I don't believe so, how would Asterisk know what phone to ring on timeout? 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
[asterisk-users] Pereserving the callerid value when presentation set to witheld over sip
We are a telco so when we receive calls via ISDN and the number is witheld we see the correct presentation value but also still see the actual callers number in the callerid(num) variable. I am trying to forward some of these calls over to one of our other boxes via SIP but I have found that if the number is withend then the sip packet contains :- From: Anonymous sip:anonym...@anonymous.invalid I am running Asterisk 1.4.30 Is there a way to work around this? I would prefer not to have to prefix the number with '141' for example. Thanks Gareth -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with E1
Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 This happens on a regular basis and the E1 becomes up after some time. My zaptel.conf is as follows: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 hardhdlc=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 hardhdlc=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,3,0,ccs,hdb3 # termtype: te bchan=63-77,79-93 hardhdlc=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,4,0,ccs,hdb3 # termtype: te bchan=94-108,110-124 hardhdlc=109 # Global data loadzone = us defaultzone = us Any advice what could be the problem..? Thanks in advance ! -Chetan Meshram-- _ -- 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 when dialing multiple registrations
Hi Nasir, Please don't send me direct emails, unless you want to secure my paid consultancy services or want to do some other business. For setting up the RTP, you need to do it on your firewall, which is receiving RTP traffic from these particular IP address. I can't guess how to do it on your router/firewall. And it may still not solve your problem. I would suggest using separate extensions for separate IP addresses. For wireshark sniffing, my following blog might be helpful: http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ Zeeshan -- www.ilovetovoip.com www.trashinternetexplorer.com On Fri, Jul 16, 2010 at 12:21 PM, Zeeshan Zakaria zisha...@gmail.comwrote: Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp -- _ -- 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 -- Zeeshan A Zakaria -- _ -- 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] Skype for Asterisk, Skype For SIP
--- On Mon, 7/19/10, Kevin P. Fleming kpflem...@digium.com wrote: Usage of the standard Skype client is not free; it involves acting as part of the peer-to-peer Skype network The Skype business solutions (including Skype For Asterisk) don't participate in the peer-to-peer network Any solution that uses a regular Skype client will be limited to one call at a time; Thanks for the explanation! It's crystal-clear now. Vieri -- _ -- 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] Problem with E1
Have you restarted zaptel since making any changes? You are receiving FCS errors but you dont appear to have crc4 specified in your span lines. If you have removed the option but not restarted zaptel yet then do that to see if it fixes the problem. Chetan Meshram wrote: Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 This happens on a regular basis and the E1 becomes up after some time. My zaptel.conf is as follows: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 hardhdlc=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 hardhdlc=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,3,0,ccs,hdb3 # termtype: te bchan=63-77,79-93 hardhdlc=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,4,0,ccs,hdb3 # termtype: te bchan=94-108,110-124 hardhdlc=109 # Global data loadzone= us defaultzone = us Any advice what could be the problem..? Thanks in advance ! -Chetan Meshram -- _ -- 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] Multiple sip.conf files?
Hey, all. I'm trying to do some fun with auto-provisioning of Polycom phones, and one thing that would make life easier for me would be if I could have a per-phone sip.conf file. If not, no biggie -- but if there's a way to do an include (as per extensions.conf) or something, that would be great. I've gone through docs, and an older version of Asterisk: the Future of Telephony implied there was such a feature, but I've seen no mention elsewhere (including, alas, a newer version of the same book). So: can I? Or is it time to just sit down and parse the sip.conf file? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- 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] Multiple sip.conf files?
Ken D'Ambrosio wrote: Hey, all. I'm trying to do some fun with auto-provisioning of Polycom phones, and one thing that would make life easier for me would be if I could have a per-phone sip.conf file. If not, no biggie -- but if there's a way to do an include (as per extensions.conf) or something, that would be great. I've gone through docs, and an older version of Asterisk: the Future of Telephony implied there was such a feature, but I've seen no mention elsewhere (including, alas, a newer version of the same book). So: can I? Or is it time to just sit down and parse the sip.conf file? Thanks! -Ken Why not just use asterisk realtime and store all the information in a database. You can use the same database table to create the provisioning file for the phones aswell. http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip -- _ -- 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] Problem with E1
I did restart the zaptel after making changes.. but just to reconfirm I restarted it again.. but the problem still persists. On Mon, 19 Jul 2010 16:43:57 +0100, Gareth Blades list-aster...@skycomuk.com wrote: Have you restarted zaptel since making any changes? You are receiving FCS errors but you dont appear to have crc4 specified in your span lines. If you have removed the option but not restarted zaptel yet then do that to see if it fixes the problem. Chetan Meshram wrote: Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 This happens on a regular basis and the E1 becomes up after some time. My zaptel.conf is as follows: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 hardhdlc=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 hardhdlc=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,3,0,ccs,hdb3 # termtype: te bchan=63-77,79-93 hardhdlc=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,4,0,ccs,hdb3 # termtype: te bchan=94-108,110-124 hardhdlc=109 # Global data loadzone= us defaultzone = us Any advice what could be the problem..? Thanks in advance ! -Chetan Meshram -- _ -- 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] Problem with E1
When I had this problem I contacted digium who sent me instructions on how to setup the span and make a loopback plug. I then left it running for a while but no errors were reported. The telco then started monitoring the line and after a couple of days diagnosed a faulty card in the local exchange. Chetan Meshram wrote: I did restart the zaptel after making changes.. but just to reconfirm I restarted it again.. but the problem still persists. On Mon, 19 Jul 2010 16:43:57 +0100, Gareth Blades list-aster...@skycomuk.com wrote: Have you restarted zaptel since making any changes? You are receiving FCS errors but you dont appear to have crc4 specified in your span lines. If you have removed the option but not restarted zaptel yet then do that to see if it fixes the problem. Chetan Meshram wrote: Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 This happens on a regular basis and the E1 becomes up after some time. My zaptel.conf is as follows: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 hardhdlc=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 hardhdlc=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,3,0,ccs,hdb3 # termtype: te bchan=63-77,79-93 hardhdlc=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,4,0,ccs,hdb3 # termtype: te bchan=94-108,110-124 hardhdlc=109 # Global data loadzone= us defaultzone = us Any advice what could be the problem..? Thanks in advance ! -Chetan Meshram -- _ -- 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 Queue + Caller ID issue
Let me rephrase this question. What context does a queue use for dialing out? James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A. Shigley Sent: Monday, July 19, 2010 7:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Queue + Caller ID issue Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption is right. How can I check if those calls are being sent with caller ID. Because all I see on console output for the phone call is this -- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27 instead) -- Nobody picked up in 1000 ms -- Hungup 'DAHDI/56-1' It doesn't show where it actually tried to dial or not. I know it works because if I sent it to the in house number it calls that number and if someone answers it they get the person who is on hold in the queue. It only fails on outside the building calls. So where do I check to see if it is or isn't attaching caller ID. Let's assume I'm right and the CID is the issue; What config and/or context do I need to change so that the when a queue tries to place a call to an agent there is caller ID? James Shigley -- _ -- 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 Queue + Caller ID issue
On 07/19/2010 11:08 AM, James A. Shigley wrote: Let me rephrase this question. What context does a queue use for dialing out? It doesn't, it dials the member directly. If you need it to dial out through the dialplan, add a Local channel as a member, instead of the actual channel, and then do your logic in the context/extension you specified before performing the actual dial operation. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Asterisk Queue + Caller ID issue
Could you give me an example because I understand what you said, but not sure what to put in my extensions.conf to accomplish that. James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Monday, July 19, 2010 11:14 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Queue + Caller ID issue On 07/19/2010 11:08 AM, James A. Shigley wrote: Let me rephrase this question. What context does a queue use for dialing out? It doesn't, it dials the member directly. If you need it to dial out through the dialplan, add a Local channel as a member, instead of the actual channel, and then do your logic in the context/extension you specified before performing the actual dial operation. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] digium HW echocancellation - fax tone detection
Hello !
I 'm using a TE405P with a HW echocanceller module attached on it.
dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
As far as I know, the fax tone detection is done on the FW board.
How can I verify that the echo canceller has been turned off ?
When I do a cat /proc/dahdi/1 for span 1 I see still the VPM450 entry
near the channel although this was a fax call with CED tone.
1 TE4/0/1/1 Clear (In use) (EC: VPM450M)
Greping through the source, i see only a CED tone detection
entry in:
static const struct dahdi_echocan_features vpm450m_ec_features = {
.NLP_automatic = 1,
.CED_tx_detect = 1,
.CED_rx_detect = 1,
};
Does it mean that CNG tone is not going to be detected ?
Does the CED tone detection routine also detect ANSam tone for Super G3 fax ?
Thanks
Regards
Hans
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Re: [asterisk-users] Multiple sip.conf files?
Yes, you could do includes in sip.conference like: [general] ... ... ... #include sip1.conf #include sip2.conf Just make sure to do it AFTER the [general] section. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:00 PM, Gareth Blades list-aster...@skycomuk.com wrote: Ken D'Ambrosio wrote: Hey, all. I'm trying to do some fun with auto-provisioning of Polycom pho... Why not just use asterisk realtime and store all the information in a database. You can use the same database table to create the provisioning file for the phones aswell. http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip -- _ -- Bandwidth and Colocati... -- _ -- 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] One way audio when dialing multiple registrations
thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678 i dial using following dial string Dial( SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid == thanks 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] Swedish voiceprmpts
Exist it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium HW echocancellation - fax tone detection
- Johann Steinwendtner steinwendt...@gmx.net wrote:
Hello !
I 'm using a TE405P with a HW echocanceller module attached on it.
dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
As far as I know, the fax tone detection is done on the FW board.
How can I verify that the echo canceller has been turned off ?
When I do a cat /proc/dahdi/1 for span 1 I see still the VPM450 entry
near the channel although this was a fax call with CED tone.
1 TE4/0/1/1 Clear (In use) (EC: VPM450M)
Greping through the source, i see only a CED tone detection
entry in:
static const struct dahdi_echocan_features vpm450m_ec_features = {
.NLP_automatic = 1,
.CED_tx_detect = 1,
.CED_rx_detect = 1,
};
Does it mean that CNG tone is not going to be detected ?
Does the CED tone detection routine also detect ANSam tone for Super
G3 fax ?
Check your '/var/log/messages' or run 'dmesg'. It should say something like
this if the tone is detected and appropriate action taken:
dahdi: Disabled echo canceller NLP because of CED rx detected on channel 97
--Tim
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Re: [asterisk-users] Multiple sip.conf files?
On Mon, Jul 19, 2010 at 12:22:32PM -0400, Zeeshan Zakaria wrote: Yes, you could do includes in sip.conference like: [general] ... ... ... #include sip1.conf #include sip2.conf Just make sure to do it AFTER the [general] section. Actually, you can also use: [general] ... [some-other-stuff] ... #include sip1.conf in sip1.conf: [general](+) ; Be sure to have the '(+)' ; extra lines for the section [general] -- 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
Re: [asterisk-users] Swedish voiceprmpts
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Sent: Monday, July 19, 2010 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Swedish voiceprmpts Exist it? -- -- Try this link http://www.voip-forum.se/ _ -- 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] One way audio when dialing multiple registrations
I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, Nasir Javaid nasirjavaidna...@gmail.com wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678 i dial using following dial string Dial( SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid == thanks 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 -- _ -- 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] digium HW echocancellation - fax tone detection
On 07/19/2010 11:28 AM, Tim Nelson wrote:
- Johann Steinwendtner steinwendt...@gmx.net wrote:
Hello !
I 'm using a TE405P with a HW echocanceller module attached on it.
dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
As far as I know, the fax tone detection is done on the FW board.
How can I verify that the echo canceller has been turned off ?
When I do a cat /proc/dahdi/1 for span 1 I see still the VPM450 entry
near the channel although this was a fax call with CED tone.
1 TE4/0/1/1 Clear (In use) (EC: VPM450M)
Greping through the source, i see only a CED tone detection
entry in:
static const struct dahdi_echocan_features vpm450m_ec_features = {
.NLP_automatic = 1,
.CED_tx_detect = 1,
.CED_rx_detect = 1,
};
Does it mean that CNG tone is not going to be detected ?
Does the CED tone detection routine also detect ANSam tone for Super
G3 fax ?
Check your '/var/log/messages' or run 'dmesg'. It should say something like
this if the tone is detected and appropriate action taken:
dahdi: Disabled echo canceller NLP because of CED rx detected on channel 97
Not quite; none of Digium's hardware echocan products report CED tone
detection or NLP enable/disable events, so you'll never see any messages
that indicate whether the tones were detected or whether the NLP was
disabled automatically or not. Those messages are only generated by the
software CED detector in DAHDI, which isn't used if the hardware echocan
can detect CED itself.
To the original poster: all of Digium's hardware echocan products are
compliant with G.165 and G.168 for tone detection and either completely
disabling the LEC or just disabling the NLP portion, depending on which
tone is detected. CED detectors will typically detect ANSam as if it was
CED, which is the way ANSam was designed to operate on echocan units
that don't have specific ANSam detectors.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Swedish voiceprmpts
Thanks Some companies here in swiden have a swedish female And on the link only male voices -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Danny Nicholas Skickat: den 19 juli 2010 19:11 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Re: [asterisk-users] Swedish voiceprmpts -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Sent: Monday, July 19, 2010 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Swedish voiceprmpts Exist it? -- -- Try this link http://www.voip-forum.se/ _ -- 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] Voice prompts
Have now installed a swedish prompt set In /var/lib/asterisk/sounds/se I run elastix And set Language=se in /etc/asterisk/sip.conf But not work -- _ -- 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] Voice prompts
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Sent: Monday, July 19, 2010 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voice prompts Have now installed a swedish prompt set In /var/lib/asterisk/sounds/se I run elastix And set Language=se in /etc/asterisk/sip.conf But not work -- -- Show your CLI output so we see that you are getting correct playback Here's a QD snippet to let you do a verification Exten = 1234,1,answer Exten = 1234,n,Set(CHANNEL(language)=se) Exten = 1234,n,playback(tt-monkeys) Exten = 1234,n,playback(vm-goodbye) Exten = 1234,n,hangup _ -- 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] DTMF issues/redial tones with rfc2833
Hi, I got the captured packet traces and we could see that it was coming from our asterisk server. Is there any other things that I need to look into, also we have updated from Zaptel to Dahdi 2.0.2.2, but no luck, still the random redial dtmf tones are coming in between calls...Can anyone share their opinion on this...Thank you. Regards Sandesh On Thu, Jul 8, 2010 at 5:21 PM, das sandesh sandesh...@gmail.com wrote: Thanks Zeeshan.that server is located at the headquaters and phones are at different locations, even with default rfc2833 mode, other party IVR prompts was not able to detect the tones, also 'Info' works good but not with internal options like voicemail, etc. And inband is not being used as we are using few g729 calls..Origination source of incoming calls would be from outside numbers.and we have one non sip device FXS router that handles the fax, but its not related to the voice packets... On Thu, Jul 8, 2010 at 3:42 PM, Zeeshan Zakaria zisha...@gmail.comwrote: From what you explained, it seems obvious that there exists some non-SIP device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, das sandesh sandesh...@gmail.com wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can any one share your thoughts on this, also asterisk version should not be a problem as we have other servers with same version and dtmf work good...Aslo since we also use g729 for some extensions we did not inband Also recently we got one more issue in this server, that as we talk on the phone randomly we get redial dtmf tones during the conversation, this suddenly started happening as this was good few months backI tried researching but could not find any ideas in regards to why this tones are coming into picture..I really appreciate if anyone can share their thoughts in regards to this.. Thank you very much Regards Sandesh -- _ -- 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] Voice prompts
On 07/19/2010 01:23 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Sent: Monday, July 19, 2010 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voice prompts Have now installed a swedish prompt set In /var/lib/asterisk/sounds/se I run elastix And set Language=se in /etc/asterisk/sip.conf But not work \-\- \-\- Show your CLI output so we see that you are getting correct playback Here's a QD snippet to let you do a verification Exten = 1234,1,answer Exten = 1234,n,Set(CHANNEL(language)=se) Exten = 1234,n,playback(tt-monkeys) Exten = 1234,n,playback(vm-goodbye) Exten = 1234,n,hangup Danny, When bottom posting, something you should keep in mind is that a -- on a line by itself causes most email clients to consider anything below it a signature (a sane client will lighten the text, and it won't appear when you hit reply). It would make things much nicer if you were to also remove that part of the signature on your replies. -- _ -- 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] Voice prompts
Trying my best here, don't want to start another TOP/Bottom flame war -- -- _ -- 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] Voice prompts
Thanks for the example But still english in the godbye message -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Danny Nicholas Skickat: den 19 juli 2010 20:24 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Re: [asterisk-users] Voice prompts -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Sent: Monday, July 19, 2010 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voice prompts Have now installed a swedish prompt set In /var/lib/asterisk/sounds/se I run elastix And set Language=se in /etc/asterisk/sip.conf But not work -- -- Show your CLI output so we see that you are getting correct playback Here's a QD snippet to let you do a verification Exten = 1234,1,answer Exten = 1234,n,Set(CHANNEL(language)=se) Exten = 1234,n,playback(tt-monkeys) Exten = 1234,n,playback(vm-goodbye) Exten = 1234,n,hangup _ -- 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] Voice prompts
One of two things is happening (In my opinion) 1. You aren't pointing to the right place Or 2. The language variable isn't getting set , so Asterisk gets lost. The CLI output with verbose 5 would tell you 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
Re: [asterisk-users] Voice prompts
Ok How to test on the cli As i say I running elastix and yes i know there a mailing list about elastix but the people there only point me to the book about elastix And i haven't adobe acrobat and have no plan to get it on this machine -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Danny Nicholas Skickat: den 19 juli 2010 21:08 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Re: [asterisk-users] Voice prompts One of two things is happening (In my opinion) 1. You aren't pointing to the right place Or 2. The language variable isn't getting set , so Asterisk gets lost. The CLI output with verbose 5 would tell you 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 -- _ -- 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] Voice prompts
Flash Operator Panel (2?) Is by best guess -- _ -- 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] Problems with Dahdi 2.3.0.1 trying to load OSLEC
Hello list, I'm facing a little issue with dahdi attempting to load the OSLEC echo canceller into my current kernel. After compiling dahdi 2.3.0.1 with OSLEC support, I get the following error when set 'oslec' as the echocanceller: DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) - Similar errors are *NOT* present using other echo canncelers. - I tried adding the 'dahdi_echocan_oslec' line to /etc/dahdi/modules and the error continues. I'm running Slackware Linux 13.0, Kernel 2.6.29.6-smp # dmesg ... dahdi_echocan_oslec: Unknown symbol oslec_create dahdi_echocan_oslec: Unknown symbol oslec_update dahdi_echocan_oslec: Unknown symbol oslec_free ... Could someone point me to some documentation about this incident? Regards, -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
On Monday, July 19, 2010 01:03:57 am Peter Childs wrote: One of the problems with Distinctive Ring tones is that its not consistent, between different phones so if you have a mix of phone types you have a problem. Agreed. I only mentioned what I did since I, along with the OP use Aastra phones. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] T.30 fax receiving problem with app_fax
I tried other fax machine and fax succesfully received. -- _ -- 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
