[asterisk-users] Using hint priority with LDAP extensions and users
Hi! I've configured LDAP to read both users and extensions from LDAP server. However, I'm experiencing problems with state tracking. Previously when using static files, I was able to map extension number with channel state using: [sip_phones] exten = 100,hint,SIP/user exten = user,hint,SIP/user .. rest of the dialplan ... Thus when someone called the user, hint SIP/user showed channel state as BUSY and I was able to use call limits etc. Now I've added this line to [sip_phones]: switch = Realtime/@ My hints, and call limits as well, stopped working. I've tried to move hints to LDAP (which would be ideal situation for me), setting AstPriority to hint but I don't think they are event fetched. So the question is... I'm I doing something wrong or it's just impossible to use those two solutions (hints + LDAP) together? PS. I'm using Asterisk 1.6.2 if it helps with anything. -- Maciej Paszta Mobile Systems Research Labs, Poznan University of Technology smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] warning diego viola the trouble maker for the world
i folk warning from diegoviola from paragway he say working for bridgecomm then teliax then flowroute isn't that a lie only? he need free morocco mobile traffic from me i refuse him i say to him if you help me with some web developement i can provide you lowered rate because was a friend of mine but now i am avoiding him step by step the asterisk folk may allready know him 190.23.0.0/16 he have a VPS in germany and US tacking contact from me and from the irc channels to sell traffic to? WTF that's not a traffic but a fad full route :@ diegoviola is the VoIp world killer and trouble maker __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5530 (20101014) __ Le message a été vérifié par ESET NOD32 Antivirus. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using hint priority with LDAP extensions and users
On 2010-10-14 11:29, Maciej Paszta wrote: Hi! I've configured LDAP to read both users and extensions from LDAP server. However, I'm experiencing problems with state tracking. Previously when using static files, I was able to map extension number with channel state using: [sip_phones] exten = 100,hint,SIP/user exten = user,hint,SIP/user .. rest of the dialplan ... Thus when someone called the user, hint SIP/user showed channel state as BUSY and I was able to use call limits etc. Now I've added this line to [sip_phones]: switch = Realtime/@ My hints, and call limits as well, stopped working. I've tried to move hints to LDAP (which would be ideal situation for me), setting AstPriority to hint but I don't think they are event fetched. So the question is... I'm I doing something wrong or it's just impossible to use those two solutions (hints + LDAP) together? PS. I'm using Asterisk 1.6.2 if it helps with anything. Seems like I've managed to fix the issue. The first thing was that my LDAP schema didn't allow for gecos attribute to be inside AsteriskSIPUser (provided asterisk.ldif misses a few things ;). Another one was to add rtcachefriends=yes to general section of sip.conf. The last was to add call-limit=AstAccountCallLimit to res_ldap.conf in [sip] section. Another issue is - can I keep hints in LDAP? -- Maciej Paszta Mobile Systems Research Labs, Poznan University of Technology smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Create channel bank with TDMoE
Tanks I want to create a channel bank with TDMoE. I have not to buy a product. Best, 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
[asterisk-users] How to connect asterisk PBX to PSTN
Hello community, I have successfully set up asterisk free PBX server and I am also able to connect to it by softphone. Now as next step I want to extend this to PSTN , My Required scenario: I need a number which will connect outside PSTN world to my PBX and by applying extension particular softphone or connected normal phone should get connected. Which hardware I need for it. Also please explain a bit of dial plans. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect asterisk PBX to PSTN
Hi Joshi, To connect with PSTN line you need FXO / FXS card. FXO is used to connect CO line and FXS is used to connect internal station line. With help of FXO you can connect the outside world and with help of FXS you can connect normal analog phones. Inspite of normal analog phones you can connect SIP phones (soft phones) also. Some vendors are there for these PSTN cards like Digium, Sangoma, Openvox. Good luck:) On Thu, Oct 14, 2010 at 7:05 PM, Jigar Joshi jiga...@gmail.com wrote: Hello community, I have successfully set up asterisk free PBX server and I am also able to connect to it by softphone. Now as next step I want to extend this to PSTN , My Required scenario: I need a number which will connect outside PSTN world to my PBX and by applying extension particular softphone or connected normal phone should get connected. Which hardware I need for it. Also please explain a bit of dial plans. 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 -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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] Passing variables into macros?
Hi,I cannot get this to work..I have two application maps that call these two
macro's...transfer is done on sip phone and transfer2 is done on the incoming
dahdi linethats all workingbut the value stored in dtmf12 is never
passed into the second macro so I get in the NoOp..
So how exactly can I do this...global variable,setGlobalVar,import etc..tried a
few combos but they dont seem to work as I think they should?
this is in etensions_custom.. have no Global variable in this config..
[macro-transfer]
;performed on callerexten = s,1,Read(dtmf12,,5,,2,10)exten =
s,n,NoOp(${dtmf12} [macro-transfer2]
;performed on callee
exten = s,1,Flash()exten = s,n,NoOp(${dtmf12} exten =
s,n,SendDTMF($dtmf12})exten = s,n,Hangup
ThanksJames
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[asterisk-users] clustering
Hi all, I am planning to do clustering for my company's asterisk servers. I dont know much about it, just read some articles on the internet and learned how to use DUNDi and some basic information about clustering. What I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk server as a gateway and multiple clustered asterisk servers behind it? cheers Thanks in advance -- Best Regards Rizwan Qureshi -- _ -- 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] Explain core show translation
Hi, I've read in http://www.voip-info.org/wiki/view/Asterisk+codecs but I still have questions about core show translation. How are values replied by core show translation computed in the the first place ? I've got 2 machines using a 3GHz Intel CPU : 1 is a Xeon, 1 is a Pentium 4 (gathered with cat /proc/cpuinfo) The Xeon machine is showing, for instance: gsm ulaw - 1601 The other shows: gsm ulaw - 2 Why are these values so different ? Is it correct to say if core show translation is showing a 4 digits value in its matrix, then the translation path between corresponding codecs is unusable. 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] clustering
use camailio for SIP SLB sip load balancer - Original Message - From: Rizwan Hisham To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, October 14, 2010 5:01 PM Subject: [asterisk-users] clustering Hi all, I am planning to do clustering for my company's asterisk servers. I dont know much about it, just read some articles on the internet and learned how to use DUNDi and some basic information about clustering. What I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk server as a gateway and multiple clustered asterisk servers behind it? cheers Thanks in advance -- Best Regards Rizwan Qureshi -- -- _ -- 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 __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a iti virifii par ESET NOD32 Antivirus. http://www.eset.com __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a été vérifié par ESET NOD32 Antivirus. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Explain core show translation
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, October 14, 2010 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Explain core show translation Hi, I've read in http://www.voip-info.org/wiki/view/Asterisk+codecs but I still have questions about core show translation. How are values replied by core show translation computed in the the first place ? I've got 2 machines using a 3GHz Intel CPU : 1 is a Xeon, 1 is a Pentium 4 (gathered with cat /proc/cpuinfo) The Xeon machine is showing, for instance: gsm ulaw - 1601 The other shows: gsm ulaw - 2 Why are these values so different ? Is it correct to say if core show translation is showing a 4 digits value in its matrix, then the translation path between corresponding codecs is unusable. Regards Perhaps the Xeon machine needs some optimization since a ulaw-to-gsm translation take 1.6 seconds for 1 second of data as opposed to .002 seconds on the P4 machine. -- _ -- 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] AstriCon update - less than two weeks!
[ text with links can be found on http://blogs.digium.com/2010/10/14/astricon-update/ ] AstriCon is less than two weeks away! If you haven’t booked your flight to Washington DC, now’s your chance! The main hotel (the Gaylord) is pretty booked, but that’s OK – there are still rooms a few hundred feet away at some of the hotels around the complex (Aloft, Wyndham, Hampton Inn, Residence Inn) and there are more hotels within a short drive/cab of the venue. Speakers at AstriCon We’ve got some great last-minute speakers to announce – I’m pleased to say that Ruben Sousa will be giving a talk on one of the largest open- source Asterisk installations in the world (100,000 users, 184 servers) which is arrayed across the University system in Portugal. We have a really solid line-up this year of talks focused on security and scalability from Kevin Lynn, Sandro Gauci, the Great Olle Johansson, and more! Many of the most active community developers, integrators, and speakers will be on hand, along with some very interesting announcements from Digium including the yearly roadmap and status update from the Digium engineering group – don’t miss out on hearing what’s new and what’s coming up! With the huge number of features that have been added to 1.8, it’s possible that you’ll learn from someone at the show how the newest release of Asterisk can benefit your organization in a way you never expected. Win an Apple iPAD at the AstriCon Ringer Rodeo! “Saddle up pardners!” and prepare to reap the benefit of being the fastest Asterisk geek in the West… make that “East”. Well, the fastest Asterisk geek at AstriCon, anyways. We’ve devised a stunningly simple contest (it’s almost too simple) to give away an Apple iPad – and with only a low number of opportunities to compete (during the show party), this could be the easiest iPad you win this millennium. Second and third prizes are a Polycom IP-650 deskphone and aPanasonic KX-TGP500 DECT wireless phone. The contest will demand the ability to hook up two SIP phones and an IAX ‘trunk’, in addition to a small amount of dialplan programming. You’ll be given all the details you need and you will not have to know how to set up the actual phones – we’ve done that part for you. Just like last year (when we gave away an unlocked HTC Hero Android phone) there will be a number of timed rounds with the winning time in each round going on to the leader board – the dude (or dudette) with the fastest time on that board at the end of the party will be presented with said iPad at the end of conference session in addition to being admired and envied in equal measure by the gathered crowd! David Duffett will be the chief rodeo wrangler – at the all-conference party on Wednesday night, look for the man in pinstriped jacket and cane. Etc. etc. I’ve been told there is a Water Taxi from the hotel dock to old Town Alexandria. This is a great place to have dinner, wander around, and see some of the sights of the DC area. And a boat taxi is always fun! A mile or two up the river from Alexandria, there is also lots to do in Washington, DC – it’s a great time of year to be there when it’s not too hot, and not too cold. Developers: there is the AstriDevCon on Friday from around 8:00 to 5:00, which will focus on hard-core code development and discussion of particular issues in the codebase. If you speak C and find yourself typing “make config” in your head when you meet someone new, this is probably where you’ll find some interesting discussion. Sign up here if you haven’t already. We do ask this to be a developer-only session, so please be familiar with the code and the issue tracker if you plan to attend. Also, don’t forget there is the chance to take the dCAP test at AstriCon – kill two birds with one stone! See the AstriCon website for more details. Just two more weeks! See everyone there. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default MOH not working on 1.6.1
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, October 14, 2010 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default MOH not working on 1.6.1 2010/10/14 Olivier oza_4...@yahoo.fr Hello, I've configured with the very same script 1 Intel Xeon and 1 Intel Pentium4 machines. On one MOH is working properly On the other, I can read on console, lines such as those bellow but I can't hear anything. In which direction, should I further investigate ? If this help, here is my setup: me ---PSTN-ISDN Patton 4638 ---SIP--- Asterisk 1.6.1.18 -- Started music on hold, class 'default', on SIP/patton-002b == Using SIP RTP CoS mark 5 == Extension Changed 249[subs] new state Ringing for Notify User 749 == Extension Changed 249[subs] new state Ringing for Notify User 750 -- SIP/249-002c is ringing Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043637, ts 000160, len 000160) Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043638, ts 000320, len 000160) Thanks PS: I used the standard i386 Lenny image on the Xeon machine. Should I favor another image, such as amd64 or em64t, instead ? To answer this one intelligently, you need to provide a little more info about the xeon machine since they come in many flavors. -- _ -- 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] Routers that do not show external IPs...
I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register. The problem is that I have no audio because Asterisk thinks that the phone is on the internal network and does not use the NAT and externip settings. How do you deal with this kind of router so you can have external phones? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding callerID
We have a T1 of sorts, ATT ip flex reach basically voip over a t1 line i think. I will ask them and see what they say, I'm already able to set our outgoing callerID to any number we own, just no other ones.. there some other way to handle this? It depends on the technology and the carrier. A simple POTS line and you're out of luck. If you have a T1 (i.e. ISDN-PRI) and a co-operative carrier, it may just work or they may enable it if requested. You could always use a co-operative SIP carrier (like Vitelity). A penny or 2 per minute will keep your someone happy. -- Gerard Saraber Network Admin. Rarcoa, Inc (630) 654-2580 x199 (630) 654-3556 (fax) (630) 915-4122 (cell) -- _ -- 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] Explain core show translation
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, October 14, 2010 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Explain core show translation 2010/10/14 Danny Nicholas da...@debsinc.com _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, October 14, 2010 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Explain core show translation Hi, I've read in http://www.voip-info.org/wiki/view/Asterisk+codecs but I still have questions about core show translation. How are values replied by core show translation computed in the the first place ? I've got 2 machines using a 3GHz Intel CPU : 1 is a Xeon, 1 is a Pentium 4 (gathered with cat /proc/cpuinfo) The Xeon machine is showing, for instance: gsm ulaw - 1601 The other shows: gsm ulaw - 2 Why are these values so different ? Is it correct to say if core show translation is showing a 4 digits value in its matrix, then the translation path between corresponding codecs is unusable. Regards Perhaps the Xeon machine needs some optimization since a ulaw-to-gsm translation take 1.6 seconds for 1 second of data as opposed to .002 seconds on the P4 machine. Where this data comes from in the first place ? Is it computed each time core show translation is typed ? What does core show translation recalc 60 add to core show translation ? Ahh - questions that make me read. The core show translation invokes the translate.c module. If you do c s t it does a one shot display of the current values. If you do c s t r 60 it recalculates and redisplays the values every 60 seconds. If your machine is a variable load state, you could get significantly different output. If your machine is running at a 20% load, it's probably not going to vary very much. -- _ -- 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 forwarding callerID
On 10/13/10 14:52, Danny Nicholas wrote: I think FOLLOWME is going to fix this for you Can you elaborate please? is this a feature from our carrier? or something that will be built into asterisk? sounds like a useful fix :) -- Gerard Saraber Network Admin. Rarcoa, Inc (630) 654-2580 x199 (630) 654-3556 (fax) (630) 915-4122 (cell) -- _ -- 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] Routers that do not show external IPs...
On 10-10-14 12:18 PM, Carlos Chavez wrote: I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register. The problem is that I have no audio because Asterisk thinks that the phone is on the internal network and does not use the NAT and externip settings. How do you deal with this kind of router so you can have external phones? Typically that is an option you can turn off. It is meant to help with SIP translations and such through the router, but as you're finding out, they typically just get in the way. Check through the web interface/configuration and see if there is anything about VoIP or SIP support in the router, and disable it. Leif. -- _ -- 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] Default MOH not working on 1.6.1
2010/10/14 Danny Nicholas da...@debsinc.com -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Thursday, October 14, 2010 11:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Default MOH not working on 1.6.1 2010/10/14 Olivier oza_4...@yahoo.fr Hello, I've configured with the very same script 1 Intel Xeon and 1 Intel Pentium4 machines. On one MOH is working properly On the other, I can read on console, lines such as those bellow but I can't hear anything. In which direction, should I further investigate ? If this help, here is my setup: me ---PSTN-ISDN Patton 4638 ---SIP--- Asterisk 1.6.1.18 -- Started music on hold, class 'default', on SIP/patton-002b == Using SIP RTP CoS mark 5 == Extension Changed 249[subs] new state Ringing for Notify User 749 == Extension Changed 249[subs] new state Ringing for Notify User 750 -- SIP/249-002c is ringing Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043637, ts 000160, len 000160) Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043638, ts 000320, len 000160) Thanks PS: I used the standard i386 Lenny image on the Xeon machine. Should I favor another image, such as amd64 or em64t, instead ? To answer this one intelligently, you need to provide a little more info about the xeon machine since they come in many flavors. $ cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Xeon(TM) CPU 3.00GHz stepping: 1 cpu MHz : 2992.566 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 apicid : 0 initial apicid : 0 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl cid cx16 xtpr bogomips: 5990.39 clflush size: 64 power management: processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Xeon(TM) CPU 3.00GHz stepping: 1 cpu MHz : 2992.566 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 apicid : 1 initial apicid : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl cid cx16 xtpr bogomips: 5984.99 clflush size: 64 power management: -- _ -- 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] Routers that do not show external IPs...
On Thu, Oct 14, 2010 at 11:18:59AM -0500, Carlos Chavez wrote: I opened up the ports on the router and my phone can register. The problem is that I have no audio because Asterisk thinks that the phone is on the internal network and does not use the NAT and externip settings. How do you deal with this kind of router so you can have external phones? By replacing the crappy router with a decent one. An other good reason for replacement would be the lack of WPA AES on it. Yet another reason to replace it is that your labour costs are probably higher than this 90$ router. But what ports did you open? Only sip or also the RTP ports? -- 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] call forwarding callerID
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Thursday, October 14, 2010 11:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding callerID
On 10/13/10 14:52, Danny Nicholas wrote:
I think FOLLOWME is going to fix this for you
Can you elaborate please? is this a feature from our carrier? or
something that will be built into asterisk? sounds like a useful fix :)
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)
Check this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
A simpler solution (perhaps) would be a forwarding context like this
[forward-with-announce]
Exten = s,1,dial(DAHDI/g1/w${ARG1},30,mKkTt)
Exten = s,n,playback(followme/call-from)
Exten = s,n,SayDigits(${ARG2})
Exten = 393,1,Set(ARG1=201212)
Exten = 393,2,Set(ARG2=${EXTEN})
Exten = 393,3,Goto(forward-with-announce,s,1)
Dependent on carrier and other considerations, you can also spoof the
caller-id. That's a different google-search.
--
_
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Re: [asterisk-users] Routers that do not show external IPs...
On Thu, 2010-10-14 at 18:35 +0200, Daniel Tryba wrote: On Thu, Oct 14, 2010 at 11:18:59AM -0500, Carlos Chavez wrote: I opened up the ports on the router and my phone can register. The problem is that I have no audio because Asterisk thinks that the phone is on the internal network and does not use the NAT and externip settings. How do you deal with this kind of router so you can have external phones? By replacing the crappy router with a decent one. An other good reason for replacement would be the lack of WPA AES on it. Yet another reason to replace it is that your labour costs are probably higher than this 90$ router. But what ports did you open? Only sip or also the RTP ports? I opened SIP and RTP, after that I put the server on the DMZ but I still get no audio on the external phone. My problem is that we do not administer the customers network and the just bought their brand new super router. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routers that do not show external IPs...
the sip port and rtp port range 1 to 2 i guess... - Original Message - From: Daniel Tryba dan...@tryba.nl To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 14, 2010 6:35 PM Subject: Re: [asterisk-users] Routers that do not show external IPs... On Thu, Oct 14, 2010 at 11:18:59AM -0500, Carlos Chavez wrote: I opened up the ports on the router and my phone can register. The problem is that I have no audio because Asterisk thinks that the phone is on the internal network and does not use the NAT and externip settings. How do you deal with this kind of router so you can have external phones? By replacing the crappy router with a decent one. An other good reason for replacement would be the lack of WPA AES on it. Yet another reason to replace it is that your labour costs are probably higher than this 90$ router. But what ports did you open? Only sip or also the RTP ports? -- 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 __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a été vérifié par ESET NOD32 Antivirus. http://www.eset.com __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a été vérifié par ESET NOD32 Antivirus. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routers that do not show external IPs...
http://www.showmyip.com http://www.whatismyip.com http://www.maxmind.com - Original Message - From: Carlos Chavez cur...@telecomabmex.com To: Asterisk asterisk-users@lists.digium.com Sent: Thursday, October 14, 2010 6:18 PM Subject: [asterisk-users] Routers that do not show external IPs... -- _ -- 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 __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a iti virifii par ESET NOD32 Antivirus. http://www.eset.com __ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 5531 (20101014) __ Le message a �t� v�rifi� par ESET NOD32 Antivirus. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routers that do not show external IPs...
You are missing the point completely. Maybe I did not explain myself clearly. The problem is that when you connect to the server from outside the network (Internet), Asterisk does not see the IP address of the device, it thinks the device is connecting from the IP address of the router itself (192.168.X.X). This means that even if you have externip, nat=yes and localnet configured properly, Asterisk will think that the phone is on the internal network (because of localnet) and will NOT use the external IP address to communicate with the external phone. This is not a problem with Asterisk. The router rewrites all external connections with its own IP so even a SSH connection will seem to be coming from the router (the 'w' command will say you are connected from the router and not from the IP address of your Internet connection). OMG thats the worst kind of doing everything wrong as possible i ever heard of. I wonder if this router works in ANY way. You can try to turn of these ALG features which the router have build in and also these SPI (statefull packet inspection). best regards stefan -- _ -- 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] Routers that do not show external IPs...
- Stefan Schmidt s...@sil.at wrote: This is not a problem with Asterisk. The router rewrites all external connections with its own IP so even a SSH connection will seem to be coming from the router (the 'w' command will say you are connected from the router and not from the IP address of your Internet connection). Isn't this the purpose and definition of NAT? Your private network sits behind the NAT while outbound traffic has it's source IP (maybe port...) rewritten to that of the external IP of the router? This holds true if the router's public interface is on another RFC1918 private network. OMG thats the worst kind of doing everything wrong as possible i ever heard of. I wonder if this router works in ANY way. Uhm... You can try to turn of these ALG features which the router have build in and also these SPI (statefull packet inspection). NAT isn't exactly an ALG... --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] call forwarding callerID
ah-ha,
thank you very much, that's what I found when googling, I'll ask my user
and see if Asterisk announcing the call is acceptable to him, if I can't
spoof the callerID.
Followme would alternatively work pretty well, press 1 to accept the
call etc. is a pretty nice feature, I'll see if that works for him.
Thanks!
On 10/14/10 11:41, Danny Nicholas wrote:
Check this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
A simpler solution (perhaps) would be a forwarding context like this
[forward-with-announce]
Exten = s,1,dial(DAHDI/g1/w${ARG1},30,mKkTt)
Exten = s,n,playback(followme/call-from)
Exten = s,n,SayDigits(${ARG2})
Exten = 393,1,Set(ARG1=201212)
Exten = 393,2,Set(ARG2=${EXTEN})
Exten = 393,3,Goto(forward-with-announce,s,1)
Dependent on carrier and other considerations, you can also spoof the
caller-id. That's a different google-search.
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Create channel bank with TDMoE
Hi Karim, Here you find a basic example for configuration: http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE First step is to connect two asterisk using this basic configuration. did you get that ? There are other possibilites when using two asterisks like connecting them using IAX trunking. Luis A P Barbosa 2010/10/14 Karim Davoodi karimdavo...@gmail.com Tanks I want to create a channel bank with TDMoE. I have not to buy a product. Best, 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 -- _ -- 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 forwarding callerID
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Thursday, October 14, 2010 1:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding callerID
ah-ha,
thank you very much, that's what I found when googling, I'll ask my user
and see if Asterisk announcing the call is acceptable to him, if I can't
spoof the callerID.
Followme would alternatively work pretty well, press 1 to accept the
call etc. is a pretty nice feature, I'll see if that works for him.
Thanks!
On 10/14/10 11:41, Danny Nicholas wrote:
Check this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
A simpler solution (perhaps) would be a forwarding context like this
[forward-with-announce]
Exten = s,1,dial(DAHDI/g1/w${ARG1},30,mKkTt)
Exten = s,n,playback(followme/call-from)
Exten = s,n,SayDigits(${ARG2})
Exten = 393,1,Set(ARG1=201212)
Exten = 393,2,Set(ARG2=${EXTEN})
Exten = 393,3,Goto(forward-with-announce,s,1)
Dependent on carrier and other considerations, you can also spoof the
caller-id. That's a different google-search.
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)
--
FWIW, there are also professional spoofing services but they cost
$0.02-$0.05/minute.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Audiocodes firmware
Oh right... MP-118 Thanks. On 10/14/2010 03:38 PM, Bryant Zimmerman wrote: For which device models? From: "Mark Murawski" markm-li...@intellasoft.net Sent: Thursday, October 14, 2010 3:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Audiocodes firmware Does anyone have links to the most recent audiocodes firmware? -- _ -- 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] Routers that do not show external IPs...
Am 14.10.2010 21:06, schrieb Tim Nelson: The TCP header is exactly what the NAT changes, no? --Tim to the outside yes but not inside. for example thats how a typical nat table looks like. (its from a zyxel adsl router with nat) Nat session table== Slot Prot Int-IP :PortOut-IP :PortExt-IP :Port Idle === 45 TCP 192.168.0.1:6023 xxx :6023 zzz :44450 0 121 UDP 192.168.0.129 :5061 xxx :10619 sip1:5060 4 135 UDP 192.168.0.129 :5060 xxx :10618 sip2:5060 3 Summary information= 192.168.0.129 is a sip phone with 2 accounts registered to sip1 and sip2. if i take a look at sip1 i will see the package from ip xxx port 10619. Ofcourse its behind nat but i will see in the contact header 192.168.0.129 port 5061. With sip ALG active also the contact header would be changed to xxx port 10619. Other way if i look on the phone i see the answer from sip1 directly as a message from sip1 port 5060 and not from xxx port 10619 or 192.168.0.1. several things wont work if you dont get the original source ip through a nat router. thats how i have learned it and see it everyday in practice. best regards stefan -- _ -- 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] Default MOH not working on 1.6.1
2010/10/14 Olivier oza_4...@yahoo.fr 2010/10/14 Olivier oza_4...@yahoo.fr Hello, I've configured with the very same script 1 Intel Xeon and 1 Intel Pentium4 machines. On one MOH is working properly On the other, I can read on console, lines such as those bellow but I can't hear anything. In which direction, should I further investigate ? If this help, here is my setup: me ---PSTN-ISDN Patton 4638 ---SIP--- Asterisk 1.6.1.18 -- Started music on hold, class 'default', on SIP/patton-002b == Using SIP RTP CoS mark 5 == Extension Changed 249[subs] new state Ringing for Notify User 749 == Extension Changed 249[subs] new state Ringing for Notify User 750 -- SIP/249-002c is ringing Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043637, ts 000160, len 000160) Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043638, ts 000320, len 000160) Thanks PS: I used the standard i386 Lenny image on the Xeon machine. Should I favor another image, such as amd64 or em64t, instead ? If this matters, I must also add MOH is triggered here by Queue application. -- _ -- 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] Default MOH not working on 1.6.1
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, October 14, 2010 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default MOH not working on 1.6.1 2010/10/14 Olivier oza_4...@yahoo.fr 2010/10/14 Olivier oza_4...@yahoo.fr Hello, I've configured with the very same script 1 Intel Xeon and 1 Intel Pentium4 machines. On one MOH is working properly On the other, I can read on console, lines such as those bellow but I can't hear anything. In which direction, should I further investigate ? If this help, here is my setup: me ---PSTN-ISDN Patton 4638 ---SIP--- Asterisk 1.6.1.18 -- Started music on hold, class 'default', on SIP/patton-002b == Using SIP RTP CoS mark 5 == Extension Changed 249[subs] new state Ringing for Notify User 749 == Extension Changed 249[subs] new state Ringing for Notify User 750 -- SIP/249-002c is ringing Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043637, ts 000160, len 000160) Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043638, ts 000320, len 000160) Thanks PS: I used the standard i386 Lenny image on the Xeon machine. Should I favor another image, such as amd64 or em64t, instead ? If this matters, I must also add MOH is triggered here by Queue application. I assume MOH is working on Pentium 4 and failing on Xeon? Try this snippet Exten = 664,1,answer exten = 664,n,SetMusicOnHold(default) exten = 664,n,WaitMusicOnHold(20) exten = 664,n,Background(vm-goodbye) exten = 664,n,Hangup This should play your default MOH for 20 seconds, then say goodbye and hangup. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default MOH not working on 1.6.1
2010/10/14 Danny Nicholas da...@debsinc.com -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Thursday, October 14, 2010 3:34 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Default MOH not working on 1.6.1 2010/10/14 Olivier oza_4...@yahoo.fr 2010/10/14 Olivier oza_4...@yahoo.fr Hello, I've configured with the very same script 1 Intel Xeon and 1 Intel Pentium4 machines. On one MOH is working properly On the other, I can read on console, lines such as those bellow but I can't hear anything. In which direction, should I further investigate ? If this help, here is my setup: me ---PSTN-ISDN Patton 4638 ---SIP--- Asterisk 1.6.1.18 -- Started music on hold, class 'default', on SIP/patton-002b == Using SIP RTP CoS mark 5 == Extension Changed 249[subs] new state Ringing for Notify User 749 == Extension Changed 249[subs] new state Ringing for Notify User 750 -- SIP/249-002c is ringing Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043637, ts 000160, len 000160) Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043638, ts 000320, len 000160) Thanks PS: I used the standard i386 Lenny image on the Xeon machine. Should I favor another image, such as amd64 or em64t, instead ? If this matters, I must also add MOH is triggered here by Queue application. I assume MOH is working on Pentium 4 and “failing” on Xeon? Try this snippet Exten = 664,1,answer exten = 664,n,SetMusicOnHold(default) exten = 664,n,WaitMusicOnHold(20) exten = 664,n,Background(vm-goodbye) exten = 664,n,Hangup This should play your default MOH for 20 seconds, then say goodbye and hangup. I'll give it a try ASAP (this WE ) -- _ -- 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] Audiocodes firmware
On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski markm-li...@intellasoft.net wrote: Does anyone have links to the most recent audiocodes firmware? Why not contact Audiocodes? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes firmware
Because audiocodes does not provide support to end users and will tell you to contact your vendor that sold you the box. The problem is, the vendor that sold me the box is really hard to deal with and has been brushing me off all week on getting firmware. On 10/14/2010 05:14 PM, Paul Belanger wrote: On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski markm-li...@intellasoft.net wrote: Does anyone have links to the most recent audiocodes firmware? Why not contact Audiocodes? -- _ -- 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] Audiocodes firmware
From: Paul Belanger paul.belan...@polybeacon.com Sent: Thursday, October 14, 2010 5:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audiocodes firmware On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski markm-li...@intellasoft.net wrote: Does anyone have links to the most recent audiocodes firmware? Why not contact Audiocodes? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- You have to normally get the Audiocodes firmware from your reseller, or you have to buy a support contract on the device to get current firmware. Audiocodes for some reason does not offer simple just download the current version and install it as an option. They have stated that too many people tend to mess up firmware upgrades so they want you to have the support contract from them or your resellar. It is really hard to select a bin file and hit update without shutting off your device until it's done. $$$ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes firmware
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Thursday, October 14, 2010 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audiocodes firmware _ From: Paul Belanger paul.belan...@polybeacon.com Sent: Thursday, October 14, 2010 5:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audiocodes firmware On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski markm-li...@intellasoft.net wrote: Does anyone have links to the most recent audiocodes firmware? Why not contact Audiocodes? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- You have to normally get the Audiocodes firmware from your reseller, or you have to buy a support contract on the device to get current firmware. Audiocodes for some reason does not offer simple just download the current version and install it as an option. They have stated that too many people tend to mess up firmware upgrades so they want you to have the support contract from them or your resellar. It is really hard to select a bin file and hit update without shutting off your device until it's done. $$$ In Perfectland you could tell Audiocodes that your dealers is being an SOB and they would refer you to somebody else or make his happy hinney help you. Sorry it's not. -- _ -- 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] Audiocodes firmware
On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski markm-li...@intellasoft.net wrote: Because audiocodes does not provide support to end users and will tell you to contact your vendor that sold you the box. That is ridiculous, how hard is it to provide a download link and disclaimer about no support. Unless Audiocodec's simply wants to charge you more money. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some give 603 Declined
Here is the sip log ns*CLI sip set debug peer hkbn2b SIP Debugging Enabled for IP: 203.80.89.139:5060 [Oct 15 06:35:19] NOTICE[2462]: chan_sip.c:18334 handle_response_register: Outbound Registration: Expiry for sip.voipuser.org is 120 sec (Scheduling reregistration in 105 s) == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [8935944...@dlpn_dp1:1] Dial(SIP/6100-0006, SIP/35944...@hkbn2b,,r) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 113.253.226.153 port 10650 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 203.80.89.139:5060: INVITE sip:35944...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport Max-Forwards: 70 From: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as12eb85f9 To: sip:35944...@s2hkbntel.net:5060 Contact: sip:3594410...@113.253.226.153 sip%3a3594410...@113.253.226.153 Call-ID: 3f603bea2560e9b836ea250932486...@s2hkbntel.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.12 Date: Thu, 14 Oct 2010 22:35:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 241 v=0 o=root 316173620 316173620 IN IP4 113.253.226.153 s=Asterisk PBX 1.6.2.12 c=IN IP4 113.253.226.153 t=0 0 m=audio 10650 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 35944...@hkbn2b --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 100 Trying t: sip:35944...@s2hkbntel.net:5060 f: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as12eb85f9 i: 3f603bea2560e9b836ea250932486...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.226.153:5060 ;received=113.253.226.174;rport;branch=z9hG4bK1880eaca Server: MCS5x00_3.0 k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 487 Request Terminated t: sip:35944...@s2hkbntel.net:5060;tag=781480306 f: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as12eb85f9 i: 3f603bea2560e9b836ea250932486...@s2hkbntel.net CSeq: 102 INVITE v: SIP/2.0/UDP 113.253.226.153:5060 ;received=113.253.226.174;rport;branch=z9hG4bK1880eaca k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (8 headers 0 lines) --- Transmitting (NAT) to 203.80.89.139:5060: ACK sip:35944...@s2hkbntel.net:5060 SIP/2.0 Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport Max-Forwards: 70 From: ck...@mobile sip:3594410...@s2hkbntel.netsip%3a3594410...@s2hkbntel.net ;tag=as12eb85f9 To: sip:35944...@s2hkbntel.net:5060;tag=781480306 Contact: sip:3594410...@113.253.226.153 sip%3a3594410...@113.253.226.153 Call-ID: 3f603bea2560e9b836ea250932486...@s2hkbntel.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.12 Content-Length: 0 --- Scheduling destruction of SIP dialog ' 3f603bea2560e9b836ea250932486...@s2hkbntel.net' in 6400 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [8935944...@dlpn_dp1:2] Hangup(SIP/6100-0006, ) in new stack == Spawn extension (DLPN_DP1, 8935944101, 2) exited non-zero on 'SIP/6100-0006' [Oct 15 06:35:23] NOTICE[2462]: chan_sip.c:11601 sip_reregister:-- Re-registration for 8887109...@sip.pennytel.com Reliably Transmitting (NAT) to 203.80.89.139:5060: OPTIONS sip:s2hkbntel.net SIP/2.0 Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK703ea06a;rport Max-Forwards: 70 From: asterisk sip:aster...@sip.etransmed.netsip%3aaster...@sip.etransmed.net ;tag=as1d0ccbd8 To: sip:s2hkbntel.net Contact: sip:aster...@113.253.226.153 sip%3aaster...@113.253.226.153 Call-ID: 67f6129e02db3377276c62f209913...@sip.etransmed.net CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.12 Date: Thu, 14 Oct 2010 22:35:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 100 Trying t: sip:s2hkbntel.net f: asterisk sip:aster...@sip.etransmed.netsip%3aaster...@sip.etransmed.net ;tag=as1d0ccbd8 i: 67f6129e02db3377276c62f209913...@sip.etransmed.net CSeq: 102 OPTIONS v: SIP/2.0/UDP 113.253.226.153:5060 ;received=113.253.226.174;rport;branch=z9hG4bK703ea06a Server: MCS5x00_3.0 k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:203.80.89.139:5060 --- SIP/2.0 404 Not Found t: sip:s2hkbntel.net;tag=820879923 f: asterisk sip:aster...@sip.etransmed.netsip%3aaster...@sip.etransmed.net ;tag=as1d0ccbd8 i: 67f6129e02db3377276c62f209913...@sip.etransmed.net CSeq: 102 OPTIONS v: SIP/2.0/UDP 113.253.226.153:5060 ;received=113.253.226.174;rport;branch=z9hG4bK703ea06a k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0
Re: [asterisk-users] Some give 603 Declined
On Thu, Oct 14, 2010 at 6:46 PM, asterisk asterisk aster...@ck-lee.com wrote: Here is the sip log 487 Request Terminated, the far end is killing your session. Talk to your ITSP. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes firmware
We are being forced to move away from audiocodes ATA's because they refuse to fix a few minor bugs unless we commit to a 1000 piece order. This is on their 2 port ATA's. Their response to us is that ATA's are intended for serious carriers that are using them in conjunction with their higher end gateways. And we use their PRI gateways and a few of their 4 and 8 port gateways but we can't user their 2 ports. From: Paul Belanger paul.belan...@polybeacon.com Sent: Thursday, October 14, 2010 6:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audiocodes firmware On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski markm-li...@intellasoft.net wrote: Because audiocodes does not provide support to end users and will tell you to contact your vendor that sold you the box. That is ridiculous, how hard is it to provide a download link and disclaimer about no support. Unless Audiocodec's simply wants to charge you more money. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Audiocodes firmware
On Oct 14, 2010, at 5:40 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski markm-li...@intellasoft.net wrote: Because audiocodes does not provide support to end users and will tell you to contact your vendor that sold you the box. That is ridiculous, how hard is it to provide a download link and disclaimer about no support. Unless Audiocodec's simply wants to charge you more money. Sounds like Cisco... Thanks, --Warren Selby -- _ -- 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] fraud advice
Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was taken advantage of over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN trunk, and of course there was no warning until the bill came today. Unfortunately the bill only covered the first few days of this fiasco, and was only $700. I am afraid the one that is on the way will be tens of thousands. ONE CALL on the bill that just arrived was $200 (80 minutes to Sierra Leone). I'm sure this started out as a single scan. It must have been posted, because I have at least ten IP addresses now that were placing calls via the same peer. They are from all over the world. So what is the accepted procedure? I'm in the US Virgin Islands, so do I go to the FBI? Police? Is their some telecom fraud body to report such things to? Does any one ever get any relief from such events? I'm basically sick to my stomach right now. j -- _ -- 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] fraud advice
As a practical matter, on anything that can generate endless billings, there should be a dumb trap that compares current usage to history (last month) and if usage exceeds 2/1 or 3/1 for instance then usage is choked or denied enough to cause the user to complain or perhaps generate a message to call customer support, (or call your cell phone!) Then if it is valid, raise last month's reference enough to let current calling continue. If it isn't valid you have found a problem and saved your or your customer's caboose. As to who to complain to, gather all info possible and report to everyone you can find. Someone may investigate, but there isn't likely anyone who will absolve the problem. Some will just take the report and ... as far as you are concerned, do nothing. There isn't much a local police dept. can do about a hacker in Western Slobovia cracking your server. Generally the FBI doesn't take matters of less than $10,000. But it sounds like you may meet that test. But they could take months or years or never finding the culprit and finding the culprit will likely net you nothing financial for you will be 1/10,000 of the fraud they did. This is a problem like spam in email. But this has cash costs to the server operator/customer. Passwords need to be un-crack-able, and there should be usage alarms, as described above. Depending on the situation even a single counter to your upstream billable sip server for all usage would likely trip on excessive usage and save your bacon. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, October 14, 2010 8:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] fraud advice Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was taken advantage of over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN trunk, and of course there was no warning until the bill came today. Unfortunately the bill only covered the first few days of this fiasco, and was only $700. I am afraid the one that is on the way will be tens of thousands. ONE CALL on the bill that just arrived was $200 (80 minutes to Sierra Leone). I'm sure this started out as a single scan. It must have been posted, because I have at least ten IP addresses now that were placing calls via the same peer. They are from all over the world. So what is the accepted procedure? I'm in the US Virgin Islands, so do I go to the FBI? Police? Is their some telecom fraud body to report such things to? Does any one ever get any relief from such events? I'm basically sick to my stomach right now. j -- _ -- 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] fraud advice
Jeff, I suggest talking to your PSTN/VoIP provider. We had a large amount going through TATA communications and have not accepted their word for payment because they had a duty to not allow traffic if our credit went down to $1k while the calls charged were actually more than that. Unfortunately, probably there is no one you can complain to. But it also sickens me at how badly Asterisk is made to not cope with situations like this and worse than that is FreePBX. I suggest checking your contract terms with your provider as they might have some sort of restrictions. At the very least PSTN providers try to bring the price per minute lowered to their buy rate which is usually less than half of the original bill. Regards, Bruce On Thu, Oct 14, 2010 at 9:10 PM, Jeff LaCoursiere j...@sunfone.com wrote: Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was taken advantage of over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN trunk, and of course there was no warning until the bill came today. Unfortunately the bill only covered the first few days of this fiasco, and was only $700. I am afraid the one that is on the way will be tens of thousands. ONE CALL on the bill that just arrived was $200 (80 minutes to Sierra Leone). I'm sure this started out as a single scan. It must have been posted, because I have at least ten IP addresses now that were placing calls via the same peer. They are from all over the world. So what is the accepted procedure? I'm in the US Virgin Islands, so do I go to the FBI? Police? Is their some telecom fraud body to report such things to? Does any one ever get any relief from such events? I'm basically sick to my stomach right now. j -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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] Audiocodes firmware
Crazy. What do you plan on using for an ATA now? The problems I'm having are getting 500 "Server Internal Error" on just about every other call placed out of this mp-118. The box has been installed and in use for quite some time and recently started having problems. Reboots, etc don't make a difference. I noticed it had newer firmware than what I had on some other boxes that had no issues whatsoever. I do have a 5.80 firmware I had downloaded a while back and put that on. Now the internal server errors are happening on 70-80% of sip-pstn calls. pstn-sip calls seem to be coming in just fine. Ever since I did the firmware downgrade, now my ssh sessions to the box get disconnected after about 30 seconds with invalid packet errors. I've had problems with earlier firmware as well... once the 5.x firmware started shipping on audiocodes it seemed they were just about DOA. The web interface worked but nothing else worked right. Perfectly working configurations on other boxes that were copied to the new boxes with new firmware would just fail in various ways... disconnect supervision not working, internal routing not working. Finally I managed to get a hold of the 5.80 firmware which got rid of all those problems. Now I'm stuck again. I have a box in service that's having problems and I can't get new firmware. On 10/14/2010 07:17 PM, Bryant Zimmerman wrote: We are being forced to move away from audiocodes ATA's because they refuse to fix a few minor bugs unless we commit to a 1000 piece order. This is on their 2 port ATA's. Their response to us is that ATA's are intended for serious carriers that are using them in conjunction with their higher end gateways. And we use their PRI gateways and a few of their 4 and 8 port gateways but we can't user their 2 ports. NetVanta 6330 From: "Paul Belanger" paul.belan...@polybeacon.com Sent: Thursday, October 14, 2010 6:43 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audiocodes firmware On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski markm-li...@intellasoft.net wrote: Because audiocodes does not provide support to end users and will tell you to contact your vendor that sold you the box. That is ridiculous, how hard is it to provide a download link and disclaimer about no support. Unless Audiocodec's simply wants to charge you more money. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone
What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Here is little more console output: localhost*CLI queue show Sales Saleshas 0 calls (max 10) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers localhost*CLI core show channels Channel Location State Application(Data) SIP/101-000b s...@macro-tl-userexten Up VoiceMailMain(101) 1 active channel 1 active call 'core show channels' show SIP/101 is use but 'queue show' does not. -Matt -- _ -- 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
