Re: [asterisk-users] Generating a different countries ringtone on a per call basis
On 26/09/13 16:43, Rusty Newton wrote: On Thu, Sep 26, 2013 at 10:08 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 26/09/13 14:59, Rusty Newton wrote: Try the following: extension = 6001,1,Set(CHANNEL(tonezone)=us) same = n,Dial(SIP/6001,,r(ring)) The argument passed to the r option should be the specific tone in the category of the tonezone you are setting. Thanks. I did try that as pretty much the first thing I tried but it continued to play the UK ring tone. Its not a big issue as we can work around it by playing music on hold instead which is a recording of the required ring tone. Having asterisk generate it just seemed the neater option. Are you sure you specified an argument to the 'r' option? Or did you just try 'r' without an argument? For me.. if I specify a uk tonezone, to get it playing uk tones I have to specify an argument to the 'r' option. If I try just 'r' by itself then I get US tones. You would think, that without specifying an argument, it should default to the tonezone in use on the channel. That may be a bug or oversight. What version of Asterisk were you using, and what channel type? That could well be it. It would have been with the standard 'r' option and not 'r(ring)' as thats the way our feature is currently programmed. I have just tested it with r(ring) and that works. 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] Queue Management
This should happen automatically - not sure what you want to do. l. 2013/9/26 akhilesh chand omakhileshch...@gmail.com: Dear All, I have six different campaign and 5 different agent have login on that campaign.Same thing i have done using agi and database,i never use queue management on this scenario. Agent can also shuffling one campaign to anther campaign. Now i want to do some work with queue.I want to use single queue to managing this. Eg: campaign Agent Login A a_1,a_3 (In campaign A 2 agents are login) B a_2,a_1 (In campaign B 2 agents are login) C a_3,a_1,a_4 (In campaign C 3 agents are login) D a_4,a_5,a_3 (In campaign D 3 agents are login) E a_1,a_3,1_2 (In campaign E 3 agents are login) Fa_5,a_4 (In campaign F 2 agents are login) When a call come to campaign A that call goes to agent a_1 or a_3 not goes to other campaigns agents. Regards Akhilesh -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Problems sending log to rsyslog
On Thu, Sep 26, 2013 at 11:16 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 26/09/13 15:25, Mauricio Tavares wrote: So I have asterisk 1.8.23 and want to send my logs to rsyslog. I tell asterisk to use syslog in addition to messages: root@voip:~# tail -10 /etc/asterisk/logger.conf ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose,dtmf,fax ;syslog keyword : This special keyword logs to syslog facility ; syslog.local0 = notice,warning,error ; root@voip:~# After reloading (asterisk -rx 'logger reload') the logger, it seems that Asterisk is happy: root@voip:~# asterisk -rx 'logger show channels' Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR /var/log/asterisk/messages File Enabled- NOTICE WARNING ERROR Console Enabled- NOTICE WARNING ERROR root@voip:~# So I set rsyslog: root@voip:~# fgrep asterisk /etc/rsyslog.d/50-default.conf local0.* /var/log/asterisk/messages.log root@voip:~# and restart it. And then check the asterisk log directory: root@voip:~# ls -lh /var/log/asterisk/ total 3.7M drwxr-xr-x 2 asterisk asterisk 4.0K Jul 22 20:57 cdr-csv drwxr-xr-x 2 asterisk asterisk 4.0K Jun 28 14:16 cdr-custom -rw-rw 1 asterisk asterisk 252K Sep 26 09:37 messages -rw-rw 1 asterisk asterisk 248K Sep 22 05:14 messages.1 -rw-r- 1 syslog adm 0 Sep 26 06:47 messages.log -rw-rw 1 asterisk asterisk 118 Sep 26 10:07 queue_log root@voip:~# It does not seem like much is being written to messages.log compared to messages. Anything I missed? Have you checked the /var/log/asterisk directory permissions? I dont know how rsyslog is setup on your system but its possible it gets started as root, sees the destination file doesnt exist so creates it and sets the file permissions, and then drops down to running as the syslog user. At this point it doesnt have write permission to the /var/log/asterisk directory so cannot append to the file. And you were absolutely right: root@voip:~# sudo -u syslog touch /var/log/asterisk/my_nose touch: cannot touch `/var/log/asterisk/my_nose': Permission denied root@voip:~# ls -lhd /var/log/asterisk drwxr-xr-x 4 asterisk asterisk 4.0K Sep 26 10:10 /var/log/asterisk root@voip:~# getent group asterisk asterisk:x:114:www-data root@voip:~# So, I decided to be lazy and add syslog to the asterisk group: root@voip:~# id syslog uid=101(syslog) gid=103(syslog) groups=114(asterisk),103(syslog) root@voip:~# chmod g+w /var/log/asterisk root@voip:~# sudo -u syslog touch /var/log/asterisk/my_nose root@voip:~# 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 -- _ -- 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] Is this SDP payload Asterisk created valid?
We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488 response from the far end. This packet seems to be updating the version for the o= session id which is fair enough. Its sending the c= connection data but not the m=audio line which appears to be what the remote end is complaining about. Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? Thanks Gareth U 2013/09/27 11:04:55.352854 88.x.x.25:5060 - 213.x.x.24:5060 INVITE sip:0844xx@146.x.x.10:54900 SIP/2.0. Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713. Route:sip:213.x.x.24;lr=on;ftag=as691af817;did=ecd.c2dc96e6. Max-Forwards: 70. From:sip:01628xx@88.x.x.25;tag=as691af817. To:sip:0844xxx...@freespeech.co.uk;tag=ee7a6c7cad57f096i1. Contact:sip:01628xx@88.x.x.25:5060. Call-ID: 2eeb643d086234de59a1fd4e78170d3f@88.x.x.25:5060. CSeq: 104 INVITE. User-Agent: Asterisk PBX 11.2-cert2. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 110. . v=0. o=root 716216031 716216033 IN IP4 88.x.x.35. s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.35. t=0 0. # U 2013/09/27 11:04:55.365458 213.x.x.24:5060 - 88.x.x.25:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713;rport=5060. From:sip:01628xx@88.x.x.25;tag=as691af817. To:sip:0844xxx...@freespeech.co.uk;tag=ee7a6c7cad57f096i1. Call-ID: 2eeb643d086234de59a1fd4e78170d3f@88.x.x.25:5060. CSeq: 104 INVITE. Server: OpenSIPS (1.5.3-notls (x86_64/linux)). Content-Length: 0. . # U 2013/09/27 11:04:55.431674 213.x.x.24:5060 - 88.x.x.25:5060 SIP/2.0 488 Not Acceptable Here. To:sip:0844xxx...@freespeech.co.uk;tag=ee7a6c7cad57f096i1. From:sip:01628xx@88.x.x.25;tag=as691af817. Call-ID: 2eeb643d086234de59a1fd4e78170d3f@88.x.x.25:5060. CSeq: 104 INVITE. Via: SIP/2.0/UDP 88.x.x.25:5060;rport=5060;received=88.x.x.25;branch=z9hG4bK62215713. Record-Route:sip:213.x.x.24;lr=on;ftag=as691af817. Contact: freespeechsip:0844xx@146.x.x.10:54900. Warning: 304 spa Media type not available. Server: Cisco/SPA303-7.5.4. Content-Length: 0. -- _ -- 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] Is this SDP payload Asterisk created valid?
On 27/09/13 14:15, Gareth Blades wrote: Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? Reading RFC2327 it cais the c= line 'must' be present in all updates while 'm=' media lines are optional. I am therefore inclined to believe that Asterisk is working correctly and there is a bug in the customers SIP equipment. But thats just my personal interpretation from briefly reading the standard. -- _ -- 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] Is this SDP payload Asterisk created valid?
Gareth Blades wrote: We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488 response from the far end. This packet seems to be updating the version for the o= session id which is fair enough. Its sending the c= connection data but not the m=audio line which appears to be what the remote end is complaining about. Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? The SDP you posted should be fine BUT my question becomes... have you modified chan_sip at all? I don't think it should be possible for it to not put any media lines in... Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to bind to ipv4 ipv6
Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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] Is this SDP payload Asterisk created valid?
On 27/09/13 14:36, Joshua Colp wrote: Gareth Blades wrote: We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488 response from the far end. This packet seems to be updating the version for the o= session id which is fair enough. Its sending the c= connection data but not the m=audio line which appears to be what the remote end is complaining about. Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? The SDP you posted should be fine BUT my question becomes... have you modified chan_sip at all? I don't think it should be possible for it to not put any media lines in... Cheers, No we havent made any changes to chan_sip. The servers were a fresh install a short while ago straight to 11.2-cert1 as we wanted a later kernel version to make use of the new timing source it provides. We then upgraded to cert2 after it was released. The only thing we have changed is the setting of a DYNAMIC_FEATURES variable which was stopping remote bridging from being performed which is probably what has highlighted this fault. 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
Re: [asterisk-users] How to bind to ipv4 ipv6
Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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] Is this SDP payload Asterisk created valid?
On 27/09/13 14:28, Gareth Blades wrote: On 27/09/13 14:15, Gareth Blades wrote: Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? Reading RFC2327 it cais the c= line 'must' be present in all updates while 'm=' media lines are optional. I am therefore inclined to believe that Asterisk is working correctly and there is a bug in the customers SIP equipment. But thats just my personal interpretation from briefly reading the standard. Well looking at RFC4566 section 5 :- Some lines in each description are REQUIRED and some are OPTIONAL, but all MUST appear in exactly the order given here (the fixed order greatly enhances error detection and allows for a simple parser). OPTIONAL items are marked with a *. Session description v= (protocol version) o= (originator and session identifier) s= (session name) i=* (session information) u=* (URI of description) e=* (email address) p=* (phone number) c=* (connection information -- not required if included in all media) b=* (zero or more bandwidth information lines) One or more time descriptions (t= and r= lines; see below) z=* (time zone adjustments) k=* (encryption key) a=* (zero or more session attribute lines) Zero or more media descriptions Time description t= (time the session is active) r=* (zero or more repeat times) Media description, if present m= (media name and transport address) i=* (media title) c=* (connection information -- optional if included at session level) b=* (zero or more bandwidth information lines) k=* (encryption key) a=* (zero or more media attribute lines) So if I am reading that correctly the m= line is required only if we include a media description entry. It basically sais its required if we decide to include it (yes I know that doesnt make sense). What I presume this means is that if a media description is included such as there being a 'a=' line then the 'm=' line then becomes required. So it still sounds like Asterisk is behaving correctly. -- _ -- 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] Generating a different countries ringtone on a per call basis
On Fri, Sep 27, 2013 at 3:20 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 26/09/13 16:43, Rusty Newton wrote: On Thu, Sep 26, 2013 at 10:08 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 26/09/13 14:59, Rusty Newton wrote: Try the following: extension = 6001,1,Set(CHANNEL(tonezone)=us) same = n,Dial(SIP/6001,,r(ring)) The argument passed to the r option should be the specific tone in the category of the tonezone you are setting. Thanks. I did try that as pretty much the first thing I tried but it continued to play the UK ring tone. Its not a big issue as we can work around it by playing music on hold instead which is a recording of the required ring tone. Having asterisk generate it just seemed the neater option. Are you sure you specified an argument to the 'r' option? Or did you just try 'r' without an argument? For me.. if I specify a uk tonezone, to get it playing uk tones I have to specify an argument to the 'r' option. If I try just 'r' by itself then I get US tones. You would think, that without specifying an argument, it should default to the tonezone in use on the channel. That may be a bug or oversight. What version of Asterisk were you using, and what channel type? That could well be it. It would have been with the standard 'r' option and not 'r(ring)' as thats the way our feature is currently programmed. I have just tested it with r(ring) and that works. Thanks Woot. Glad its working at least. I'll file an issue on the tracker, as it is fairly non-intuitive that without specifying an argument the 'r' option won't use indications from the tonezone being used by the channel. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] user list archive
What's up with the user list archive? It hasn't been updated since the 23rd. -- _ -- 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] How to bind to ipv4 ipv6
Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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] How to bind to ipv4 ipv6
http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.html Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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] user list archive
On Fri, Sep 27, 2013 at 10:20 AM, Paul Albrecht palbre...@glccom.com wrote: What's up with the user list archive? It hasn't been updated since the 23rd. We did some Mailman maintenance on the 23rd, some configuration may have been goofed up. We'll look into it. Thanks! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] user list archive
Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 06:10 PM, Rusty Newton wrote: On Fri, Sep 27, 2013 at 10:20 AM, Paul Albrecht palbre...@glccom.com wrote: What's up with the user list archive? It hasn't been updated since the 23rd. We did some Mailman maintenance on the 23rd, some configuration may have been goofed up. We'll look into it. 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] user list archive
On Fri, Sep 27, 2013 at 11:14 AM, Daniel van den Berg aster...@suretel.co.za wrote: Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. You responded to the wrong thread. oops! :) -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] user list archive
Sorry :) On 09/27/2013 06:16 PM, Rusty Newton wrote: On Fri, Sep 27, 2013 at 11:14 AM, Daniel van den Berg aster...@suretel.co.za wrote: Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. You responded to the wrong thread. oops! :) -- _ -- 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] How to bind to ipv4 ipv6
Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 05:59 PM, Johan Wilfer wrote: http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.html Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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] How to bind to ipv4 ipv6
From sip.conf.sample included in your Asterisk source tree. See item c) and the Note: ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for ; udpbindaddr, tcpbindaddr, and tlsbindaddr.) ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) ; ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 ; for TLS). ; IPv4 example: bindaddr=0.0.0.0:5062 ; IPv6 example: bindaddr=[::]:5062 ; ; The address family of the bound UDP address is used to determine how Asterisk performs ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only ; records are considered. In case d), both A and records are considered. Note, ; however, that Asterisk ignores all records except the first one. In case d), when both A ; and records are available, either an A or record will be first, and which one ; depends on the operating system. On systems using glibc, records are given ; priority. udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel van den Berg Sent: Friday, September 27, 2013 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to bind to ipv4 ipv6 Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 05:59 PM, Johan Wilfer wrote: http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.htm l Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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] How to bind to ipv4 ipv6
This means i can't use IPv4 and IPv6 together. Right? El 27/09/2013 11:25, Eric Wieling escribió: From sip.conf.sample included in your Asterisk source tree. See item c) and the Note: ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for ; udpbindaddr, tcpbindaddr, and tlsbindaddr.) ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) ; ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 ; for TLS). ; IPv4 example: bindaddr=0.0.0.0:5062 ; IPv6 example: bindaddr=[::]:5062 ; ; The address family of the bound UDP address is used to determine how Asterisk performs ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only ; records are considered. In case d), both A and records are considered. Note, ; however, that Asterisk ignores all records except the first one. In case d), when both A ; and records are available, either an A or record will be first, and which one ; depends on the operating system. On systems using glibc, records are given ; priority. udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel van den Berg Sent: Friday, September 27, 2013 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to bind to ipv4 ipv6 Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 05:59 PM, Johan Wilfer wrote: http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.htm l Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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] How to bind to ipv4 ipv6
Sorry, I meant item d) which says and I quote Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: and Note that using bindaddr=:: will show only a single IPv6 socket in netstat. . IPv4 is supported at the same time using IPv4-mapped IPv6 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko Sent: Friday, September 27, 2013 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to bind to ipv4 ipv6 This means i can't use IPv4 and IPv6 together. Right? El 27/09/2013 11:25, Eric Wieling escribió: From sip.conf.sample included in your Asterisk source tree. See item c) and the Note: ; With the current situation, you can do one of four things: ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 ; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for ; udpbindaddr, tcpbindaddr, and tlsbindaddr.) ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) ; ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 ; for TLS). ; IPv4 example: bindaddr=0.0.0.0:5062 ; IPv6 example: bindaddr=[::]:5062 ; ; The address family of the bound UDP address is used to determine how Asterisk performs ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only ; records are considered. In case d), both A and records are considered. Note, ; however, that Asterisk ignores all records except the first one. In case d), when both A ; and records are available, either an A or record will be first, and which one ; depends on the operating system. On systems using glibc, records are given ; priority. udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel van den Berg Sent: Friday, September 27, 2013 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to bind to ipv4 ipv6 Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 05:59 PM, Johan Wilfer wrote: http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.ht m l Google :-) /J 2013-09-27 17:47, Daniel van den Berg skrev: Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za mailto:aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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:
[asterisk-users] Realtime Mysql
Hello, I am looking to know if it is possible to modify the SQL query that is on Realtime sip accounts. I want multiple servers use the same sql table, so getting an extra server field to indicate that the data is valid on the X server is this possible? thank you in advance jerome -- _ -- 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] Realtime Mysql
On 27/09/13 17:47, Phibee Network Operation Center wrote: Hello, I am looking to know if it is possible to modify the SQL query that is on Realtime sip accounts. I want multiple servers use the same sql table, so getting an extra server field to indicate that the data is valid on the X server is this possible? thank you in advance jerome I dont know but you should be able to at least create multiple mysql views with each one being referenced by the appropiate server. -- _ -- 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] Function CDR_PROP
In Asterisk 12, how should I call the function CDR_PROP set(CDR_PROP(disable)=true) or simply CDR_PROP(disable) I am getting two records per call attempt, and I cannot figure out how to go back to get only one record. So far I am using this technique, but it changes nothing. My calls always involve a single caller a single calee exten = 100,1,NoOp() same = n,Dial(SIP/bob,,b(default^callee_handler^1)) same = n,Hangup() exten = callee_handler,1,NoOp() same =n,Set(CDR_PROP(disable)=true) same =n,Return() Philip -- _ -- 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] iax: unable to transfer - one way audio
We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from zoiperipaddr: requested format = speex, requested prefs = (), actual format = ulaw, host prefs = (silk16|ulaw|gsm|g722), priority = mine -- Executing [8447@nz-in:1] Dial(IAX2/n4-270, IAX2/sydney) in new stack -- Called IAX2/sydney -- Call accepted by nyipaddr (format ulaw) -- Format for call is (ulaw) -- IAX2/sydney-8819 is ringing -- IAX2/sydney-8819 answered IAX2/n4-270 -- Channel 'IAX2/n4-270' unable to transfer -- Channel 'IAX2/sydney-8819' unable to transfer -- Channel 'IAX2/sydney-8819' unable to transfer -- Channel 'IAX2/sydney-8819' unable to transfer The NY server: -- Accepting AUTHENTICATED call from sydneyipaddr: -- requested format = ulaw, -- requested prefs = (ulaw|silk16|gsm|g722), -- actual format = ulaw, -- host prefs = (ulaw|gsm|g722), -- priority = mine -- Executing [s@incoming-nz:1] Goto(IAX2/home-2152, incoming,s,nz-in) in new stack -- Goto (incoming,s,5) -- Executing [s@incoming:5] Dial(IAX2/home-2152, DAHDI/g0SIP/250SIP/251,60,tT) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called DAHDI/g0 -- Called SIP/250 -- Called SIP/251 -- DAHDI/1-1 is ringing -- SIP/251-001d is ringing -- SIP/250-001c is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered IAX2/home-2152 -- Channel 'IAX2/home-2152' unable to transfer -- Hanging up on 'DAHDI/1-1' Any help appreciated. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
