Re: [asterisk-users] Automatic Number Identification and anonymous calls
2012/2/9, Maximilian Grobecker m.grobec...@portunity.de: Hello, I know about the german phone system that the sense of an anonymous call is, that the called party has no way to get the caller's number in any way. The last switch honours the anonymous bit and removes the phone numbers before sending the call to the called party. In EURO-ISDN you have a feature called CLIRO which means Calling Line Identification Restriction Override and is only availiable to emergency services or special governmental services. With this line feature the last switch ignores the anonymous bit and you can see both caller id and ANI. The US phone system may work in another way, but if I call another person and want to hide my number I would appreciate if my phone number is not sent to the called party in any field. So this issue sounds to me more like feature than a bug ;-) Yes, I agree but what I rate as a bug is asterisk copying CID data into ANI field. The reasoning is asterisk should don't modify, unless explicitely configured to do so, whatever comes from the public network side. For instance, it seems analog lines cannot support ANI. So when receiving a call through an analog line, this ANI field should be simply empty. It seems that today default is to copy CID into ANI field. This is misleading as you may think this ANI is somehow more trustable than CID alone. I agree that if a caller requires anonymity, then, except for very special cases (emergency services, for instance), ANI should also be hidden by the telco. Greetings from Wuppertal Max Grobecker Am 09.02.2012 11:12, schrieb Olivier: 2012/2/8, Kevin P. Fleming kpflem...@digium.com: On 02/08/2012 12:40 PM, Olivier wrote: 2012/2/8, Kevin P. Flemingkpflem...@digium.com: On 02/08/2012 10:06 AM, Carlos Alvarez wrote: On Wed, Feb 8, 2012 at 2:35 AM, Olivieroza_4...@yahoo.fr mailto:oza_4...@yahoo.fr wrote: I always thought that ANI (Automatic Number Identification) could not directly be set or changed by end users. In this experiment, it seems that if an end user calls anonymously, the ANI is also hidden to the receiving party. I never work with analog lines, but I do believe that anything less than a PRI would have the blocking done at the telco end before you get it. Mostly correct; ANI is only delivered over ISDN, SS7 and *some* EM trunk circuits. Analog circuits have no ability to transport ANI. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for telling ANI feature can't come along analog lines. I did another test towards an ISDN BRI line and got the same results : ANI and CID are either both displayed or both hidden at the same time. You need to talk to your provider to see if your circuit is provisioned to send you ANI at all; if it's not sent by the provider, Asterisk just duplicates the CLID into the ANI channel variable. Then, may I say that I would prefer to have the ANI left blank, if the network doesn't provide this data, and let system administrators know about this. Most end-customer (retail) ISDN circuits are not configured to send ANI unless the customer asks for it. This matches what I had in mind : ANI being something dedicated to emergency services that may also be available to others requiring it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --
Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.
Hi To the best of my understanding this is the correct behaviour. When you add a peer to the database and a device configured for that peer registers, it enters that peer into the RealTime cache. When you do a sip reload you fully clear that RealTime cache so the asterisk process will lose knowledge of that peer until it registers again and gets re entered into the RealTime cache, which most SIP phones are set to do after a number of minutes. The real question here is, if you are using RealTime architecture for your peers, why are you doing a sip reload? On Fri, 2012-02-10 at 10:55 +0530, DHAVAL INDRODIYA wrote: nobody facing any issue with this or nobody using real time architecture On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi Group. I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing registration it registered fine on asterisk as peer is available in Database. But now i am doing 'sip reload' or 'reload' due to some reason my peer registration is going out and i cannot able to call that peer even though in SIP client it shows me 'registered'. Can any body elaborate on this issue which settings i need to put in sip.conf. I also tried to follow this patch https://issues.asterisk.org/view.php?id=14196 But it allready applied in code base so why it wont work? Here is my sip.conf settings. [general] context=from-internal; Default context for incoming cal rtcachefriends=no rtupdate=yes rtautoclear=yes rtsavesysname=yes callcounter = yes callevents=yes bindport=5060; UDP Port to bind to (SIP standard port is 5060) srvlookup=yes; Enable DNS SRV lookups on outbound calls pedantic=yes; Enable slow, pedantic checking for Pingtel tos=184; Set IP QoS to either a keyword or numeric val tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpiry=3600; Max length of incoming registration we allow defaultexpiry=120; Default length of incoming/outoing registration preferred_codec_only=yes disallow=all; First disallow all codecs allow=ulaw; Allow codecs in order of preference allow=alaw insecure=invite language=en ; Default language setting for all users/peers rtpholdtimeout=300; Terminate call if 300 seconds of no RTP activity useragent=dhaval ; Allows you to change the user agent string dtmfmode = rfc2833; Set default dtmfmode for sending DTMF. Default: rfc2833 qualify=yes nat=yes ;canreinvite=yes directmedia=yes directrtpsetup=yes And here is DB fields snapshots. id: 1 name: 201 ipaddr: 172.18.100.243 port: 53624 regseconds: 1328716180 defaultuser: 201 fullcontact: NULL regserver: dhaval useragent: CSipSimple r1133 / b lastms: 554 host: dynamic type: friend context: from-internal permit: NULL deny: NULL secret: 201 md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: yes nat: NULL allow: ulaw disallow: g729 insecure: invite callerid: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL Kindly help me to resolve this. Thanks Dhaval -- _ -- 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
Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?
Hi, I'm working on a small php program for just this. I guess from your question that you have Asterisk writing to a CDR database table, in which case you should be able to use my .php code fairly easily. It's nothing fancy but does give me a graphical presentation of calls/15minute segments. Attached is a screenshot of a graph, I have 1,5+ million entries in the table but there is no noticeable lag in refreshing the graph. At the moment it refreshes only when the button is pressed (the text is in Danish ...) but changing it to refresh automatically every 15 minutes wouldn't be a major problem. I'm working on adding the option of selecting date ranges, it's all still a work in progress! Regards Binni Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af asterisk jobs Sendt: 9. februar 2012 16:36 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? Hi everyone, I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would amazing to feed the asteriskcdrdb table to the program and get back the results without installing anything on the Asterisk server as I don't want to tamper with the server. Is there such a tool? Thanks, attachment: graph.gif-- _ -- 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 vs FreeSwitch vs Asterisk + OpenSIP
I hope I'm not flogging a dead horse here, but the discussion around the whole scalability issue in Asterisk have opened my eyes to a whole lot of issues, making me increasingly confused! We have a fully functioning and stable installation where we offer PBX services to some 15 small firms (basically medical practices). These are based all over the country, with between 2 and 15 SIP phones each. We have a Web front end where each firm can configure their own queues, menus, forwarding etc. My problem is that my bosses want to expand massively, they are currently talking of at least a tenfold increase in the number of clients. I'm fairly certain our Asterisk server won't be able to handle that. Our current 15 clients all have peak usage at the same time (with 2/3 of all traffic between 8 and 9 in the morning). At peak times, we have 20% CPU load with some 100 concurrent calls and a little under one call/second. I have to solve the scalability problem within a relatively short timeframe so starting from scratch with something new is out of the question. My first thought was to add another Asterisk server and use DUNDi load balancing between the two. But looking around and reading the discussion on this list got me to thinking whether some sort of SIP switch or router/proxy could take some load off the Asterisk server(s). One of my main concerns is to change our current setup as little as possible. It's a mishmash of Asterisk, MySQL, Rails and RAGI/RAMI. The original programmers are no longer available to me and I am still very wet behind the ears when it comes to VOIP. So should I be looking at adding e.g. OpenSIP as a sip proxy to our current setup or adding a second (and then a third and a fourth ...) Asterisk server with DUNDi? Or both? Will adding OpenSIP require a change in the way in which we handle SIP peers or require some major reconfiguration of Asterisk? It seems to me that DUNDi requires minimal configuration changes but I don't really know. Any information and recommendations will be greatly appreciated! Regards Binni -- _ -- 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 there a php script to analyse and show call detail reports from Asterisk CDR?
Hi, I forgot to add that you are free to use my code, I'll mail it later today. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Brynjolfur Thorvardsson Sendt: 10. februar 2012 09:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? Hi, I'm working on a small php program for just this. I guess from your question that you have Asterisk writing to a CDR database table, in which case you should be able to use my .php code fairly easily. It's nothing fancy but does give me a graphical presentation of calls/15minute segments. Attached is a screenshot of a graph, I have 1,5+ million entries in the table but there is no noticeable lag in refreshing the graph. At the moment it refreshes only when the button is pressed (the text is in Danish ...) but changing it to refresh automatically every 15 minutes wouldn't be a major problem. I'm working on adding the option of selecting date ranges, it's all still a work in progress! Regards Binni Fra: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af asterisk jobs Sendt: 9. februar 2012 16:36 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? Hi everyone, I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would amazing to feed the asteriskcdrdb table to the program and get back the results without installing anything on the Asterisk server as I don't want to tamper with the server. Is there such a tool? 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
[asterisk-users] DTMF forwarding and Page
Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the callees. I found option 'F' for MeetMe, but I have no control on Page(). TIA, Matteo -- _ -- 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] Garbled voicemail
Hello, this is a know problem when you are writing the voicemails over a nfs link. you have to start asterisk with the -t option to write voicemail records to the local /tmp and copy it to the final destination after it is finished. as far as i remember the first 10 seconds are ok and then the speed up started but with the -t option it was completly solved. best regards stefan Am 09.02.12 18:16, schrieb Ruben Rögels: Hi Dan, my wild speculation: It's some kind of timing/synchronisation problem. Do you use jitter buffer an/or echo cancelation? Best regards, Ruben -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Dan Ritter Gesendet: Donnerstag, 9. Februar 2012 17:33 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Garbled voicemail Our Asterisk system (1.8.8.1-1digium1~squeeze) has been very stable and generally doing a good job -- except that one day, voicemail recordings started being garbled. It only manifests when the VM comes from our telco gateway service -- OnSIP/Junction -- and not from internal phones or from an Asterisk box I have at home. We have voicemail set to record to WAV, and real files are being generated -- but it sounds incredibly sped up, faster than chipmunks. Completely unintelligible, even if you pull it into an audio editor and slow down playback. It is not perfectly consistent, but it happens in about 85% of voicemail recordings left from the outside world through OnSIP. We've had several years of trouble-free voicemail before this. Anyone seen anything similar? Advice? Wild speculation? -dsr- -- _ -- 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 -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Teamleiter VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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] dial plan with hangup cause 34
This is a FreePBX question as the Asterisk dialplan is managed by it.
I suggest to use 'extensions_override_freepbx.conf' (details in
extensions.conf) and place there your modified [macro-dialout-trunk].
HTH,
Ioan
On Fri, Feb 10, 2012 at 1:13 PM, ing.Achim Alexandru
alexandru.achi...@gmail.com wrote:
Dear Asterisk Users,
I have a question. I use asterisk 1.6 withh freepbx on ubuntu ,
compiled manually.
I want to change the route congestion message ( all-circuit-bussy)
wiyh a hangup cause 34 ( something like that in dialplan
s,n,GotoIf($[${HANGUPCAUSE} = 34]?failover,1). Have any ideas?
Thanks
Alexandru Achim
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Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.
I see this on some peers every time I do a sip reload and I am not using real-time. I use qualify and every time a sip reload occurs the device goes unreachable. I have shortend the register time to 5 min so the device comes back with-in about two min but it is very annonying to me and my user. I have tracked my issue back to cusomters using netgear routers. If they replace the device the issue goes away. On netgear routers we have found we have to shut of SIP AGL to get them to register right but this quark won't go away. Maybe your issue is endpoint releated as well? Bryant From: DHAVAL INDRODIYA dhaval.it01...@gmail.com Sent: Friday, February 10, 2012 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug. nobody facing any issue with this or nobody using real time architecture On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi Group. I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing registration it registered fine on asterisk as peer is available in Database. But now i am doing 'sip reload' or 'reload' due to some reason my peer registration is going out and i cannot able to call that peer even though in SIP client it shows me 'registered'. Can any body elaborate on this issue which settings i need to put in sip.conf. I also tried to follow this patch https://issues.asterisk.org/view.php?id=14196 But it allready applied in code base so why it wont work? Here is my sip.conf settings. [general] context=from-internal; Default context for incoming cal rtcachefriends=no rtupdate=yes rtautoclear=yes rtsavesysname=yes callcounter = yes callevents=yes bindport=5060; UDP Port to bind to (SIP standard port is 5060) srvlookup=yes; Enable DNS SRV lookups on outbound calls pedantic=yes; Enable slow, pedantic checking for Pingtel tos=184; Set IP QoS to either a keyword or numeric val tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpiry=3600; Max length of incoming registration we allow defaultexpiry=120; Default length of incoming/outoing registration preferred_codec_only=yes disallow=all; First disallow all codecs allow=ulaw; Allow codecs in order of preference allow=alaw insecure=invite language=en ; Default language setting for all users/peers rtpholdtimeout=300; Terminate call if 300 seconds of no RTP activity useragent=dhaval ; Allows you to change the user agent string dtmfmode = rfc2833; Set default dtmfmode for sending DTMF. Default: rfc2833 qualify=yes nat=yes ;canreinvite=yes directmedia=yes directrtpsetup=yes And here is DB fields snapshots. id: 1 name: 201 ipaddr: 172.18.100.243 port: 53624 regseconds: 1328716180 defaultuser: 201 fullcontact: NULL regserver: dhaval useragent: CSipSimple r1133 / b lastms: 554 host: dynamic type: friend context: from-internal permit: NULL deny: NULL secret: 201 md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: yes nat: NULL allow: ulaw disallow: g729 insecure: invite callerid: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL Kindly help me to resolve this. Thanks Dhaval -- _ -- 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 vs FreeSwitch vs Asterisk + OpenSIP
2012/2/10 Brynjolfur Thorvardsson bi...@itanet.nu: I hope I'm not flogging a dead horse here, but the discussion around the whole scalability issue in Asterisk have opened my eyes to a whole lot of issues, making me increasingly confused! We have a fully functioning and stable installation where we offer PBX services to some 15 small firms (basically medical practices). These are based all over the country, with between 2 and 15 SIP phones each. We have a Web front end where each firm can configure their own queues, menus, forwarding etc. My problem is that my bosses want to expand massively, they are currently talking of at least a tenfold increase in the number of clients. I'm fairly certain our Asterisk server won't be able to handle that. Our current 15 clients all have peak usage at the same time (with 2/3 of all traffic between 8 and 9 in the morning). At peak times, we have 20% CPU load with some 100 concurrent calls and a little under one call/second. I have to solve the scalability problem within a relatively short timeframe so starting from scratch with something new is out of the question. My first thought was to add another Asterisk server and use DUNDi load balancing between the two. But looking around and reading the discussion on this list got me to thinking whether some sort of SIP switch or router/proxy could take some load off the Asterisk server(s). One of my main concerns is to change our current setup as little as possible. It's a mishmash of Asterisk, MySQL, Rails and RAGI/RAMI. The original programmers are no longer available to me and I am still very wet behind the ears when it comes to VOIP. So should I be looking at adding e.g. OpenSIP as a sip proxy to our current setup or adding a second (and then a third and a fourth ...) Asterisk server with DUNDi? Or both? Will adding OpenSIP require a change in the way in which we handle SIP peers or require some major reconfiguration of Asterisk? It seems to me that DUNDi requires minimal configuration changes but I don't really know. Any information and recommendations will be greatly appreciated! Regards Binni There are a lots of solutions to asterisk scalability. Each one with its own pros and cons. If you have several small firms, the easiest path will be to duplicate your installation and share your clients among all the servers. Firm01 to Firm15 will be on server01, Firm16 to Firm25 on server02 and so on... However if you have such big numbers of contemporary calls (the max I recorded on one of my server was 60 active calls), maybe you need to think better to high availability, duplicating each server and putting them in high availability. One other way, the one I prefer is to completely share the load among a bunch of servers using mysql multimaster replication and asterisk realtime. Client's phones will use SRV to locate the best server. This way, you can just increase the capacity adding servers and you are completely fault tolerant. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP
--- On Fri, 2/10/12, Leandro Dardini ldard...@gmail.com wrote: mysql multimaster replication and asterisk realtime. Just a word of caution: I've had terrible luck with MySQL NDB tables in a multimaster setup. I'm not a big expert but v.5.0 and 5.1 have given me lots of reliability issues (I lost table data several times). I'd like to try postgresql in a multimaster setup. Realtime with a clustered database is a nice idea but is it reliable? Any long-term success stories? 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] TE410P (1st) without cables always green
Shaun, Follwing more tests : 1) In production box asterisk works with 2.4.X dahdi tree + kernel 2.33.x tree. I can put the trunk up and recieve some calls . 2) In the same box I had tested dahdi 2.6.X + same kernel, when run dahdi_cfg this error messages comes , in dahdi_cfg running and no interrupts are in dmesg .. Any idea ? Loading DAHDI hardware modules: wct4xxp: [ OK ] wcfxo: [ OK ] Running dahdi_cfg: DAHDI startup failed: Input/output error Please see the follwing quoted ( in old messages ) text about 2.6 dahdis, as you can see in devel box, dahdi_cfg works in new dadhdi ( I not know if pri goes up in asterisk, because this box are without ) Any idea here ? 3) I remember some problems in dahdi=2.6 our 2.5 with newer kernel trees. This issue ( new dahd's not works and old dahd's not compile in new kerne tree) will be a problem in future time. A brazilian guy post to me that are problems in new dahdi and old boards.. , can you confirm any issue in this points ? Regards, Em 09/02/2012 20:30, Marcio Gomes escreveu: Shaun, snip Just thinking out loud here but I'm guessing it may be fair to just set alarmdebounce to 0 by default on gen1 cards. With dahdi-linux 2.2.0.2 does your card function if you set alarm debounce to 2500? /etc/init.d/dahdi stop sleep 3 modprobe dahdi modprobe wct4xxp debug=1 alarmdebounce=2500 dahdi_cfg cat /proc/interrupts | grep wct4xxp; sleep 5 ; cat /proc/interrupts | grep wct4xxp 20: 62045611 62097065 IO-APIC-fasteoi wct4xxp 20: 62045613 62097065 IO-APIC-fasteoi wct4xxp No interrupts, I do some tests and alarmdebounce should be 0 or 1 to generate interrupts correctly. Just thinking out loud here but I'm guessing it may be fair to just set alarmdebounce to 0 by default on gen1 cards. Yeah, it's correct IMO. After working hours ( until next monday ) I will do some tests in production box.. ( I have to change the kernel config and others small things I have modified in last week and are generating some Kpanic's in module load ). I can only do this in time slot 00:00 - 07:00 am regards, Marcio Gomes -- _ -- 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] distributed queue information over several Asterisk nodes
Is it possible to distribute QUEUE information among several Asterisk nodes in a multimaster or load balancing setup? I haven't tried this yet but if one uses realtime with a clustered multimaster database and the queue agents/members are fixed SIP channels (eg. SIP/100) then I guess that the Queue app will be able to contact the member no matter to which Asterisk node it registered. However, what happens if incoming calls enter more than one queue (a queue on any Asterisk node, as it would be expected in a fully load-balanced setup)? Let's say QUEUE1 on ASTNODE1 has 1 incoming call waiting to be picked up and a second call comes in but enters QUEUE1 on ASTNODE2 which was previously empty. So for example, how can the caller in QUEUE1 on ASTNODE2 be placed in position 2 instead of 1? In other words, can the same QUEUE work/collaborate over different Asterisk nodes? 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
Re: [asterisk-users] SIP hardware phones
I'm in a similar situation. However, most of my buildings were re-wired around 1994 to provide Cat5 or 5E to the desktop for data, and 2-pair Cat3 for voice, all in a star topology. I can move my voice infrastructure to an IP-based one running 10Mbps, utilize existing wiring infrastructure, with the only cost outlay being low cost PoE managed switches (48 ports for about a grand), and it ends up a lot cheaper than upgrading the data network to support the phones. ...and I can still stay within standard. Is this an option for you or are you still living with the remnants of an old key system or something like that? The journey of a thousand miles begins with a broken fan belt and a flat tire. On 2/8/2012 10:46 AM, Vieri wrote: Let me answer that, Carlos. A big hospital. These big infrastructures can be quite outdated and messy. Getting someone to cable old parts of the buildings can be very expensive. However, replacing just the backbone switches is something they can afford. And they don't need PoE, really. What kind of applications benefit from gigabit speed? Well, plenty, such as MDs having to view a whole bunch of x-ray images of several patients, as fast as possible. Believe me, doctors aren't patient and Gbps makes a big difference. So basically, that's your answer: these sites don't need PoE, just Gbps and can't afford cabling a huge old building. Now, they don't care for PoE on the hardphones either. So in these cases, I think it's clearly justifiable to have a low-budget Digium D40 or Grandstream GXP280 with a 2-NIC Gbps switch. Not a big deal anyway, because they can always add a mini 5 or 8-port gigiabit switch for around 20$ between the wall socket and the hardphone+PC, but that just adds another appliance to the doctor's office... --- On *Wed, 2/8/12, Carlos Alvarez /car...@televolve.com/* wrote: From: Carlos Alvarez car...@televolve.com Subject: Re: [asterisk-users] SIP hardware phones To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, February 8, 2012, 9:26 AM If the customer is so cheap that they won't properly build out the network, why would they have gigabit switches to the desktop which have a limited set of applications that actually benefit from it? Then there's PoE, which is expensive to start and very expensive with gigabit. So this mythical customer is too cheap to cable, but will buy a gigabit switch of dubious value, will they buy a PoE gigabit switch? If not, why not buy a value-priced PoE 100m switch which has a clear benefit instead of a low-end GB switch of dubious value? I just don't see the fit, and I'm guessing the vendors don't either. What is the exact network topology (brands/models) and applications that justify GB to the desktop, don't justify additional cabling, and how do you account for PoE in this environment? On Wed, Feb 8, 2012 at 7:13 AM, Vieri rentor...@yahoo.com /mc/compose?to=rentor...@yahoo.com wrote: --- On Wed, 2/8/12, Jason W. Parks jason.w.pa...@gmail.com /mc/compose?to=jason.w.pa...@gmail.com wrote: From everything I've researched to date, my understanding is most locations have chosen to double their port density and continue to service the phone and computer on separate ports than to share a single line for both computer and phone. Reason primarily mentioned being troubleshooting concerns. If this is the case, the second port is not required, and become nothing but another gimmick to sell to you. Is this everyone else's experience as well? Well, at some locations, for technical and mostly political reasons, doubling port density so that the computer connects to a separate port is too costly, way over what a 60$ hardphone can cost (eg. Grandstream GXP285). I'd be glad to pay just a tad more for hundreds of basic hardphones, just as long as they can do gigabit. 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
Re: [asterisk-users] SIP hardware phones
2012/2/10, Jason W. Parks jason.w.pa...@gmail.com: I'm in a similar situation. However, most of my buildings were re-wired around 1994 to provide Cat5 or 5E to the desktop for data, and 2-pair Cat3 for voice, all in a star topology. I can move my voice infrastructure to an IP-based one running 10Mbps, utilize existing wiring infrastructure, with the only cost outlay being low cost PoE managed switches (48 ports for about a grand), and it ends up a lot cheaper than upgrading the data network to support the phones. ...and I can still stay within standard. Is this an option for you Yes this is an option but the original question why no low-end Gigabit phones on the market ?. Try to find a PC motherboard with 10/100 interface. Now, it's Gigabit for all, no matter if people need its speed or not. And both, IP phones and PC motherboard are 100$ products. What strikes me is that it's still not the case in 2012, for IP phones. I can live with that but I'm still a bit surprised by this remaining year after year. or are you still living with the remnants of an old key system or something like that? The journey of a thousand miles begins with a broken fan belt and a flat tire. On 2/8/2012 10:46 AM, Vieri wrote: Let me answer that, Carlos. A big hospital. These big infrastructures can be quite outdated and messy. Getting someone to cable old parts of the buildings can be very expensive. However, replacing just the backbone switches is something they can afford. And they don't need PoE, really. What kind of applications benefit from gigabit speed? Well, plenty, such as MDs having to view a whole bunch of x-ray images of several patients, as fast as possible. Believe me, doctors aren't patient and Gbps makes a big difference. So basically, that's your answer: these sites don't need PoE, just Gbps and can't afford cabling a huge old building. Now, they don't care for PoE on the hardphones either. So in these cases, I think it's clearly justifiable to have a low-budget Digium D40 or Grandstream GXP280 with a 2-NIC Gbps switch. Not a big deal anyway, because they can always add a mini 5 or 8-port gigiabit switch for around 20$ between the wall socket and the hardphone+PC, but that just adds another appliance to the doctor's office... --- On *Wed, 2/8/12, Carlos Alvarez /car...@televolve.com/* wrote: From: Carlos Alvarez car...@televolve.com Subject: Re: [asterisk-users] SIP hardware phones To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, February 8, 2012, 9:26 AM If the customer is so cheap that they won't properly build out the network, why would they have gigabit switches to the desktop which have a limited set of applications that actually benefit from it? Then there's PoE, which is expensive to start and very expensive with gigabit. So this mythical customer is too cheap to cable, but will buy a gigabit switch of dubious value, will they buy a PoE gigabit switch? If not, why not buy a value-priced PoE 100m switch which has a clear benefit instead of a low-end GB switch of dubious value? I just don't see the fit, and I'm guessing the vendors don't either. What is the exact network topology (brands/models) and applications that justify GB to the desktop, don't justify additional cabling, and how do you account for PoE in this environment? On Wed, Feb 8, 2012 at 7:13 AM, Vieri rentor...@yahoo.com /mc/compose?to=rentor...@yahoo.com wrote: --- On Wed, 2/8/12, Jason W. Parks jason.w.pa...@gmail.com /mc/compose?to=jason.w.pa...@gmail.com wrote: From everything I've researched to date, my understanding is most locations have chosen to double their port density and continue to service the phone and computer on separate ports than to share a single line for both computer and phone. Reason primarily mentioned being troubleshooting concerns. If this is the case, the second port is not required, and become nothing but another gimmick to sell to you. Is this everyone else's experience as well? Well, at some locations, for technical and mostly political reasons, doubling port density so that the computer connects to a separate port is too costly, way over what a 60$ hardphone can cost (eg. Grandstream GXP285). I'd be glad to pay just a tad more for hundreds of basic hardphones, just as long as they can do gigabit. 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
Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?
Hi Brynjolfur, Yes, this is exactly what I am looking for - hopefully in English :-) Date or range selection would make this perfect. I have been looking for something like this for quite a while but there is none. I would really appreciate it if you share this with me. Question here, does the .php code read from database and displays or does it analyse the custom-cdr.csv file? Best, On Fri, Feb 10, 2012 at 4:10 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: Hi, I forgot to add that you are free to use my code, I’ll mail it later today. ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *Brynjolfur Thorvardsson *Sendt:* 10. februar 2012 09:47 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? ** ** Hi, I’m working on a small php program for just this. I guess from your question that you have Asterisk writing to a CDR database table, in which case you should be able to use my .php code fairly easily. It’s nothing fancy but does give me a graphical presentation of calls/15minute segments. ** ** Attached is a screenshot of a graph, I have 1,5+ million entries in the table but there is no noticeable lag in refreshing the graph. At the moment it refreshes only when the button is pressed (the text is in Danish ...) but changing it to refresh automatically every 15 minutes wouldn’t be a major problem. I’m working on adding the option of selecting date ranges, it’s all still a work in progress! ** ** Regards ** ** Binni ** ** *Fra:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] *På vegne af *asterisk jobs *Sendt:* 9. februar 2012 16:36 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? ** ** Hi everyone, ** ** I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would amazing to feed the asteriskcdrdb table to the program and get back the results without installing anything on the Asterisk server as I don't want to tamper with the server. ** ** Is there such a tool? ** ** 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] Question for the group
Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: * - Email to Fax and Fax to Email - Inward DID local and 800 services - Calling card SIP based and ANI authenticated * I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? Thanks G -- _ -- 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 there a php script to analyse and show call detail reports from Asterisk CDR?
- Original Message - Yes, this is exactly what I am looking for - hopefully in English :-) Date or range selection would make this perfect. I have been looking for something like this for quite a while but there is none. I would really appreciate it if you share this with me. Question here, does the .php code read from database and displays or does it analyse the custom-cdr.csv file? Don't forget about the ever-popular Asterisk-stat and the newly revised cdr-stats projects, both web based, proven, and work fantastic: http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 http://www.cdr-stats.org/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question for the group
- Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: • Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticated I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: Thanks for this - but I am looking really for a software type solution. The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question for the group
I assume that solution was A2Billing? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group - Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: • Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticated I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: Thanks for this - but I am looking really for a software type solution. The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 there a php script to analyse and show call detail reports from Asterisk CDR?
No, that doesn't do the job I specifically asked and installation instructions are all over the place... Thanks though. On Fri, Feb 10, 2012 at 11:36 AM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Yes, this is exactly what I am looking for - hopefully in English :-) Date or range selection would make this perfect. I have been looking for something like this for quite a while but there is none. I would really appreciate it if you share this with me. Question here, does the .php code read from database and displays or does it analyse the custom-cdr.csv file? Don't forget about the ever-popular Asterisk-stat and the newly revised cdr-stats projects, both web based, proven, and work fantastic: http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 http://www.cdr-stats.org/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Question for the group
There is none. We are looking to develop our own currently and in the process of hunting down best developers. We have a great deal of experience with billing systems but doing a fron-end for this purpose just requires multiple developers. You can e-mail me in private if interested in a shared project. Best, On Fri, Feb 10, 2012 at 11:34 AM, James Wystead szilvertho...@gmail.comwrote: Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: * - Email to Fax and Fax to Email - Inward DID local and 800 services - Calling card SIP based and ANI authenticated * I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? Thanks G -- _ -- 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] Question for the group
Yes, I like the look of that. Researching it too - the commercial one looks nice too, but I don't know if there is a budget. G On Fri, Feb 10, 2012 at 11:52, Terry Brummell te...@brummell.net wrote: I assume that solution was A2Billing? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group - Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: • Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticated I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: Thanks for this - but I am looking really for a software type solution. The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Question for the group
What have you looked for yet? There are no commercial ones that do all that in one. On Fri, Feb 10, 2012 at 11:57 AM, James Wystead szilvertho...@gmail.comwrote: Yes, I like the look of that. Researching it too - the commercial one looks nice too, but I don't know if there is a budget. G On Fri, Feb 10, 2012 at 11:52, Terry Brummell te...@brummell.net wrote: I assume that solution was A2Billing? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group - Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: • Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticated I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: Thanks for this - but I am looking really for a software type solution. The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Question for the group
Thanks for this - but I am looking really for a software type solution.I would venture say that he means he wants it for free.Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Question for the group From: James Wystead szilvertho...@gmail.com Date: Fri, February 10, 2012 11:57 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Yes, I like the look of that.Researching it too - the commercial one looks nice too, but I don't know if there is a budget.GOn Fri, Feb 10, 2012 at 11:52, Terry Brummell te...@brummell.net wrote: I assume that solution was A2Billing? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group - Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestionsWe are going to be setting up a prepaid PBX service with the following features:• Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticatedI see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: "Thanks for this - but I am looking really for a software type solution". The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Call Completion
Hello, I'm trying the Call Completion system. All work fine. I still don't undesrtand how Asterisk work. Does Asterisk use sip signaling or other protocol to send notifications? On the Asterisk Wiki seems that the system is based on draft-ietf-bliss-call-completion-04 but this draft talk about SUBSCRIBE and NOTIFY. I see nothing in sip capture. Can you help me with this question? Thank's Regards -- _ -- 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] Call Completion
On 02/10/2012 11:32 AM, bakko wrote: Hello, I'm trying the Call Completion system. All work fine. I still don't undesrtand how Asterisk work. Does Asterisk use sip signaling or other protocol to send notifications? On the Asterisk Wiki seems that the system is based on draft-ietf-bliss-call-completion-04 but this draft talk about SUBSCRIBE and NOTIFY. I see nothing in sip capture. If your SIP UAs (phones) don't implement that draft, then they won't SUBSCRIBE for call-completion events. That means you'll be using the 'generic' call-completion agent and monitor in your sip.conf file. Asterisk can use SUBSCRIBE/NOTIFY to handle call-completion with another Asterisk server if your network has such a connection, but otherwise we aren't aware of any SIP UAs (phones or otherwise) other than Asterisk that have implemented this draft. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question for the group
Great subject, regardless of the name you post from or the list you post to. I'm sure this is the only question that will ever be asked of either group. Better subject = better replies = better value for future archive searchers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 SIP Realtime Architecture Issue/Bug
I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing registration it registered fine on asterisk as peer is available in Database. But now i am doing 'sip reload' or 'reload' due to some reason my peer registration is going out and i cannot able to call that peer even though in SIP client it shows me 'registered'. Can any body elaborate on this issue which settings i need to put in sip.conf. I also tried to follow this patch https://issues.asterisk.org/view.php?id=14196 But it allready applied in code base so why it wont work? Here is my sip.conf settings. [general] context=from-internal ; Default context for incoming cal rtcachefriends=no rtupdate=yes rtautoclear=yes rtsavesysname=yes callcounter = yes callevents=yes bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) srvlookup=yes ; Enable DNS SRV lookups on outbound calls pedantic=yes ; Enable slow, pedantic checking for Pingtel tos=184 ; Set IP QoS to either a keyword or numeric val tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpiry=3600 ; Max length of incoming registration we allow defaultexpiry=120 ; Default length of incoming/outoing registration preferred_codec_only=yes disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw insecure=invite language=en ; Default language setting for all users/peers rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=dhaval ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 qualify=yes nat=yes ;canreinvite=yes directmedia=yes directrtpsetup=yes And here is DB fields snapshots. id: 1 name: 201 ipaddr: 172.18.100.243 port: 53624 regseconds: 1328716180 defaultuser: 201 fullcontact: NULL regserver: dhaval useragent: CSipSimple r1133 / b lastms: 554 host: dynamic type: friend context: from-internal permit: NULL deny: NULL secret: 201 md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: yes nat: NULL allow: ulaw disallow: g729 insecure: invite callerid: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL Kindly help me to resolve this. Thanks Dhaval The first thing I would try is 'rtcachefriends=yes', that should do it. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Call Completion
Thank you for your answer Kevin. Effectively, I'm using CCSS in generic mode. I hope to try the draft between two Asterisk Server soon. Regards -- _ -- 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 vs FreeSwitch vs Asterisk + OpenSIP
Hi Leandro, that's a really good suggestion. Thanks a lot, I'll certainly give it a try. -Oprindelig meddelelse- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Leandro Dardini Sendt: 10. februar 2012 14:03 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP 2012/2/10 Brynjolfur Thorvardsson bi...@itanet.nu: I hope I'm not flogging a dead horse here, but the discussion around the whole scalability issue in Asterisk have opened my eyes to a whole lot of issues, making me increasingly confused! We have a fully functioning and stable installation where we offer PBX services to some 15 small firms (basically medical practices). These are based all over the country, with between 2 and 15 SIP phones each. We have a Web front end where each firm can configure their own queues, menus, forwarding etc. My problem is that my bosses want to expand massively, they are currently talking of at least a tenfold increase in the number of clients. I'm fairly certain our Asterisk server won't be able to handle that. Our current 15 clients all have peak usage at the same time (with 2/3 of all traffic between 8 and 9 in the morning). At peak times, we have 20% CPU load with some 100 concurrent calls and a little under one call/second. I have to solve the scalability problem within a relatively short timeframe so starting from scratch with something new is out of the question. My first thought was to add another Asterisk server and use DUNDi load balancing between the two. But looking around and reading the discussion on this list got me to thinking whether some sort of SIP switch or router/proxy could take some load off the Asterisk server(s). One of my main concerns is to change our current setup as little as possible. It's a mishmash of Asterisk, MySQL, Rails and RAGI/RAMI. The original programmers are no longer available to me and I am still very wet behind the ears when it comes to VOIP. So should I be looking at adding e.g. OpenSIP as a sip proxy to our current setup or adding a second (and then a third and a fourth ...) Asterisk server with DUNDi? Or both? Will adding OpenSIP require a change in the way in which we handle SIP peers or require some major reconfiguration of Asterisk? It seems to me that DUNDi requires minimal configuration changes but I don't really know. Any information and recommendations will be greatly appreciated! Regards Binni There are a lots of solutions to asterisk scalability. Each one with its own pros and cons. If you have several small firms, the easiest path will be to duplicate your installation and share your clients among all the servers. Firm01 to Firm15 will be on server01, Firm16 to Firm25 on server02 and so on... However if you have such big numbers of contemporary calls (the max I recorded on one of my server was 60 active calls), maybe you need to think better to high availability, duplicating each server and putting them in high availability. One other way, the one I prefer is to completely share the load among a bunch of servers using mysql multimaster replication and asterisk realtime. Client's phones will use SRV to locate the best server. This way, you can just increase the capacity adding servers and you are completely fault tolerant. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 queuing behavior
I'm trying to implement a very simple call queue for a small, low volume helpdesk. We have 2-5 agents, and rarely does the queue get more than 1 or 2 callers deep. I'm using the ringall strategy and I want calls answered in FIFO order. Say caller A calls the queue, and there is one member logged in. Asterisk rings the member. Now, caller B calls. Asterisk rings the member. Now the member's handset is showing two incoming calls. This particular member is a bit lazy or busy, so he waits 30 seconds, and the first call times out. Asterisk says, Nobody picked up in 3 ms, the caller hears the periodic announcement, and Asterisk stops ringing the member. Now, the member is unbusy, so he answers a call. But, he's connected to caller B, even though caller A called first. That's not what I'd expect - I want callers to be answered in FIFO order. I suspect there's some interaction with the ringinuse and timeout settings here. I had thought, maybe I'll make the timeout very long. Since I'm using ringall, I don't have to worry about a lazy/dead member not answering and thus preventing the caller from being presented to the next member. However, if I do this, I can't seem to make it longer than 60 seconds, and also the caller seems to only be presented with announcements when the timeout expires. I'd like to tell the caller every 30 seconds that they can press 0 to leave a voicemail, regardless of any other queue activity. ringinuse=no might be nice, also so if there are more than three callers in the queue I don't eat up all the call appearance buttons on my member's handsets. However, I read that only SIP channels can report in use, and my members are on OOH323 channels. So, that's out. Coincidentally, I could make my members be just one, which is a hunt group implemented in another PBX. I'd then want Asterisk to present one caller to this one member, and keep presenting that caller to the one member until it's answered, or the caller has been waiting over five minutes, when he's sent to voicemail. Only then is the next caller presented. Even though I'd think it would be easy for app_queue to know that the member is busy (after all, it's calling them), there doesn't seem to be any way to direct app_queue to not throw every caller in the queue at the one member. Any ideas on how I might approach a better solution? -- _ -- 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 does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
May I know how the compile RPM from Digium Repo gets to install DAHDI so easily on the VM? Can you please point me to how this compilation is done so I can have my own RPM of Asterisk with all options added on (e.g. ooh323, jabber, etc...) Thanks On Mon, Jan 16, 2012 at 1:57 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/16/2012 12:52 PM, Patrick Lists wrote: On 16-01-12 19:47, Russ Meyerriecks wrote: [snip] 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Dahdi is a set of drivers for telephony hardware. You won't need it for pure sip Asterisk implementations. Unless things have changed with recent versions I think you still need DAHDI if you want to use MeetMe and maybe other modules that require proper timing (which DAHDI provides). They have changed; DAHDI is required for MeetMe/SLA/Page, but is not required for timing. In Asterisk 10, ConfBridge can be a suitable replacement for MeetMe for many users as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://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] Call queuing behavior
On Feb 10, 2012, at 14:37 , Phil Frost wrote: I'm trying to implement a very simple call queue for a small, low volume helpdesk. We have 2-5 agents, and rarely does the queue get more than 1 or 2 callers deep. I'm using the ringall strategy and I want calls answered in FIFO order. Say caller A calls the queue, and there is one member logged in. Asterisk rings the member. Now, caller B calls. Asterisk rings the member. Now the member's handset is showing two incoming calls. This particular member is a bit lazy or busy, so he waits 30 seconds, and the first call times out. Asterisk says, Nobody picked up in 3 ms, the caller hears the periodic announcement, and Asterisk stops ringing the member. Now, the member is unbusy, so he answers a call. But, he's connected to caller B, even though caller A called first. That's not what I'd expect - I want callers to be answered in FIFO order. [...] I think I've found a solution. This forum thread describes a similar situation: http://forums.asterisk.org/viewtopic.php?f=1t=79450 I set autofill=no in queues.conf. This seems to make more sense when strategy=ringall. Now any member is only ever presented with one caller, and it's always the next caller in the queue. If the timeout expires, they are presented with the same caller again. For a larger call center, this would be an unacceptable bottleneck, but for my situation it's ideal. -- _ -- 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] Weird IPs in Fail2ban list
I can't see those IPs in the /var/log/asterisk/full. I can't event see
parts of the IP address as I try *grep -o 23.20.189 full. *That is still
nothing.
I am wondering what is wrong here. This is my regex filter file:
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No
matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Device does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer
is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*')
.* SIP/HOST-.* Playing 'ss-noservice.gsm' .*
Thanks,
On Fri, Jan 27, 2012 at 2:16 AM, Mikhail Lischuk mlisc...@itx.com.uawrote:
**
asterisk jobs писал 27.01.2012 06:49:
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
Chain fail2ban-ASTERISK (1 references)
num target prot opt source destination
1DROP all -- 0.23.20.189 0.0.0.0/0
I also get things like, 0.0.5.2, etcFail2ban seems to be working
when I am testing. Are these numbers taken from the SIP packet or the
TCP/IP protocol source because they surely are not valid addresses.
Thanks
Did you find those IPs in Asterisk log?
If so - it isn't Fail2Ban problem, for it just parses logs and extracts
substring
--
With Best Regards
Mikhail Lischuk mlisc...@itx.com.ua
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Re: [asterisk-users] Weird IPs in Fail2ban list
On 27 January 2012 04:49, asterisk jobs asteriskcod...@gmail.com wrote: Hello everyone, I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen this or can explain this? Chain fail2ban-ASTERISK (1 references) num target prot opt source destination 1DROP all -- 0.23.20.189 0.0.0.0/0 I also get things like, 0.0.5.2, etcFail2ban seems to be working when I am testing. Are these numbers taken from the SIP packet or the TCP/IP protocol source because they surely are not valid addresses. Thanks -- ___ 189.20.23.0 ? __ -- 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 -- _ -- 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] Polycom firmware 4.0.1 and paging
Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? Regards, Mike -- _ -- 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] Call queuing behavior
On Fri, 10 Feb 2012 16:08:28 -0500 Phil Frost p...@macprofessionals.com wrote: On Feb 10, 2012, at 14:37 , Phil Frost wrote: Now, caller B calls. Asterisk rings the member. Now the member's handset is showing two incoming calls. [...] Now, the member is unbusy, so he answers a call. But, he's connected to caller B, even though caller A called first. That's not what I'd expect - I want callers to be answered in FIFO order. I think I've found a solution. This forum thread describes a similar situation: http://forums.asterisk.org/viewtopic.php?f=1t=79450 I set autofill=no in queues.conf. This seems to make more sense when strategy=ringall. Now any member is only ever presented with one caller, and it's always the next caller in the queue. If the timeout expires, they are presented with the same caller again. For a larger call center, this would be an unacceptable bottleneck, but for my situation it's ideal. Yes, I was going to suggest autofill=no, until I saw your reply. But I do have one thing to add, just to put it out there... I can't really think of a situation where you would want autofill=yes with ringall. They seem to me to be entirely mutually exclusive. It may be a good idea for Asterisk to either document that they don't work well together, or to make ringall disable autofill. Any comments? -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Polycom firmware 4.0.1 and paging
On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote: Hi, ** ** I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. ** ** Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it’s worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? *** * ** ** Regards, ** ** Mike ** Hi Mike, Is there a compelling reason to put version 4.0.1b on these phones? Brian ** ** ** ** ** ** ** ** ** ** ** -- _ -- 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] Polycom firmware 4.0.1 and paging
Did the 4.0.1b update overwrite sip.ld on these phones? If I recall correctly you have to tweak that file to make auto-answer work correctly. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt Sent: Friday, February 10, 2012 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? Regards, Mike Hi Mike, Is there a compelling reason to put version 4.0.1b on these phones? Brian -- _ -- 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] Polycom firmware 4.0.1 and paging
Not really, it did fix the phantom ringing I had (phone continued to ring when connected to a caller), which was the main reason to upgrade, but I believe so would upgrading to 3.3.4. Some pluses for me are: - It does make booting up MUCH faster - There is a Warning message when no registrations are successful (as opposed to just empty phone icons) making remote support easier (would you describe the phone icon gets old fast) But that's about it. Yes, I know if it ain't broke don't fix it, but in the end my decision was to move FROM 3.3.2 because of the ringing issue, so I chose 4.0.1b instead of 3.3.4. It`s a decision I am rethinking, to be honest. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt Sent: Friday, February 10, 2012 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? Regards, Mike Hi Mike, Is there a compelling reason to put version 4.0.1b on these phones? Brian -- _ -- 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] Polycom firmware 4.0.1 and paging
It does update the sip.ld file, yes. So does all upgrades, no? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, February 10, 2012 5:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Did the 4.0.1b update overwrite sip.ld on these phones? If I recall correctly you have to tweak that file to make auto-answer work correctly. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt Sent: Friday, February 10, 2012 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? Regards, Mike Hi Mike, Is there a compelling reason to put version 4.0.1b on these phones? Brian -- _ -- 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] Polycom firmware 4.0.1 and paging
Mike. Yes sip.ld is the firmware. I wanted to jump in because i saw you had the phantom ringing problem as well. I am running 3.3.1 and thought upgrading to 3.3.2 would solve that problem did you still have the problem in 3.3.2? I thought I saw in the release notes for 3.3.2 that was resolved. I dont have them infront of me but i suppose it is time to double check as I plan on upgrading 30 phones in the morning. I did test 3.3.2 but the phantom ring seemed so rand i thought i could just no reprouduce it. Thanks!! Jim - Original message - It does update the sip.ld file, yes. So does all upgrades, no? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, February 10, 2012 5:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Did the 4.0.1b update overwrite sip.ld on these phones? If I recall correctly you have to tweak that file to make auto-answer work correctly. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt Sent: Friday, February 10, 2012 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? Regards, Mike Hi Mike, Is there a compelling reason to put version 4.0.1b on these phones? Brian -- _ -- 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] Asterisk vs FreeSwitch vs Asterisk + OpenSIP
Sorry for the top post, but I am using a silly mail client. I havent talked about ndb tables, just multimaster setup. It is really stable if done with just two mysql servers. I am running a couple of asterisk servers sharing a common cdr and cnam database for at least 3 years without problems. Simple Mysql multimaster replication is really solid and easy to setup and maintain. Dont forget to handle the autoincrement columns with a distinct starting point and a common step, greater than one of course! Leandro Il giorno 10/feb/2012 14:23, Vieri rentor...@yahoo.com ha scritto: --- On Fri, 2/10/12, Leandro Dardini ldard...@gmail.com wrote: mysql multimaster replication and asterisk realtime. Just a word of caution: I've had terrible luck with MySQL NDB tables in a multimaster setup. I'm not a big expert but v.5.0 and 5.1 have given me lots of reliability issues (I lost table data several times). I'd like to try postgresql in a multimaster setup. Realtime with a clustered database is a nice idea but is it reliable? Any long-term success stories? 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
[asterisk-users] Virtual Server
Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I heard, about proxmox, but I don't know if works fine. Regards Carlos -- _ -- 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] Weird IPs in Fail2ban list
That was just another weird IP showing up. On Fri, Feb 10, 2012 at 4:50 PM, dotnetdub dotnet...@gmail.com wrote: On 27 January 2012 04:49, asterisk jobs asteriskcod...@gmail.com wrote: Hello everyone, I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen this or can explain this? Chain fail2ban-ASTERISK (1 references) num target prot opt source destination 1DROP all -- 0.23.20.189 0.0.0.0/0 I also get things like, 0.0.5.2, etcFail2ban seems to be working when I am testing. Are these numbers taken from the SIP packet or the TCP/IP protocol source because they surely are not valid addresses. Thanks -- ___ 189.20.23.0 ? __ -- 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 -- _ -- 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] Virtual Server
On Fri, Feb 10, 2012 at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I use Asterisk on FreeBSD Jails and works great: http://www.freebsd.org/doc/en_US.ISO8859-1/books/handbook/jails.html I heard, about proxmox, but I don't know if works fine. Regards Carlos -- _ -- 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] Weird IPs in Fail2ban list
On 12-01-26 11:49 PM, asterisk jobs wrote: Hello everyone, I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen this or can explain this? Chain fail2ban-ASTERISK (1 references) num target prot opt source destination 1DROP all -- 0.23.20.189 0.0.0.0/0 I also get things like, 0.0.5.2, etcFail2ban seems to be working when I am testing. Are these numbers taken from the SIP packet or the TCP/IP protocol source because they surely are not valid addresses. What version of asterisk 1.8 are you using? I suspect this is a bug we recently fixed in 1.8.8.0+ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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] Virtual Server
I run two off virtuozo vps boxes - but capacity will always be the defining value Sent from my iPhone 4S On Feb 10, 2012, at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I heard, about proxmox, but I don't know if works fine. Regards Carlos -- _ -- 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] Virtual Server
I run my Asterisk system on a quad core Opteron system running VMWare ESXI 5. On Feb 10, 2012, at 21:18, Carlos Rojas crt.ro...@gmail.com wrote: Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I heard, about proxmox, but I don't know if works fine. Regards Carlos -- _ -- 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
