[asterisk-users] dahdi + dlink du128ta
hello everyone, I do have a usb ISDN modem that I would like to make it work with dahdi. is it possible? I am running debian 7, with dahdi 2.9, asterisk 11.8 dahdi cannot find it at the moment, unless there is something else to be done. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi + dlink du128ta
I do have a usb ISDN modem that I would like to make it work with dahdi. is it possible? I am running debian 7, with dahdi 2.9, asterisk 11.8 dahdi cannot find it at the moment, unless there is something else to be done. I do not think so. If your very old ISDN modem uses a Cologne Chip (HFC), then there might be a very small chance to make it work with mISDN. jg -- _ -- 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] sorry for askingm but I can#r fund a solution [SOLVED]
On Thu, Mar 13, 2014 at 05:18:54PM -0500, Kevin Larsen wrote: I neither have a 2000 in sip,conf nor I want to have one. 2000 doesn't have an IP and I want to get rid of it, honestly. I'd really want to know, where this 2000 is burned in and how to erase it. sip show peers does'nt show a peer 2000 nor I have a user 2000. Something that lives at 10.0.1.4 thinks it is extension 2000 and is trying to register. Your problem isn't Asterisk per se, it is finding where that IP address is located and what the device is to either fix its configuration to what it should be or to take the rogue device off your network. I've got it! Thanks! At 10.0.1.4 a misconfigured linphone caused it! :-) ru :-) -- _ -- 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] sorry for askingm but I can#r fund a solution [SOLVED]
On Fri, Mar 14, 2014 at 07:44:53AM +0200, Vadim Lungu wrote: This is IP is brute forcing you with peer that don't exist. So just block it. iptalble -A INPUT -s 10.0.1.4 -p udp -dport 5060 -j DROP As I came to machine 10.0.1.4 I could make out a misconfigured linphone. It was used to get connected to another asterisk before. Thanks Vadim, greeting to Moldavia -- _ -- 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] func_odbc do not read LIKE predicate
Thank you, this statement is working. On 03/14/2014 07:08 AM, hkc323 wrote: Vadim Lungu try this one . readsql=SELECT name FROM asterisk_sippeers WHERE name LIKE '%477'; -- Mit freundlichen Gren / Best regards Vadim Lungu System Engineer Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com Web https://www.yopeso.com YOPESO s.r.l. Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschftsfhrer Svetlana Arnaut -- _ -- 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 Installation
Hello Experts, I was trying to follow the instructions in the Installation of AsteriskNow 32 bit installation. I am not able to get the graphical interface as mentioned in the wiki. Also, after reboot, the DNS settings does not seem to be fine. Yum update seems to be throwing errors. Regards, Sujatha -- _ -- 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] res-odbc sanity check reconnecting
Hello everyone, I would appreciate if someone could help me with the following issue: http://pastebin.com/bTskMLVw My res_odbc.conf file look as follows: http://pastebin.com/bhReQkXQ -- Mit freundlichen Gren / Best regards Vadim Lungu System Engineer Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com Web https://www.yopeso.com YOPESO s.r.l. Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschftsfhrer Svetlana Arnaut -- _ -- 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] res-odbc sanity check reconnecting
I would appreciate if someone could help me with the following issue: http://pastebin.com/bTskMLVw My res_odbc.conf file look as follows: http://pastebin.com/bhReQkXQ Nothing to really worry about. The ODBC driver automatically reconnects to MySQL as the system already told you. jg -- _ -- 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] sipML5, Ast12 and WebRTC: not acceptable here
Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call from the softphone, Asterisk answers with a 488 not acceptable here. I'm probably missing something but I'm not able to find what and where. Is there someone able to point me to the right direction? Below is my configuration. The sofpthone is registered as 1060. Thanks in advance. Marco Signorini. pjsip.conf: [transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/etc/asterisk/sslcert.pem method=tlsv1 [1060] type=endpoint transport=transport-tls context=from-internal use_avpf=yes media_encryption=sdes disallow=all allow=alaw allow=ulaw aors=1060 auth=1060 [1060] type=auth auth_type=userpass password=1060 username=1060 [1060] type=aor max_contacts=10 [204] http.conf: enabled=yes bindaddr=10.10.5.49 bindport=8088 CLI pjsip show endpoints Endpoint: 1060 Not in use0 of inf InAuth: 1060/1060 Aor: 1060 10 Contact: 1060/sip:1060@10.10.5.106:54083;transport=ws;rt Unknown nan Transport: transport-tls tls 0 0 0.0.0.0:5061 Endpoint: 204 Not in use0 of inf InAuth: 204/204 Aor: 2041 Contact: 204/sip:204@10.10.5.120:5066;transport=udp Unknown nan Transport: transport-udp udp 0 0 0.0.0.0:5060 *CLI pjsip show transport transport-tls Transport: TransportId Type cos tos BindAddress = Transport: transport-tls tls 0 0 0.0.0.0:5061 ParameterName : ParameterValue == async_operations : 1 bind : 0.0.0.0:5061 ca_list_file : cert_file : /etc/asterisk/sslcert.pem cipher : cos: 0 domain : external_media_address : external_signaling_address : external_signaling_port: 0 local_net : method : tlsv1 password : priv_key_file : protocol : tls require_client_cert: No tos: CS0 verify_client : No verify_server : No And this is the relevant SIP data exchange (with public IP hidden): *CLI --- Received SIP request (2420 bytes) from WS:10.10.5.106:54411 --- INVITE sip:204@10.10.5.49 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiBw81ooU7ybSRbRqr8TOqWkMPQRdkMXo;rport From: John Doe (101)sip:1060@10.10.5.49;tag=heMv1HvlT7DeQxPxuqcq To: sip:204@10.10.5.49 Contact: John Doe (101)sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1060;ha1=0b2413e6f3c96a0517b4413a6f6ce7ae;+g.oma.sip-im;+sip.ice;language=en,fr,it Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc CSeq: 38718 INVITE Content-Type: application/sdp Content-Length: 1827 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5 v=0 o=- 365893986064703740 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 m=audio 37874 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 85.0.XXX.XXX a=rtcp:37874 IN IP4 85.0.XXX.XXX a=candidate:296123718 1 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:296123718 2 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:3103388307 1 udp 1845501695 85.0.XXX.XXX 37874 typ srflx raddr 10.10.5.106 rport 63858 generation 0 a=candidate:3103388307 2 udp 1845501695 85.0.XXX.XXX 37874 typ srflx raddr 10.10.5.106 rport 63858 generation 0 a=candidate:1596293558 1 tcp 1509957375 10.10.5.106 0 typ host generation 0 a=candidate:1596293558 2 tcp 1509957375 10.10.5.106 0 typ host generation 0 a=ice-ufrag:l8AWdK4ft+AnAYGl a=ice-pwd:3tLKvT97tf0GQr+e8v8bKncd a=ice-options:google-ice a=fingerprint:sha-256 89:D4:EB:6E:9C:41:36:03:A1:44:CD:A2:08:78:CD:86:FE:EC:30:09:53:0F:77:CE:BA:8E:DE:8C:1B:A1:41:10 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0
Re: [asterisk-users] res-odbc sanity check reconnecting
I'm getting a lot of errors in CLI and this become annoying. And when I use database from localhost, I don't have such an issue. So I don't understand how to identify the reason of this problem. On 03/14/2014 01:09 PM, jg wrote: Nothing to really worry about. The ODBC driver automatically reconnects to MySQL as the system already told you -- Mit freundlichen Gren / Best regards Vadim Lungu System Engineer Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com Web https://www.yopeso.com YOPESO s.r.l. Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschftsfhrer Svetlana Arnaut -- _ -- 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] res-odbc sanity check reconnecting
I'm getting a lot of errors in CLI and this become annoying. And when I use database from localhost, I don't have such an issue. So I don't understand how to identify the reason of this problem. First, please do not add your signature in the current form to your posts. It is irrelevant here and it takes too much time to delete them... You did not mention that the db is running on a different host. You could publish your odbc.ini, but essentially you need to find out whether there is a problem with your LAN or the way you have configured things. Start a pcap trace and look for delays related to dns, etc... It is likely that your problem is not related to Asterisk. jg -- _ -- 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] Replying to Posts
On Thu, March 13, 2014 15:32, Kevin Larsen wrote: On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? +1-1 = 0 I do not care about where people put their replies so long as I can figure out who is answering what. What I do not like to read is this interminable religious dogma about the 'natural' order of writing. This is the second or third list this week in which this B.S. has shown up in my inbox. In written business communication, in contrast to tech-speak customarily found on mailing lists, ones answer always goes before any quoted context. Not because it has to, it is just that I have seldom, if ever, seen it done any other way. And regular business communication with non-technical folk comprises well over 75% of my daily written communication. And while I understand the cultural motivation behind the dogma of bottom posting I remain sceptical respecting its utility. Is there any objective evidence whatsoever that top or bottom posting makes any difference to the reader's understanding of the message? Does any rigorously determined data exist to support that contention? If not then this is simply a matter of trying to impose a set of arbitrary cultural values cloaked in the guise of technical superiority. Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus Notes actually handles it better as there is a Reply option for normal email and a Reply With Internet-Style History that I use for this list. I don't have any problems following the rules of the list, but I am fully on the side of the Replies should go at the top group and would vote for a change in the rules. And do not even start on the Chevy vs. Ford debate respecting the technical superiority of Pine over Outlook. GAWD... Life its too short as it is. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Replying to Posts
If the sequence was really important, we would just move this whole list to a LinkedIn group and have a much better environment for following threads and managing the profiles of the group's members. This is a pretty old system of managing peer support but it always takes a while to get legacy systems replaced. Ron On 14/03/2014 9:52 AM, James B. Byrne wrote: On Thu, March 13, 2014 15:32, Kevin Larsen wrote: On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? +1-1 = 0 I do not care about where people put their replies so long as I can figure out who is answering what. What I do not like to read is this interminable religious dogma about the 'natural' order of writing. This is the second or third list this week in which this B.S. has shown up in my inbox. In written business communication, in contrast to tech-speak customarily found on mailing lists, ones answer always goes before any quoted context. Not because it has to, it is just that I have seldom, if ever, seen it done any other way. And regular business communication with non-technical folk comprises well over 75% of my daily written communication. And while I understand the cultural motivation behind the dogma of bottom posting I remain sceptical respecting its utility. Is there any objective evidence whatsoever that top or bottom posting makes any difference to the reader's understanding of the message? Does any rigorously determined data exist to support that contention? If not then this is simply a matter of trying to impose a set of arbitrary cultural values cloaked in the guise of technical superiority. Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus Notes actually handles it better as there is a Reply option for normal email and a Reply With Internet-Style History that I use for this list. I don't have any problems following the rules of the list, but I am fully on the side of the Replies should go at the top group and would vote for a change in the rules. And do not even start on the Chevy vs. Ford debate respecting the technical superiority of Pine over Outlook. GAWD... Life its too short as it is. -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Working Config for Polycom VVX and Auto Answer
Hi -
Just wondering if anyone has gotten a Polycom VVX phone to
successfully do an Auto Answer with asterisk. I have an older
generation of Polycom phones that do this just fine, but I can't seem
to make the VVX phones work.
I tried the guide here:
http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167
And I have this in my diaplan:
exten = _8XX,1,SIPAddHeader(Alert-Info: info=ringAutoAnswer)
exten = _8XX,2,Dial(SIP/${EXTEN:1},20,tk)
But whenever I attempt a call to a matching exten, it just rings
normally for the 20 seconds I have indicated here and never answers.
I found a setting in the phone's GUI: Auto Answer SIP Calls. When I
set this to Yes. It will auto answer, but it auto answers ALL
calls, not just ones with the Alert-Info header set.
Any guidance is appreciated.
Thanks!
Noah
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[asterisk-users] detail records graphic
hello everyone. i am looking for a way to present my CDRs in a nice way such as https://code.google.com/p/asterisk-cdr-viewer/ , but i do not want to run any databases such as mysql. is there a software (free) that could get the asterisk's csv and present it in a nice way? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Working Config for Polycom VVX and Auto Answer
On Fri, Mar 14, 2014 at 12:36 PM, Noah Miller noahisaacmil...@gmail.com wrote:
Hi -
Just wondering if anyone has gotten a Polycom VVX phone to
successfully do an Auto Answer with asterisk. I have an older
generation of Polycom phones that do this just fine, but I can't seem
to make the VVX phones work.
I tried the guide here:
http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167
And I have this in my diaplan:
exten = _8XX,1,SIPAddHeader(Alert-Info: info=ringAutoAnswer)
exten = _8XX,2,Dial(SIP/${EXTEN:1},20,tk)
But whenever I attempt a call to a matching exten, it just rings
normally for the 20 seconds I have indicated here and never answers.
I found a setting in the phone's GUI: Auto Answer SIP Calls. When I
set this to Yes. It will auto answer, but it auto answers ALL
calls, not just ones with the Alert-Info header set.
Any guidance is appreciated.
Thanks!
Noah
Well, in case anyone else is interested, it's working now. I must
have mistyped something the first time around because it is now
working with the exact settings I describe above.
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