[asterisk-users] Radius Based Accounting for Asterisk
Hi everyone Any one used Radius based accounting for asterisk.Please give me details. Thanks Nikhil -- _ -- 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] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Hi Friends, I have occurred same problem on Asterisk-1.8.X. version. I need to upgrade our production asterisk-1.6.2.6 server to asterisk-1.8.X version. I have already used root user during configuration and installation of asterisk-1.8.X version but getting same error as *increase the maximum file descriptor* on centos-5.3 os. I have also set 32768 ulimit of file descriptor on centos but still no success. Thanks in advance. Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,Asterisk Technology On Tue, Feb 1, 2011 at 11:41 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Tuesday 01 February 2011 02:24:20 Benny Amorsen wrote: Tilghman Lesher tilgh...@meg.abyt.es writes: Correct; and Asterisk needs to be started as root, even if it will drop privileges after startup. Do this, and there should be no problems. Starting as root + dropping privileges is fine. Running configure as root is not so fine; that basically makes building RPMS impossible. Alternatively, if you can set ulimit -n 32768 in your RPM build environment (this needs to be set as a login requirement), you can sidestep the need for configure to run as root. The only reason it needs root is to expand the file descriptor limit so it can test using a file descriptor beyond 1023 (the usual limit). -- Tilghman -- _ -- 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] [newbie] Conference call
Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit R key on handset, which puts party #1 on hold and gives a dialtone 3. Call party #2 4. Once both parties are off-hook, hit R+ 3 on handset to bridge both calls and have a conference call Is MeetMe the right way to do this in Asterisk, or should I look at some other way? Ideally, I'd rather go through a VOSP to avoid the extra digital/analog conversion added by going through the FXO module, but free calls are only available when using that port :-/ Thank you. -- _ -- 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] sip trunk balancing
hi, is there some way to balance accross sip trunks by the number of calls? example 3 trunks alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3) alfa have 25 calls now i want next call terminate to delta. how to find in asterisk the current calls number on sip trunk alfa? 1) set call-limit in sip.conf. then in the dialplan sip show peer inuse|grep alfa - parse - if numcalls 25 then dial(sip/delta) 2) groupcount ? 3) what else? thanks Marek -- _ -- 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] Outgoing agent´s calls
Yes, my agents dial “willy-nilly”... I can´t use the ex-girlfriend because, the line numbers that uses the agents are diferent. May be agent 1 today use line number 553455 and tomorrow 553461... On Wed, Feb 2, 2011 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Wednesday, February 02, 2011 12:26 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Outgoing agent´s calls Hi, is there any way to manage outgoing calls from agents? Mi agents are answering in pstn lines. I can send agents outgoing calls to my Asterisk but I don't know wich agent is making the call...because, may be he is unregister... Is there any solution? Thanks You could start with DASI and ex-girlfriend logic in your dialplan. I’m assuming now that your agents dial “willy-nilly” (with no restrictions and you find out what they did when you read the CDR). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to get Current Calls details
On Wed, Feb 02, 2011 at 03:29:37PM +0530, Nikhil wrote: Thanks of reply. The command core show verbose is working. but the problem is, for one call we can see 2 results,there is no common field on these two. Take a closer look at the output. The link between the 2 can be found by matching channel==dstchannel for all channels. -- Daniel Tryba -- _ -- 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] Outgoing agent´s calls
Then DISA (I had it as DASI in OP because Im working from not so good memory) is probably your best bet. It is a simple built-in feature that lets you get an access code in the dialplan before performing an action such as dialing. Check this link http://nerdvittles.com/index.php?p=73 _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Thursday, February 03, 2011 6:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Outgoing agent´s calls Yes, my agents dial willy-nilly... I can´t use the ex-girlfriend because, the line numbers that uses the agents are diferent. May be agent 1 today use line number 553455 and tomorrow 553461... On Wed, Feb 2, 2011 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Wednesday, February 02, 2011 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Outgoing agent´s calls Hi, is there any way to manage outgoing calls from agents? Mi agents are answering in pstn lines. I can send agents outgoing calls to my Asterisk but I don't know wich agent is making the call...because, may be he is unregister... Is there any solution? Thanks You could start with DASI and ex-girlfriend logic in your dialplan. Im assuming now that your agents dial willy-nilly (with no restrictions and you find out what they did when you read the CDR). -- _ -- 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 trunk balancing
On 02/03/2011 11:41 AM, marek cervenka wrote: hi, is there some way to balance accross sip trunks by the number of calls? example 3 trunks alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3) alfa have 25 calls now i want next call terminate to delta. how to find in asterisk the current calls number on sip trunk alfa? 1) set call-limit in sip.conf. then in the dialplan sip show peer inuse|grep alfa - parse - if numcalls 25 then dial(sip/delta) 2) groupcount ? 3) what else? GROUP() would be the way to go, for sure. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe and admin users
In article 1296748085.2237.16.camel@shaft, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Is there an option on MeetMe that means the conference room is only available if an admin user is logged in? I've had a look the the application from the asterisk cli but I can't really see what I'm after. Currently using 1.4.17 (deb package) Soon moving up to 1.8.2 (rpm package) What you do is give admin users the A flag (marked user) as well as the a flag (admin user). Then you also give all users the w flag (wait until marked user joins) and optionally the x flag (exit when all marked users have left). Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] Queues and Agent penalty - how to go to second best agent when the first does not answer
Mike wrote: I was hoping to use this Queue not for professional agents in a call center, but for reception. When the receptionist (lowest penalty) is not at the desk, then some junior sales person can pick up those calls. We have our receptionist setup in a front-desk queue that has 2 phones in it. The incoming call rings directly to the phone for 30 seconds, if not answered, plays the, Please wait while we find someone and then drops them into a queue. At this point, it rings the operator phone again and if that fails, the 2nd phone. This will bounce back and forth between phones, until finally dropping the call into our dial-by-name directory if nobody answers. We also have both phones in a call group/pickup group, allowing to grab a call by doing a *7 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe and admin users
On Thu, 2011-02-03 at 16:39 +, Tony Mountifield wrote: In article 1296748085.2237.16.camel@shaft, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Is there an option on MeetMe that means the conference room is only available if an admin user is logged in? I've had a look the the application from the asterisk cli but I can't really see what I'm after. Currently using 1.4.17 (deb package) Soon moving up to 1.8.2 (rpm package) What you do is give admin users the A flag (marked user) as well as the a flag (admin user). Then you also give all users the w flag (wait until marked user joins) and optionally the x flag (exit when all marked users have left). Cheers Tony Thanks Tony That makes sense, however, I have a problem that this is on an incoming real phone number, but I'm sure I can work something out now I know the underlying principle. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Queues and Agent penalty - how to go to second best agent when the first does not answer
We have our receptionist setup in a front-desk queue that has 2 phones in it. The incoming call rings directly to the phone for 30 seconds, if not answered, plays the, Please wait while we find someone and then drops them into a queue. At this point, it rings the operator phone again and if that fails, the 2nd phone. This will bounce back and forth between phones, until finally dropping the call into our dial-by-name directory if nobody answers. We also have both phones in a call group/pickup group, allowing to grab a call by doing a *7 Thanks Doug. I realize there are many things I can do, I was just hoping to use an application command to do it all. What you described might just be what I end up doing. 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] how to get Current Calls details
I think you are looking for a way to have such a report, on console: CallCenter*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 2.168.11.731ea659 00:03:39 00236 ( 0.00%) 0.000 004685 00 ( 0.00%) 0.0011 2.168.21.556897bd 00:38:11 01153 0105 ( 0.09%) 0.000 000113 000231 ( 0.20%) 0.0052 it's possible, as u see i have it on my console, but i didn't remember in which configuration part, I had enabled it... I hope someone remember it... On Thu, Feb 3, 2011 at 3:54 PM, Daniel Tryba dan...@tryba.nl wrote: On Wed, Feb 02, 2011 at 03:29:37PM +0530, Nikhil wrote: Thanks of reply. The command core show verbose is working. but the problem is, for one call we can see 2 results,there is no common field on these two. Take a closer look at the output. The link between the 2 can be found by matching channel==dstchannel for all channels. -- Daniel Tryba -- _ -- 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 -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- 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] T.38 negotiation error
Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-' is in an unsupported T.38 negotiation state, cannot continue. In my sip.config general section I have added this lines t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no If I comment this lines, the fax is sended by G711 but this modality is too slow and instable. How can I do to solve this problem? Thank you Dott. Marcello Colucci Microsoft Certified Professional (70-176/70-155/70-100) mcolu...@sirioinformatica.it - SIRIO Informatica s.a.s. P.za Pericle Fazzini, 8 63039 San Benedetto del Tronto (AP) Tel.: 0039-0735-56.83.75 oppure 0039-06-916.503.224 FAX: 0039-0735-56.02.13 www.SIRIOInformatica.it http://www.sirioinformatica.it/ i...@sirioinformatica.it * Questo messaggio è stato inviato da SIRIO Informatica s.a.s. e può contenere informazioni di carattere estremamente riservato e confidenziale. Qualora non fosse il destinatario, la preghiamo di informarci immediatamente con lo stesso mezzo ed eliminare il messaggio, con gli eventuali allegati, senza trattenerne copia. Qualsiasi utilizzo non autorizzato del contenuto di questo messaggio costituisce violazione della legge 196/2003 in materia di protezione e trattamento dei dati personali, secondo la quale si ha l'obbligo di non prendere cognizione della corrispondenza tra altri soggetti, salvo più grave illecito, ed espone il responsabile alle relative conseguenze civili e penali. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 negotiation error
2011/2/3 Marcello Colucci (SIRIO Informatica s.a.s.) mcolu...@sirioinformatica.it: Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-' is in an unsupported T.38 negotiation state, cannot continue. In my sip.config general section I have added this lines t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no If I comment this lines, the fax is sended by G711 but this modality is too slow and instable. It depends on who you're faxing with. If you can control who you'll be T38-ing with, you might be able to solve this particular problem. But if you're T.38-ing with the whole world, in my opinion, there are too many broken T.38 implementations for you to depend on this. If you want reliable faxing, you should avoid voip altogether, and use a modem. I've gotten pretty good faxing by using audio passthrough, over sip, but it's only sip on a LAN, to a PRI gateway system. Trying to do voip fax over internet is cheap, but it's not reliable. -- _ -- 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] asterisk18 rpm issues
On Wednesday 02 February 2011 14:21:50 Jason Parker wrote: On 02/02/2011 02:14 PM, Frank Liu wrote: Hi there, Per the instruction from http://www.asterisk.org/downloads/yum , I setup the yum repository on my Centos 5 x86_64 machine and did a yum install asterisk18 asterisk18-configs then I startup the asterisk (with no changes to config) just to see if it runs, but see below errors in the /var/log/asterisk/messages: [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open shared object file: No such file or directory [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'chan_mgcp.so': /usr/lib/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc I checked the system and can't find the file /usr/lib/asterisk/modules/res_pktccops.so at all. I double checked the rpm file downloaded by yum and res_pktccops.so is not in any rpms. Asterisk should still load fine with this warning. chan_mgcp wouldn't work, but that isn't used very often. I will take a look at it. This is not true for CentOS 5 and other distributions where the version of GCC does not support attributes weak or weakref. See this issue: https://issues.asterisk.org/view.php?id=17707 I'm guessing a packaging error simply did not include that new module as an oversight. -- Tilghman -- _ -- 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 about EuroBRI final 2 digits
Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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 Blf / Directed Pickup
After someone sent me an email saying his directed pickup did not work. I
realized I forgot to mention that directed pickup needs to be enabled in
extensions.conf i.e. add the following
exten=_*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)
On Mon, Jan 17, 2011 at 4:00 PM, Gord Urquhart gord...@gmail.com wrote:
With SIP 3.2.X firmware (available on the Polycom download site) and
Asterisk 1.6.1, Polycom phones now support a full featured BLF showing
statuses of Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension. On the phone side for each line that is going to be monitored add
lines like the following to the phone's cfg file.
attendant.reg=1
attendant.resourceList.1.address=sip:205@192.168.1.102sip%3A205@192.168.1.102
attendant.resourceList.1.label=205
attendant.resourceList.2.address=sip:217@192.168.1.102sip%3A217@192.168.1.102
attendant.resourceList.2.label=217
Following 4 lines added Sept/10
call.directedCallPickupMethod=legacy
call.directedCallPickupString=*8
feature.12.name=directed-call-pickup
feature.12.enabled=1
Assuming my server is at 192.168.1.102, this will add two BLF lines to the
phone for extensions 205 and 217. Calls incoming to those extensions will
show a blinking green led on the monitoring phone, pressing the hard key
will pick the call up, if it is answered elsewhere the led will change to
solid red. AFAIK this cannot be configured via the phones web gui, you must
use the cfg files. You can also use versions of Asterisk older than 1.6.1 if
you remove the restriction on what asterisk thinks Polycom phones can
handle. Look in chan_sip.c for
if (strstr(p-useragent, Polycom)) {
p-subscribed = XPIDF_XML;
and change that line to
p-subscribed = DIALOG_INFO_XML;
cheers
gord
On Thu, Jan 13, 2011 at 4:26 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Thanks! Blf is working now. I forgot I had to set set subscribecontext.
When a phone is ringing, the blf light is solid red and the icon is a (/)
type icon indicating unavailable. I'm also interested in directed pickup.
I set up the following:
call.directedCallPickupString=*6 call.directedCallPickupMethod=legacy
Hitting the button next to the contact will speed dial the contact instead
of pick up the ringing call.
On 01/13/2011 10:54 AM, Sebastien Thomas wrote:
Ok, that looks good.
We use FreePBX, and I know I had to modify a couple Asterisk files to
get the BLF working ... here are some of my mods but may also be used
for FOP2 (I dont recall which go for BLF and which go FOP2).
vi /etc/asterisk/sip_registrations_custom.conf
allowsubscribe=yes
vi /etc/asterisk/sip_custom.conf
callevents=yes
notifyringing=yes
limitonpeers=yes
I also override some of the sip.cfg settings in the polycom dir with:
feature
feature.1.enabled=1
feature.9.enabled=0
feature.18.enabled=1
/
pres
pres.reg=1
pres.idleSoftkeys=0
/
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
*** Need help? Contact supp...@amplisys.ca mailto:supp...@amplisys.ca
***
On 2011-01-13, at 10:29 AM, Mark Murawski wrote:
Yeah... My directory looks like this:
directory
item_list
item
ln6288/lnfn/fnct6288/ctsd1/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6208/lnfn/fnct6208/ctsd2/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6234/lnfn/fnct6234/ctsd3/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6205/lnfn/fnct6205/ctsd4/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
item
ln6231/lnfn/fnct6231/ctsd5/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb
/item
/item_list
/directory
On 01/13/2011 10:20 AM, Sebastien Thomas wrote:
Is the buddy watch tag activated in yourmac-directory.xml file
?bw1/bw
item
lbSebastien/lb
fnSebastien/fn
lnThomas/ln
ct222/ct
sd1/sd
bw1/bw
/item
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
On 2011-01-13, at 1:32 AM, Mark Murawski wrote:
Would anyone happen to have some examples of polycom configs,
specifically the 650 with sidecar for blf.
I have the asterisk side all configured since I've set up blf with
other types of phones, but I'm missing the polycom side.
I've put together amac-directory.xml, and the sidecar now lists
numbers as speed dials but does not subscribe to blf.
--
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Re: [asterisk-users] RTP keepalive doesn't work
Hi Kevin, Did you have any luck tracking down the missing rtpkeepalive code? I'm really looking to get this working asap so I'd be happy to copy in/compile/trial some code if there's any available. Regards, Ryan. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Saturday, 29 January 2011 1:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] RTP keepalive doesn't work On 01/28/2011 09:24 AM, Ryan Tucker wrote: Thanks for the info, I guess I would expect asterisk to send 'silence' (in blank RTP form or something) if silence suppression is disabled. Just as I would expect any end point to send 'silence' if it was muted when silence suppression was disabled. It seems that RTP keepalives would serve this purpose, however this doesn't seem to be available either... Should I file a bug report re rtpkeepalive? No need... I'm already trying to track down when the code was removed, and for what reason. Once that is done I'll enter an issue to get it addressed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] standalone NOTIFY message handling for Asterisk
Hi, I am using Asterisk 1.6.2.11 to test 3rd party Voice Message server (VMS), currently when VMS send NOTIFY message (standalone NOTIFY, no previous SUBSCRIBE for the dialog for SIP), asterisk responded with 489 Bad Event, in the debug log it indicates as the following: [Feb 4 13:27:06] DEBUG[8353] chan_sip.c: Invalid SIP message - rejected , no callid, len 771 I have googled around regarding to this topic, I found similar issues reported, but that's all the old version back to 2004 around. So I would like to confirm, does Asterisk support standalone NOTIFY message for message waiting indicator regarding to voice mail. If yes, what's the configurations should be applied? My current configuration with Asterisk: sip.conf = [vms] type = peer -- tried both friend and peer host = a.b.c.d -- the VMS IP address unsolicited_mailbox = yes fromuser = lab_vms . [user1] mailbox = user1@SIP_Remote The NOTIFY message received: == NOTIFY sip:+user1@a.b.c.d:5060 SIP/2.0^M Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK-781458841-7-21411^M CSeq: 1 NOTIFY^M From: sip:lab_vms@a.b.c.d:5060;tag=1426753238-7761-21411^M To: sip:+user1@a.b.c.d:5060^M Call-ID: 1231979610-5-21411-CmvtCallId^M Route: sip:a.b.c.d:5060;transport=udp;lr^M Event: message-summary^M Accept: application/sdp,application/media_control+xml,message/sipfrag^M Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,INFO,PRACK,UPDATE^M Contact: sip:a.b.c.d:5060;transport=udp^M Max-Forwards: 70^M Supported: 100rel,timer,histinfo^M Subscription-State: active^M MIME-Version: 1.0^M Content-Type: application/simple-message-summary^M Content-Length: 40^M ^M Messages-Waiting: yes^M None: 5/0 (0/0)^M Thanks a lot for your help! Felton -- _ -- 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] Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes answeronpolarityswitch=yes usecallerid=yes cidsignalling=bell cidstart=ring ;hidecallerid=yes ;hidecalleridname=yes ;waitfordialtone=yes ;mwimonitor=no ;mwilevel=512 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ;mwisendtype=rpas,lrev callwaiting=yes ;restrictcid=no usecallingpres=yes sendcalleridafter = 1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=no group=1 callgroup=1 pickupgroup=1 ;immediate=yes immediate=no callerid = asreceived useincomingcalleridondahditransfer = yes callprogress=yes progzone=us faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no faxbuffers=6,full -- _ -- 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] Queues and Agent penalty - how to go to second best agent when the first does not answer
On Feb 3, 2011, at 11:12 AM, Kevin P. Fleming wrote: The Queue() application can automatically pause members who fail to answer; this would be the solution to your problem. With that solution in place, though, the agent will still need to be able to un-pause when they return to their desk, and since that is the case, they really should be taught to go on pause when they leave their desk as well :-) Not to mention that your caller has to wait for however long your agent timeout is when this happens the first time, which is bad customer service. I am a little confused as to what the OP wants the system to do? Call the proper agent, but when they don't answer, on the next call, it shouldn't call the same agent? OK, but for how long? 5 minutes? Until they manually unpause (current option as described by Kevin), 30 minutes? Should it then up their penalty? For how long? Maybe some more specifics would help here. Tom -- _ -- 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] PRI voice optimization
Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any voice related parameter that we need to set for INDIA specific region and is ther any voice hardware tester for PRI that we can use and tell us our PRI [telco] provider that problem is not from our side. let give some idea . below are my configuration as well. # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Global data loadzone= in defaultzone = in span = 1,0,0,ccs,hdb3 bchan = 1-15 dchan = 16 bchan = 17-31 span = 2,0,0,ccs,hdb3 bchan = 32-46 dchan = 47 bchan = 48-62 span = 3,0,0,ccs,hdb3 bchan = 63-77 dchan = 78 bchan = 79-93 span = 4,0,0,ccs,hdb3 bchan = 94-108 dchan = 109 bchan = 110-124 [channels] language=en context=from-pstn switchtype=euroisdn pridialplan=local prilocaldialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes resetinterval=never rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 0 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 -- _ -- 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] Email alerts for trunks (peers)
Hey Guys, I'm after a way to monitor our sip trunks (peers) and send an email if they go down. I know I could use 'asterisk -rx sip show peers' in a shell script but that seems messy, especially since I'd like to monitor it fairly closely (so I'd like to run it every 20 or 30 seconds or so). Is there a better way to do it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP keepalive doesn't work
On 02/04/2011 01:34 AM, Ryan Tucker wrote: Did you have any luck tracking down the missing rtpkeepalive code? I'm really looking to get this working asap so I'd be happy to copy in/compile/trial some code if there's any available. Yes, it was lost during a merge of code into Asterisk trunk after 1.6.2 was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen entered an issue on Mantis as a blocker for any more 1.8.x releases until this is resolved, as it is clearly a regression in the 1.8.x series. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [newbie] Conference call
Dear, Meetme is a default conference application, but you can try conference or konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konferencethe installation for conference or konference are more easy best On Thu, Feb 3, 2011 at 1:48 PM, Gilles codecompl...@free.fr wrote: Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit R key on handset, which puts party #1 on hold and gives a dialtone 3. Call party #2 4. Once both parties are off-hook, hit R+ 3 on handset to bridge both calls and have a conference call Is MeetMe the right way to do this in Asterisk, or should I look at some other way? Ideally, I'd rather go through a VOSP to avoid the extra digital/analog conversion added by going through the FXO module, but free calls are only available when using that port :-/ Thank you. -- _ -- 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] R: T.38 negotiation error
Thank you Kevin. Today I will upgrade my asterisk and then I will inform you about the result. Bye -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Kevin P. Fleming Inviato: giovedì 3 febbraio 2011 22.10 A: asterisk-users@lists.digium.com Oggetto: Re: [asterisk-users] T.38 negotiation error On 02/03/2011 07:30 PM, Marcello Colucci (SIRIO Informatica s.a.s.) wrote: Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. The current version is 1.6.2.16.1, and 1.6.2.17 is on the way (in release candidate mode). Is there some reason why you are running an old release? When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-' is in an unsupported T.38 negotiation state, cannot continue. You have not reported the version of res_fax and res_fax_digium that you are using. Given that your Asterisk version is quite old (nearly a year) and you may not be running the last FAX modules either, any attempt to help you out would be just guessing... because you could easily be experiencing a problem that has already been fixed. There have been many fixes to T.38 negotiation in Asterisk since March of 2010. If I had to guess, though, I would think that that receiver of this call has sent your system a T.38 re-INVITE before you started SendFAX running. If that is the case, this problem has already been solved and upgrading will take care of it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
