[asterisk-users] Re: Digium G.729 codec binaries updated for Asterisk 1.4 beta
On 2006-09-23 12:43:32 -0700, Kevin P. Fleming [EMAIL PROTECTED] said: - Matt Riddell (IT) [EMAIL PROTECTED] wrote: Also, are you referring to newer ones than the 1.4 downloads that were available a couple of days ago or do you mean people running the 1.2 versions? The versions that were initially posted as compatible with Asterisk 1.4 became incompatible just before beta2 was released, so these versions are compatible with beta2. How about some PowerPC love? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody have the opvx1200.c driver?
On Tue, Sep 26, 2006 at 08:43:11PM -0700, Nick Ellson wrote: The link is not working at OpenVox. There's a download link in the bottom of the page, that leads to: http://www.openvox.com.cn/members_downloads.php . That page has the A1200P device driver as a download item (not just for members). That page also reads: if you are using pop-up block tools, such as google toolbar, please close the pop-up block function before download. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] my (SIP) INVITE is ignored
Hi,
I'm struggling with this kind of problem:
my hardware sip phone is registering to Asterisk 1.2.10 successfully,
but when I send INVITE to server - it receives the packet but (in sip
debug mode) I see: 'Ignoring this INVITE request'.
While searching in 'chan_sip.c' I've found that this message shows up if
variable 'ignore' is being set. This is when
if (p-ocseq (p-ocseq != seqno)) {
ignore = 1;
}
unfortunately there aren't any comments around, so anyone could explain
what exactly is happening?
regards,
L.
___
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[asterisk-users] outgoing call problem
Hi. I'm having a bit of trouble with outgoing calls on zap channels. When i try to make an outgoing call asterisk doesn't detect if the other party answers. When i run 'show channels verbose' in CLI asterisk tells me that the respective channles are in ringing state like this: Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo Zap/19-1 agentie s 1 Dialing AppDial (Outgoing Line) 00726710704 (None) Zap/15-1 int_omg 00726710704 5 Ring Dial Zap/g5/0726710704||T 00:00:14 (None) although i can speak to the called party. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voip Buster - CID
Hi List! Is there any way to set outgoing CID number when making VoIP calls using VoIP Buster? I have search on their forum and I couldn't find anything useful. There is no support mail on their web pages :(( P.S. I use them because they are cheep and sound quality is satisfying -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Asterisk 1.4-beta2 to work with jingle
Hi, I installed this beta and I'm trying to use the jingle integration, following the steps in this wiki http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk, but I'm having some problem. I registered even a SIP than a IAX user; when I try to call the jingle user connected via libjingle from Xlite i receive a call not approved message and no response from asterisk; when I try from iaxcomm I received from asterisk these errors: [Sep 26 16:20:51] WARNING[8742]: channel.c:2842 ast_request: No channel type registered for 'Jingle' [Sep 26 16:20:51] WARNING[8742]: app_dial.c:1077 dial_exec_full: Unable to create channel of type 'Jingle' (cause 66 - Channel not implemented) Hera are mi conf files: sip.conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes dtmfmode=rfc2833 relaxdtmf=no disallow=all allow=ulaw allow=alaw allow=gsm maxexpirey=30 defaultexpirey=180 canreinvite=yes nat=0 UserAgent=Asterisk [raffo6] type=friend context=default regexten=raffo6 username=raffo6 secret=raffo6 fromuser=raffo6 callerid=raffounz2 host=dynamic nat=route canreinvite=no dtmfmode=RFC2833 incominglimit=3 mailbox=1 iax.conf [general] ... ... [raffo5] type=friend context=iaxjingle regexten=raffo5 username=raffo5 secret=raffo5 fromuser=raffo5 callerid=raffounz2 host=dynamic nat=route canreinvite=no dtmfmode=RFC2833 incominglimit=3 mailbox=1 extensions.conf [general] static=yes writeprotect=yes autofallthrough=yes clearglobalvars=no priorityjumping=no [default] exten = s,1,NoOP(Incoming Call from Gtalk) exten = s,n,Answer() exten = s,n,Dial(SIP/11) exten = 11,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) exten = 22,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) exten = 33,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) exten = 44,1,JABBERSend(asterisk,[EMAIL PROTECTED],This is a test Message) exten = 55,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) [iaxjingle] exten = s,1,NoOP(Incoming Call from Gtalk) exten = s,n,Answer() exten = s,n,Dial(IAX2/10) exten = 10,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) exten = 20,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) exten = 30,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) exten = 40,1,JABBERSend(asterisk,[EMAIL PROTECTED],This is a test Message) exten = 50,1,Dial(Jingle/asterisk/[EMAIL PROTECTED]) jingle.conf [general] context=default allowguest=yes [guest] disallow=all allow=ulaw context=guest [google] username=[EMAIL PROTECTED] disallow=all allow=ulaw context=default connection=asterisk jabber.conf [general] debug=yes autoprune=no autoregister=no [asterisk] type=client serverhost=talk.google.com username=[EMAIL PROTECTED] secret=*** port=5222 usetls=yes usesasl=yes buddy=[EMAIL PROTECTED] statusmessage=I am an Asterisk Server timeout=100 Must I reinstall asterisk after removing previous installation due to module issues? Or there's some errors in the conf? Please help me! Thank everyone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can I unistall Asterisk?
Hi everyone, I need to use Asterisk 1.4-beta2 due to its jingle compatibility, but I've read that there are some modules issues upgrading from a previous version. How can I remove a previous version to have a clean install? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: max number of devices in hint
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have one extension that rings in many places. It has just come to my attention that I can only monitor 4 devices within a hint. Ex: exten = 132,hint,SIP/DEVASIP/DEVBSIP/DEVCSIP/DEVD if I add SIP/DEVF, DEVF is not monitored. I'm interested, why do you monitor multiple devices within a hint? If one device is in use (and three are free), how does it show - in use or as free? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voip Buster - CID
- Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 27, 2006 8:44 AM Subject: [asterisk-users] Voip Buster - CID Hi List! Is there any way to set outgoing CID number when making VoIP calls using VoIP Buster? I have search on their forum and I couldn't find anything useful. There is no support mail on their web pages :(( P.S. I use them because they are cheep and sound quality is satisfying There are not many that will allow you to set your own CID even then they normally ask for proof of the numbers you wish to use. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Advice of charge
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... No. I once tried to create a channel variable during hangup. Then, in the hangup extension this variable was added to the user defined CDR field. This generally works, but only if the call leg hangs up, on which the AOC is received. In other cases (e.g. sip to zap calls) when the SIP user hangs up, I had to fetch the last AOC-D value from the bridged channel, which does not work well. There should be a generic method in Asterisk for storing/retrieving AOC, thus I stoped my work. Hi Klaus! Have you provide those information's to developers? Is there any interest to make this work? Approximately, in your opinion, how much work there has to be done? P.S. There are few programmers in company I work for. Can you please send me all relevant code and maybe I can persuade them to look at it. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: max number of devices in hint
I'm interested, why do you monitor multiple devices within a hint? If one device is in use (and three are free), how does it show - in use or as free? I'm glad you asked :-) If we had Shared Line Appearances, I would not have to do this. However, I could be at any of about 6 different phones, and on any of about 4 lines per phone. Therefore, to monitor whether or not I am on the phone would take a 24 BLF buttons or just one, if hinting allowed that many. And to answer your other question, if one device is in use, it shows as being In Use. The others may be free, but I am on one, so therefore, I am busy. I can't believe I missed this in my testing. I just wonder if 1.2.7.1 worked with more devices. I was doing my testing with 1.2.7.1. I'm presently trying to setup a test with another system, without the patches that my current system has, to determine whether it is related to the patches. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - alcatel
Hi,First, Thanks a lot for your help.My responses are in your mail:2006/9/27, Frederico Madeira [EMAIL PROTECTED]: Nicolas, We use a TE110P from digium. We make the same procedures oriented in that website. the only change was in signaling as i've said previously.We try pri_net and pri_cpe on asterisk (changing the network mode too on the alcatel), but we still have a yellow alarm in zttool and a red light on the digium card. My alcatel aready have an E1 ISDN installed from local carrier. I have just a question about this. The cable is a rnis cable (rj45) on the alcatel side or a other shape ? After asterisk is setup, we change cables from carrier to asterisk, and our span stay in green state. Wich pins of cable you use in ISDN cable ?? The cable is the one we found on http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI What is the result of zttools -v ???Nico, can you copy the result of the command ?Best regards,Thomas After span configuration we have problemas making calls, se my post in other forum: http://forums.digium.com/viewtopic.php?t=9868highlight=alcatel+4200 -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: max number of devices in hint
Ok, I just setup a test setup that allowed for five devices (actually in this case five lines) to be monitored. Next question, does anyone know if there is a limit to the number of characters allowed for the hint? That may be what's causing the issue. I just switched to using the MAC addresses for all devices (as suggested in another thread, with -a at the end for line 1, and so on). I wouldn't even know where to begin to look in the code for this. I'm thinking that maybe I should just label the phones 1- whatever and then it would be SIP/1A for device 1, line 1, etc. On a simpler setup, I would just name them the extension, but nothing having to do with the three companies I'm working for is simple. Everything always has to be complicated... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] High CPU usage when Internet goes down
Greetings all, I have a problem with a PBX that I manage. The system has 2 AVM Fritz boards connected to two BRI ISDN services using chan_capi in addition to several SIP trunks going out to Internet based providers for call termination via the Internet. They experience problems when the Internet connection goes down. Obviously the SIP trunks are lost. However the strange thing is that calls are dropped on the capi channels as well during these Internet outages. One of the engineers that I work with felt that the problem was due to Asterisk persistantly trying to re register the SIP services and was using up too much CPU in the process. In fact he was able to workaround the problem temporarily by commenting out the SIP registration in sip.conf, which would confirm his theory. I suppose my question is. Has anyone else seen this sort of behaviour before? Is there any SIP settings that we should be including to try to slow down the SIP registration so that it doesn't use up too many system resources? This message was sent using MyMail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context default incoming ENUM
On 07:10, Wed 27 Sep 06, Ronald Wiplinger wrote: I want to make the context [default] as an alarm, for not having set-up correct. I am looking for a way to get incoming calls via ENUM or via names (e.g. sip:[EMAIL PROTECTED]) into a defined context. How can I do that? If you find out let me know as well. I'm interested in this. I dont think it's possible though, because the call will come in just like any other unauthenticated call. It's not like ENUM is adding sip headers or something. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] señalizacion te110p, signaling te110p
Melcon Moraes wrote: What a confused message, isn't it? As far as I could understand, if you're getting a RJ45 for conection, you won't need any kind of adaptor. For coaxial cable, you'll need a balun. That's all layer 1 talk - physic layer Yes, you need to know a lot more about your pbx to proceed with the connection to your * box(TE110P). hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexiones me sirven o tengo que comprar algun adaptador. vi algo que tenia que usar un balum, es necesario para cualquiera de las dos conexiones?. cual tipo de conexioon me recomiendan mas? necesito saber algo mas sobre la pbx para configurar en la te110p? then he'll have issues with that thing using R2 ;D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN30 and digital phones
Hi, At our other site in the UK we currently have a rather old Nortel BCM (4000 I think), with an ISDN30 feed and 15-16 or so digital extensions (Meridian of some description). The ISDN comes in as HSDSL over a twisted copper pair to a small BT box, then ethernet to the BCM. We'd like to do, at least, inter-office VoIP calls. However I believe making trunks between asterisk and BCM isn't the easiest thing in the world, and my brief exploration of the BCM configuration bears that out. Plus we don't have any licences for VoIP. If I were to recommend replacing the BCM with an asterisk machine, what special hardware/cards would I need? (I so don't understand how US line designations fit in with UK style lines) I'm open to replacing phones if the kit to interface with digital phones costs more than buying SIP phones, every desk already has at least 2 cat5e points. Thanks -- Mike Williams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ISDN30 and digital phones
You can interface between the digital phones and an Asterisk machine using a Citel SIP Handset Gateway from www.citel.com. The sales department on +44 (0)115 940 5444 will be able to give you some pricing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Williams Sent: 27 September 2006 10:56 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ISDN30 and digital phones Hi, At our other site in the UK we currently have a rather old Nortel BCM (4000 I think), with an ISDN30 feed and 15-16 or so digital extensions (Meridian of some description). The ISDN comes in as HSDSL over a twisted copper pair to a small BT box, then ethernet to the BCM. We'd like to do, at least, inter-office VoIP calls. However I believe making trunks between asterisk and BCM isn't the easiest thing in the world, and my brief exploration of the BCM configuration bears that out. Plus we don't have any licences for VoIP. If I were to recommend replacing the BCM with an asterisk machine, what special hardware/cards would I need? (I so don't understand how US line designations fit in with UK style lines) I'm open to replacing phones if the kit to interface with digital phones costs more than buying SIP phones, every desk already has at least 2 cat5e points. Thanks -- Mike Williams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN30 and digital phones
On 27 Sep 2006, at 10:56, Mike Williams wrote: Hi, At our other site in the UK we currently have a rather old Nortel BCM (4000 I think), with an ISDN30 feed and 15-16 or so digital extensions (Meridian of some description). The ISDN comes in as HSDSL over a twisted copper pair to a small BT box, then ethernet to the BCM. We'd like to do, at least, inter-office VoIP calls. However I believe making trunks between asterisk and BCM isn't the easiest thing in the world, and my brief exploration of the BCM configuration bears that out. Plus we don't have any licences for VoIP. If I were to recommend replacing the BCM with an asterisk machine, what special hardware/cards would I need? (I so don't understand how US line designations fit in with UK style lines) I'm open to replacing phones if the kit to interface with digital phones costs more than buying SIP phones, every desk already has at least 2 cat5e points. One way you could do this would be to put an asterisk box in between the Meridian and the ISDN30. You put a dual (or quad) E1 card into the asterisk machine. You then write a simple dialplan that (by default) passes all calls straight through the asterisk machine untouched. Once you have that working, you add rules such that outgoing calls to your other offices are excluded from this process and sent via VOIP. Done right the Nortel will be blissfully unaware of the fact that the asterisk box is even there. (over time you can add features/ functionality to the VOIP area - voicemail , call monitoring etc.) No new phones, no new hardware except the asterisk system. You _should_ even be able to do faxing on the Nortel, provided you prevent the asterisk machine from doing echo canceling on fax calls. The only down-side is that you only get 30 channels to your asterisk, but given that you have only 16 extensions anyway Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
It depends on the actual given environment, but you could also think about using Linksys' PAP2 adapter! mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 64 analog phones
I would think channel banks - T1s - TDM card in asterisk server would work better than a bazillion ata adapaters Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Artner Sent: Wednesday, September 27, 2006 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 64 analog phones It depends on the actual given environment, but you could also think about using Linksys' PAP2 adapter! mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 timing errors (Frame Slips) on Nortel 61C to TE110P
I am setting up an asterisk box , my first with PRI T1 interface to a Nortel 61C. We have quite a bit of experience with the 61C and do most of the programming including maintaining several other PRI interfaces in this switch. The problem we are having is as soon as we turn up the PRI, on the 61C side we get PRI0264 protocol errors. Then the circuit lays down properly. At this point we start accumulating SLIPR in the 61C which resets the circuit in about 5 minutes. Below are my configurations. Nortel 61C CEQU MPED 8D TERM REMO TERD REMD TERQ REMQ SUPL 004 012 024 V048 N156 SUPC SUPF XCT 000 016 TDS * 000 * 016 CONF * 001 * 017 MFSD * 000 * 016 DLOP NUM DCH FRM TMDI LCMT YALM TRSH TRK 009 12 ESF NO B8S FDL 00 PRI 008 24 ESF NO B8S FDL 00 010 24 ESF NO B8S FDL 00 011 24 ESF NO B8S FDL 00 018 24 ESF NO B8S FDL 00 019 24 ESF NO B8S FDL 00 020 24 ESF NO B8S FDL 00 021 24 ESF NO B8S FDL 00 030 24 ESF NO B8S FDL 00 031 24 ESF NO B8S FDL 00 Blah..blah ADAN DCH 50 CTYP MSDL DNUM 10 PORT 1 DES ippbx USR PRI DCHL 8 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K 7 TYPE RDB CUST 00 ROUT 97 DES IPPBX TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 7997 TCPP NO PII NO TARG CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO zapata.conf [trunkgroups] [channels] language=en context=default switchtype=5ess signalling=pri_net usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 musiconhold=default group = 1 channel = 1-23 zaptel.conf span = 1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us I think the configuration is right. I have tried changing the timimg source in zaptel.conf from 1 to 0 to no avail. Also I can not set up the Nortel PRI to look internal for clock. Nortel sets up by default CLOK = EXT. I have tried different cross over cables. I can point the asterisk into a T-Berd 950N set up to turn up a PRI and it will work and run clean on the Asterisk server. I can loop back the Nortel PRI and it will est wrong mode and accumulate no SLIPR. I am struggling to get this to work. The circuit does establish and pass calls but resets frequently due to slips. Dell 2850/TE110P/Asterisk business edition ABE-B.1-1/Redhat EL4/Nortel 61C/Succession R3/MSDL Dchannel/NT5D12. Any help would be appreciated. Ronnie Jones Engineer - ICT Clay Electric Cooperative, Inc 352-473-8000 ext. 8272 352-473-1929(F) 352-745-0910(C) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Right way to prevent analog channel from answering the phone?
I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good Book on Asterisk
Hi everybody! I have some Linux experience but I'm completely new to asterisk. I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk (1.2.12) preinstalled and some basic configuration (Wiht a few extensions). Now I want to implement something more, fox example voicemail (storing voicemail data in an extern mysql DB) and so on. And since I don't want to waste your time with stupid questions ... can someone of you recommend a really good book on Asterisk? (To buy or for download) ... or another online source of information which would be helpful for someone like me? I searched Amazon with Asterisk and got 21 hits.. Thanks Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book on Asterisk
Norbert Zawodsky wrote: Hi everybody! I have some Linux experience but I'm completely new to asterisk. I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk (1.2.12) preinstalled and some basic configuration (Wiht a few extensions). Now I want to implement something more, fox example voicemail (storing voicemail data in an extern mysql DB) and so on. And since I don't want to waste your time with stupid questions ... can someone of you recommend a really good book on Asterisk? (To buy or for download) ... or another online source of information which would be helpful for someone like me? I searched Amazon with Asterisk and got 21 hits.. Thanks Norbert Hi, Norbert ... The O'Reily Book for Asterisk: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Enjoy! -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 timing errors (Frame Slips) on Nortel 61C to TE110P
Ronnie Jones wrote: I am setting up an asterisk box , my first with PRI T1 interface to a Nortel 61C. We have quite a bit of experience with the 61C and do most of the programming including maintaining several other PRI interfaces in this switch. The problem we are having is as soon as we turn up the PRI, on the 61C side we get PRI0264 protocol errors. Then the circuit lays down properly. At this point we start accumulating SLIPR in the 61C which resets the circuit in about 5 minutes. Below are my configurations. Nortel 61C CEQU MPED 8D TERM REMO TERD REMD TERQ REMQ SUPL 004 012 024 V048 N156 SUPC SUPF XCT 000 016 TDS * 000 * 016 CONF * 001 * 017 MFSD * 000 * 016 DLOP NUM DCH FRM TMDI LCMT YALM TRSH TRK 009 12 ESF NO B8S FDL 00 PRI 008 24 ESF NO B8S FDL 00 010 24 ESF NO B8S FDL 00 011 24 ESF NO B8S FDL 00 018 24 ESF NO B8S FDL 00 019 24 ESF NO B8S FDL 00 020 24 ESF NO B8S FDL 00 021 24 ESF NO B8S FDL 00 030 24 ESF NO B8S FDL 00 031 24 ESF NO B8S FDL 00 Blah..blah ADAN DCH 50 CTYP MSDL DNUM 10 PORT 1 DES ippbx USR PRI DCHL 8 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K7 TYPE RDB CUST 00 ROUT 97 DES IPPBX TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 7997 TCPP NO PII NO TARG CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO zapata.conf [trunkgroups] [channels] language=en context=default switchtype=5ess signalling=pri_net usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 musiconhold=default group = 1 channel = 1-23 zaptel.conf span = 1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us I think the configuration is right. I have tried changing the timimg source in zaptel.conf from 1 to 0 to no avail. Also I can not set up the Nortel PRI to look internal for clock. Nortel sets up by default CLOK = EXT. I have tried different cross over cables. I can point the asterisk into a T-Berd 950N set up to turn up a PRI and it will work and run clean on the Asterisk server. I can loop back the Nortel PRI and it will ‘est wrong mode’ and accumulate no SLIPR. I am struggling to get this to work. The circuit does establish and pass calls but resets frequently due to slips. Dell 2850/TE110P/Asterisk business edition ABE-B.1-1/Redhat EL4/Nortel 61C/Succession R3/MSDL Dchannel/NT5D12. Any help would be appreciated. I have no experience on the Nortel side, but will comment on the timing thingie. The asterisk T1 card (port going to the Nortel) will always generate T1 timing on the transmit side of the T1. There is no way to turn it off (by T1 Spec's). So, letting the Nortel use CLOK = EXT is perfect. The sync parameter in /etc/zaptel.conf for that same T1 port should probably be set to zero, but that statement is somewhat dependent on what the other ports on the Asterisk T1 card are used for. If there are no other Asterisk T1 card ports in use, then I'd suggest setting the sync parameter to 1. If at least one other Asterisk T1 port is in use and goes to a central office, then turn that port's sync to 1 and the Nortel port sync to 0. (Keep in mind the digium T1 cards only have one clock on board, and syncing that clock to a T1 coming from a central office is the right thing to do. Once that clock is in sync, then the Nortel will sync to asterisk.) I'm a little confused with your last paragraph when you say the circuit does establish and pass calls but resets frequently due to slips. Are those calls to/from asterisk talking to the Nortel? Or, are you routing incoming pstn calls from the central office through asterisk to the Nortel? Also, have you tried any of the pri show ... commands in asterisk, or any of the pri debug items?
Re: [asterisk-users] PRI Outbound CallerID Question
On Tue, Sep 26, 2006 at 08:11:09PM -0400, Kristian Kielhofner wrote: But gratuituously making easy something that very few people have a legitimate need to do, which undermines something that -- even if you do only make the resaonable assumption that you know which phone, and not which person, is calling -- is useful and productive... is probably a Bad Idea. Full disclosure notwithstanding. jra, Sprint made the mistake. That is ridiculous... Certainly. Caller ID has not been secure for a long time. If you think that it should be made secure now, you are out of touch with reality because that is NOT going to happen. It has been made easy. It is ubiquitous. Get over it :)! Not at all. The number of ingress points to native SS7 is tiny. The number of ingress points to ISDN, while far larger, is still on the order of maybe 6-digits of sites (the end-offices), and wouldn't be all that difficult to secure at all. The only options now are to not trust caller id, ask more questions (i.e. get better identity systems and processes in place), and, as I said, enforce laws that we already have. Certainly. I think you missed my point that setting caller id in a nefarious way is almost always used as a tool in an action that is already defined as a crime. The things you are talking about doing are already illegal - whether or not you are spoofing caller id. Granted, caller id does make it easier, but if we didn't have the ability to set caller id the crooks would still be scamming, harassing, etc just like they are now. They would just be using other tools to do it or make it easier for them. Well, not all of them, actually. Telemarketers, who are constrained to send proper caller id, do not, I believe, inlcude credit bureaux, and PI pretexting is not per-se illegal either, at the moment. But let's remember one fundamental point, raised in the rollout of CNID in the first place: my phone belongs to *me*; I pay for it for *my* convenience, not that of others. The LEC's *make money* off of CNID service provision; they have, it seems to me, an obligation to make sure, collectively, that it does what they say it does. Cheers, -- jra Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
On Tue, Sep 26, 2006 at 09:30:04PM -0400, Kristian Kielhofner wrote: Steve Totaro wrote: I set caller ID to a unique identifier before sending to a transfer partner or overflow call center. This makes it much easier to match CDRs and get stats on the outcome of calls once they leave our center. It is a very valuable and legitimate use. Am I committing a crime? nah. This is exactly what I am talking about. That is a very useful application of caller id manipulation. Why should you lose that useful feature because a few misguided people sometimes use it for nefarious purposes? Strawman, Kristian; I already covered that, and confirmed that I believe it to be an acceptable use, also, covered by agreement. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I doubt it...
On Wed, Sep 27, 2006 at 10:21:31AM +0530, Benjamin Jacob wrote: Jay R. Ashworth wrote: On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote: hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, / hello to all, I have a doubt, ye I have solved some but others arrive, good *Oh*. *That's* where all these non-native English speakers are coming up with doubt. Someone's translator doesn't have an idiom for I have an inquiry. Eeediots. Easy.. Easy Jay!! easy duz it! You can't *expect* all to be native English speakers over here, or anywhere, for that matter. And am sure, he won't have a translator or a dictionary next to him, whenever he posts on this list. Re-read my comment, Ben. I'm calling the *translator writers* idiots. Not the OP. As long as ppl are harmless, are talking asterisk, are making sense, arn't cussing you out, its A-OK!! You, you... *astronaut*, you! :-) n besides, there isn't a Correct-English-talkers-only clause over here, I guess. Imagine a spanish-only (was that guy spanish??or mexican? baah.. all sound same) world and you would be sending doubts across as well, with a Spanish Jay calling you an eeedioto!! Oddly, someone just dissed Allison Smith's spanish this week, so... live n let live. or over here, lets make it as Asterisk and let Asterisk. now that i've flung my two cents, lets start a flame ;-) Not at all. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I doubt it...
On Wed, Sep 27, 2006 at 12:20:02AM -0500, Lacy Moore - Aspendora wrote: I didn't see it as making fun of anyone. I, for one, was curious about it. I suspected it was some type of translation issue, whether it was a word in another language that doesn't translate or what. I know there are many concepts in English and in other languages that just doesn't translate correctly. I can't imagine how any software could translate all the different English dialects, so I'm sure translators have problems from other languages. The issue is idiomatic usage. I've always assumed they did it in a table driven fashion, but I never delved into it. I have seen quite a few speakers of other languages use doubt in the meaning of question, inquiry though, sometimes in contexts where it would tend (IMHO) to squick potential question-answerers. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right way to prevent analog channel from answering the phone?
Is there a more elegant way to tell it to answer/not answer on command? Put your Zap line in a context that do just this : s,1,Hangup() hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right way to prevent analog channel from answering the phone?
Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? I don't have an A1200P, but most zap channel interfaces are built to not answer an incoming call unless you specifically configure asterisk to do it. There are only two basic conditions under which an incoming call will be answered: 1. by including the answer statement, like: exten = 3556,1,Answer exten = 3556,2,Wait,1 exten = 3556,3,Authenticate(3017) exten = 3556,4,Meetme(3556|pM) 2. a SIP phone (or other phone) user picks up the handset. So, in zapata.conf you have definitions for each of the A1200P ports, and one of the items in those definitions is context=something. If that context statement points to some non-existent context name (like context=xyz), there is nothing that would answer the incoming call. If the context=something points to a real context (in extensions.conf), then review that context to ensure there is nothing there to answer the incoming call. (Note: some asterisk applications will automatically answer incoming calls.) You could also define that context and include statements like: [no-answer] exten = _X.,1,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing DialPlan
Hi All Would someone be kind enough to provide/point me to a resource when I can see an example dialplan for making outgoing calls. All our calls with go out via an ISDN30 gateway so ideally the diaplan needs to be able to deal with the following errors: no free channels user busy user didnt answer number unallocated any others people can think of :-) Many Thanks Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] I doubt it...
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: 27 September 2006 15:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I doubt it... The issue is idiomatic usage. I've always assumed they did it in a table driven fashion, but I never delved into it. I have seen quite a few speakers of other languages use doubt in the meaning of question, inquiry though, sometimes in contexts where it would tend (IMHO) to squick potential question-answerers. I have a doubt about the word squick. What means this? :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX phones?
Just wondering if there are any IAX phones worthy of the name phone out there -- looking for hard phones, but I suppose a Linux-based softphone wouldn't, you know, hurt. ;-) Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM04B Installation Problem
Hi, I have got a Digium TDM04B card (4 FXO modules installed) and i'm having problems getting it working. ztcfg reports the following: asterisk:~# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) When I do: modprobe zaptel I get the following in the logs: Sep 27 15:24:50 asterisk kernel: Zapata Telephony Interface Registered on major 196 Sep 27 15:24:50 asterisk kernel: Zaptel Version: 1.2.9.1 Echo Canceller: KB1 and when I do modprobe wctdm I get the following: Sep 27 15:25:39 asterisk kernel: ACPI: PCI interrupt :01:01.0[A] - GSI 22 (level, low) - IRQ 217 Sep 27 15:25:39 asterisk kernel: Freshmaker version: 73 Sep 27 15:25:39 asterisk kernel: Freshmaker passed register test Sep 27 15:25:40 asterisk kernel: Module 0: Not installed Sep 27 15:25:40 asterisk kernel: Module 1: Not installed Sep 27 15:25:40 asterisk kernel: Module 2: Not installed Sep 27 15:25:40 asterisk kernel: Module 3: Not installed Sep 27 15:25:40 asterisk kernel: wctdm: probe of :01:01.0 failed with error -5 lspci is showing the following: :01:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0003 Flags: bus master, medium devsel, latency 64, IRQ 217 I/O ports at de00 [size=256] Memory at fe9fe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Can anyone help? Thanks Ian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right way to prevent analog channel from answering the phone?
What is wrong with using the WaitForRing app? Rich Adamson wrote: Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? I don't have an A1200P, but most zap channel interfaces are built to not answer an incoming call unless you specifically configure asterisk to do it. There are only two basic conditions under which an incoming call will be answered: 1. by including the answer statement, like: exten = 3556,1,Answer exten = 3556,2,Wait,1 exten = 3556,3,Authenticate(3017) exten = 3556,4,Meetme(3556|pM) 2. a SIP phone (or other phone) user picks up the handset. So, in zapata.conf you have definitions for each of the A1200P ports, and one of the items in those definitions is context=something. If that context statement points to some non-existent context name (like context=xyz), there is nothing that would answer the incoming call. If the context=something points to a real context (in extensions.conf), then review that context to ensure there is nothing there to answer the incoming call. (Note: some asterisk applications will automatically answer incoming calls.) You could also define that context and include statements like: [no-answer] exten = _X.,1,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Jay R. Ashworth wrote: On Tue, Sep 26, 2006 at 09:30:04PM -0400, Kristian Kielhofner wrote: Steve Totaro wrote: I set caller ID to a unique identifier before sending to a transfer partner or overflow call center. This makes it much easier to match CDRs and get stats on the outcome of calls once they leave our center. It is a very valuable and legitimate use. Am I committing a crime? nah. This is exactly what I am talking about. That is a very useful application of caller id manipulation. Why should you lose that useful feature because a few misguided people sometimes use it for nefarious purposes? Strawman, Kristian; I already covered that, and confirmed that I believe it to be an acceptable use, also, covered by agreement. Cheers, -- jra jra, If someone doesn't respect laws from Congress and State legislators, they certainly aren't going to be stopped by a $0.02 civil agreement drafted by some telcos legal team... However, if such an agreement were required to ensure the reliability and quality of caller id services, I would have no problem signing one. ;) -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX phones?
Ken - the IAX compatible phones I have seen, for the most part, are OEM looking, and overall pretty cheaply made. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Wednesday, September 27, 2006 11:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IAX phones? Just wondering if there are any IAX phones worthy of the name phone out there -- looking for hard phones, but I suppose a Linux-based softphone wouldn't, you know, hurt. ;-) Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing call problem
Zap channels consider the call answered when dialing is complete, at least with the analog interface. There is no answer supervision provided to the PSTN with a POTS line Don't know if this extends to a PRI or not. John Novack Alexandru Voinescu wrote: Hi. I'm having a bit of trouble with outgoing calls on zap channels. When i try to make an outgoing call asterisk doesn't detect if the other party answers. When i run 'show channels verbose' in CLI asterisk tells me that the respective channles are in ringing state like this: Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo Zap/19-1 agentie s 1 Dialing AppDial (Outgoing Line) 00726710704 (None) Zap/15-1 int_omg 00726710704 5 Ring Dial Zap/g5/0726710704||T 00:00:14 (None) although i can speak to the called party. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
On Wed, Sep 27, 2006 at 09:12:31AM -0400, Bill Gibbs wrote: I would think channel banks - T1s - TDM card in asterisk server would work better than a bazillion ata adapaters Assuming that you don't need to have a T-1 card in their for your *trunks*. Since I'm told that you can only have, say, one Digium card per chassis, this can be an issue. This question seems to come up a lot; am I the only person who knows about Media Gateways? :-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I doubt it...
On Wed, Sep 27, 2006 at 07:45:07AM -0700, Steve Langstaff wrote: The issue is idiomatic usage. I've always assumed they did it in a table driven fashion, but I never delved into it. I have seen quite a few speakers of other languages use doubt in the meaning of question, inquiry though, sometimes in contexts where it would tend (IMHO) to squick potential question-answerers. I have a doubt about the word squick. What means this? :) Smartass. :-) UTFW: http://en.wikipedia.org/wiki/Squick Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SER with multiple asterisk deployment
Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large or the asterisk_integration page at openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX phones?
Just wondering if there are any IAX phones worthy of the name phone out there -- looking for hard phones, but I suppose a Linux-based softphone wouldn't, you know, hurt. ;-) Idefisk looks pretty nice and there is a Linux version : http://www.asteriskguru.com/idefisk/ There is also iaxcomm : http://iaxclient.sourceforge.net/iaxcomm/index.html Also, check on iaxclient page : http://iaxclient.sourceforge.net/ hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
Jay R. Ashworth wrote: Assuming that you don't need to have a T-1 card in their for your *trunks*. Since I'm told that you can only have, say, one Digium card per chassis, this can be an issue. You were told wrong. I have had up to FOUR Digium cards in a chassis. 3xTDM400P and 1xTE110P. I have also had 2xTE110Ps in a box. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASTTAPI
Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zapata.conf
Hello, I have a problem with my X100P card I have connected it to my asterisk and it works.. but I hear an echo.. I´ve tried echocancelation... echotraining.. and nothing happens... I´ve changed the values from the rx and txgain.. from -40 to 10 and it doesn´t changes anything.. Don´t know what else to do.. [channels] language=en context=default signalling=fxs_ks channel = 1 usecallerid=no hidecallerid=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=-4.0 txgain=-4.0 group=1 pickupgroup=1 immediate=yes this is my zapata.conf... Any ideas? what is missing? Oh.. I´ve tryed to see the levels with the ztmonitor to see any change.. and nothing changes.. Thanks for reading, Danko ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 64 analog phones
Since I'm told that you can only have, say, one Digium card per chassis, this can be an issue. ??? lspci | grep Jens 01:01.0 Network controller: Individual Computers - Jens Schoenfeld Intel 537 01:04.0 Network controller: Individual Computers - Jens Schoenfeld Intel 537 asterisk -rx zap show channels -- Remote UNIX connection pseudofrom-pstn en 1 8516 from-pstn en 2from-pstn en 3 8282 from-pstn en 4 8394 from-pstn en 5from-pstn en 6 4286 from-pstn en 7from-pstn en 8 4773 from-pstn en 9from-pstn en 10 8104 from-pstn en 11 8777 from-pstn en 12 8443 from-pstn en 13 8901 from-pstn en 14 8142 from-pstn en 15 3808 from-pstn en 16 4773 from-pstn en 17 2996 from-pstn en 18from-pstn en 19from-pstn en 20from-pstn en 21 8304 from-pstn en 22from-pstn en 23 7010303from-pstn en 25from-pstn en 26 2996 from-pstn en 27from-pstn en 28 8321 from-pstn en 29 8247 from-pstn en 30 8194 from-pstn en 31 8833 from-pstn en 32 3154 from-pstn en 33 3807 from-pstn en 34 5959 from-pstn en 35 3619 from-pstn en 36from-pstn en 37from-pstn en 38from-pstn en 39from-pstn en 40from-pstn en 41 4773 from-pstn en 42 3093 from-pstn en 43from-pstn en 44 8341 from-pstn en 45from-pstn en 46 8295 from-pstn en 47 3094 from-pstn en top | grep load top - 10:27:41 up 76 days, 21:23, 1 user, load average: 1.17, 1.43, 1.33 no problems here. 4 way P-3 Netfinity. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Status via Dialplan
Using queues here (1 of them), and would like to know if anyone's written anything like a script that might tell someone by festival or the like of the status of a queue, like # of calls waiting, and hold times... Any other way of finding that out without spending a ton of money on third party packages ? R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 64 analog phones
We got a few 16 ports Media gateway for quite a reasonable price. Email me for more info. 4 of them and it will end up cost you less than getting channel banks and t1 card. Sam -Original Message- From: Jay R. Ashworth [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 27, 2006 11:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 64 analog phones On Wed, Sep 27, 2006 at 09:12:31AM -0400, Bill Gibbs wrote: I would think channel banks - T1s - TDM card in asterisk server would work better than a bazillion ata adapaters Assuming that you don't need to have a T-1 card in their for your *trunks*. Since I'm told that you can only have, say, one Digium card per chassis, this can be an issue. This question seems to come up a lot; am I the only person who knows about Media Gateways? :-) Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
On Wed, Sep 27, 2006 at 11:15:48AM -0500, Eric ManxPower Wieling wrote: Jay R. Ashworth wrote: Assuming that you don't need to have a T-1 card in their for your *trunks*. Since I'm told that you can only have, say, one Digium card per chassis, this can be an issue. You were told wrong. I have had up to FOUR Digium cards in a chassis. 3xTDM400P and 1xTE110P. I have also had 2xTE110Ps in a box. It's been mentioned on this list, repeatedly, that having more than one Digium T-span card on a bus was asking for trouble -- which, given that they apparently interrupt once per scheduler tick, doesn't surprise me much. Were those people -- who, unlike me, had done it and had problems -- wrong? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [asterisk-users] Re: asterisk to cell phone network
Well why pay more when you can get it at much cheaper price. A single port gsm gateway is around £69 GBP and if you want to know more info please email me . Sam -Original Message- From: Andrea Spadaccini [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 27, 2006 12:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: asterisk to cell phone network Ciao Michiel, http://www.junghanns.net/en/GSM-PCI_produkt.html If they are as stable as the quad/octo BRI cards they have it's a real winner. Where can I see the prices of this cards? My supplier has them listed as: UnoGSM: 900 euro DuoGSM: 1200 euro QuadGSM: 1600 euro Well, how does Asterisk interact with those devices? Is there a chan_gsm_pci? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER with multiple asterisk deployment
Adi, It is possible to do what you are looking for. It is actually easy. There is a problem that I have found with ser/openser.. Documentation is difficult to read and some things are just not there, so you get people that spend many hours trying to get these functions to work. In these days time is money, so the people that know how to do what you are seeking.. charge large amounts of money for a simple 50 line config file. I will tell you that everything you are looking for is documented in examples. You will have to piece them together and make them work in harmony like the rest of us have. I suggest you look at voip user and piece the config together from examples there. It may also help you to read the source code of the modules that handle routing in ser. There are a few tricks that are hidden in the code. I am sorry for my vagueness. I am not able to share the config information due to an IP agreement with my company.(They think it is a trade secret) I wish you the best. Cheers, Zac Amsler, Network Operations Sur-Tel Communications, Inc. NetIQ Systems, LLC * US48, Canada, A-Z Wholesale Termination. * US48 Origination, Toll Free DIDs. * Toll Free Termination (FREE). Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large http://www.voip-info.org/wiki-Asterisk+at+large or the asterisk_integration http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration page at openser.org http://openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any suggestions about VoIP provider?
Hi, I have a client operating a call center in Jordan, he has a new 5 years project to make and receive VoIP calls to/from the US. The project requires a T1 US termination (24 lines) with at least 99.9% uptime and perfect voice quality and multiple area codes. Can anyone suggest a VoIP provider in the United States (preferably out of experience) who can provide the above termination needs? Quality and uptime is the prime factor in the decision and not price. You can email me off list if you want, Thank you and best regards, Antoine Megalla SAND (S.A.E) __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right way to prevent analog channel from answering the phone?
Well, that just makes too much sense.. starting to feel a tad embarrased here ;) Ok, I will simply remove the Dial(IAX2/4005) and have it not do anything, that will error on the console, but that's ok and let the parallel land line have the call (AKA: The wife) Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Wed, 27 Sep 2006, Rich Adamson wrote: Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? I don't have an A1200P, but most zap channel interfaces are built to not answer an incoming call unless you specifically configure asterisk to do it. There are only two basic conditions under which an incoming call will be answered: 1. by including the answer statement, like: exten = 3556,1,Answer exten = 3556,2,Wait,1 exten = 3556,3,Authenticate(3017) exten = 3556,4,Meetme(3556|pM) 2. a SIP phone (or other phone) user picks up the handset. So, in zapata.conf you have definitions for each of the A1200P ports, and one of the items in those definitions is context=something. If that context statement points to some non-existent context name (like context=xyz), there is nothing that would answer the incoming call. If the context=something points to a real context (in extensions.conf), then review that context to ensure there is nothing there to answer the incoming call. (Note: some asterisk applications will automatically answer incoming calls.) You could also define that context and include statements like: [no-answer] exten = _X.,1,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right way to prevent analog channel from answering the phone?
Erm.. nothing that I know of, other than I do not yet know what that means? :) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Wed, 27 Sep 2006, Eric ManxPower Wieling wrote: What is wrong with using the WaitForRing app? Rich Adamson wrote: Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? I don't have an A1200P, but most zap channel interfaces are built to not answer an incoming call unless you specifically configure asterisk to do it. There are only two basic conditions under which an incoming call will be answered: 1. by including the answer statement, like: exten = 3556,1,Answer exten = 3556,2,Wait,1 exten = 3556,3,Authenticate(3017) exten = 3556,4,Meetme(3556|pM) 2. a SIP phone (or other phone) user picks up the handset. So, in zapata.conf you have definitions for each of the A1200P ports, and one of the items in those definitions is context=something. If that context statement points to some non-existent context name (like context=xyz), there is nothing that would answer the incoming call. If the context=something points to a real context (in extensions.conf), then review that context to ensure there is nothing there to answer the incoming call. (Note: some asterisk applications will automatically answer incoming calls.) You could also define that context and include statements like: [no-answer] exten = _X.,1,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 64 analog phones
Were those people -- who, unlike me, had done it and had problems -- wrong? There are more variables than the Digium card itself. Things like bus design, chipset etc all come into play. I've noticed that there is a concerted effort with Asterisk implmentors to often roll out Asterisk in a white box clone with a $129 integrated motherboard in an effort to drive down the cost which (IMO) is foolish. Why would you (not you, but people) spec a DL380G4 for a database server then turn around and use an ECS brand motherboard for a telephony platform - a platform which by definition requires sub-millisecond response time? Tier 1 boxes are *designed* for this type of application and are engineered to conform to spec + safety margin, wheras Taiwan clone boards are usually designed to roughly confirm to spec, and that's all (notable exception: I have used ASUS motherboards for Asterisk installs, and they all work flawlessly) I've said it many times before on the list: It's trivial to make a crappy Asterisk install. Anyone can do it. It's really, really, hard to make a *good* Asterisk install. You need cross-discipline experience, a lot of which is hard to come by in the closed, secretive telephony world. Tip o' the hat to SHSU. I wouldn't touch *that* install with a space tether. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault on Asteriskstartup:res_config_mysql.so problem?
Did you do a make make install for add-ons BEFORE doing
so for asterisk?
If so try asterisk first and when all is installed install
add-ons.
--
I tried a make clean make make install for asterisk and then for
asterisk-addons but am still getting the segmentation fault on asterisk
startup. rm res_config_mysql.so allows Asterisk to start.
Still trying...
mkdir /usr/lib/asterisk.backup.20060928
mv /usr/lib/asterisk/* /usr/lib/asterisk.backup.20060928
mkdir /usr/include/asterisk.backup.20060928
mv /usr/include/asterisk/* /usr/include/asterisk.backup.20060928/
cd /usr/src/asterisk-1.2.12.1
make clean make make install
cd /usr/src/asterisk-addons-1.2.4
perl -p -i.bak -e
's/CFLAGS.*D_GNU_SOURCE/CFLAGS+=-D_GNU_SOURCE\nCFLAGS+=-DMYSQL_LOGUNIQUEID/'
Makefile
make clean make make install
Install logs look fine.
STARTING ASTERISK
/usr/sbin/safe_asterisk: line 40: 6631 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 40: 6690 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
rm res_config_mysql.so allows Asterisk to start.
Any advice appreciated.
Cameron
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Re: asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)
2006/9/26, Steve Underwood [EMAIL PROTECTED]:snipT.38 termination is now fairly solid. T.38 gateway is also basically working, snipHi,For may understanding, what is the difference between T.38 termination and T.38 gateway ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right way to prevent analog channel from answering the phone?
Nick Ellson wrote: Erm.. nothing that I know of, other than I do not yet know what that means? :) pbx-1*CLI show application waitforring pbx-1*CLI -= Info about application 'WaitForRing' =- [Synopsis] Wait for Ring Application [Description] WaitForRing(timeout) Returns 0 after waiting at least timeout seconds. and only after the next ring has completed. Returns 0 on success or -1 on hangup pbx-1*CLI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys/Sipura 3K, Calls Timing Out
Hi All, I have a Linksys SPA-3000 [Hardware version 3.0.0(1178), Software version 3.1.10(GWd)], with both the FXO and FXS interfaces registering with asterisk via SIP seperatley. I also have a Cisco 7940 and 7960 using the sccp2 (chan_sccp) driver, and a couple of IAX softphones Both inbound and outbound calls to/from the FXO interface time out after around 17-20 minutes. With SIP debug turned on, it looks like the call was just ended normally. This problem doesnt occur with IAX-SCCP calls, just those via the SPA-3K FXO interface. Ive checked all the timeouts in the Linksys configuration, and set them all way higher than 17-20 minutes. I've tried with both G729, and uLaw CODECs, same thing. I've tried turning off silence detection, and the hangup detection is set correctly for the UK. Has anyone else had this happen, or any idea what the problem might be ? As you might imagine, its rather frustraiting to be half way thru a call, and it just hang up on you! All the Best Iain ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] txfax question
I am playing with txfax. I have gotten a fax to send which is great. However now I am creating a multipage fax, I can view all the pages with viewfax (mgetty-viewfax package) but when I fax it with txfax I only get 1 page Any ideas there? Jerry I basically do: gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 -dNOPAUSE -dBATCH -sOutputFile=/tmp/file.g3 file1.pdf file2.pdf when I viewfax /tmp/file.g3 I see all the pages. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemailmail hanging after entering password
I had a problem with the voicemail system hanging after certain users would enter their password. I found that lock files get left behind. In order to fix this, in my startup script I put this line: rm -f /var/spool/asterisk/voicemail/*/*/*/.lock* Works nicely. Hope it helps someone else. W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco ATA escaping # into %23?
I'm setting up a * system for a friend. Instead of dial 9, he wants his internal extentions to be prefaced with #. We have it working on his kid's mac with softphone, his desk with a gxp2000, but he wants to replace his house phones with two ata-186's . We have a problem though. The ATA's (3.2.1atasip) seem to be escaping the hash into %26. So dialing #50 from the ata causes the following request: To: sip:[EMAIL PROTECTED];user=phone;tag=as6aa26793 and asterisk cannot find it, since it's listed in extentions.conf as #50. I've tried googling for it and going over Cisco's docs, voip-info.org, etc and haven't found any references to this. Is the fault in the ATA or asterisk? It's an SVN version that will be updated to 1.4b2 shortly, so if it was fixed in the interum I apologize for wasting time. Thanks -Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
I have no experience on the Nortel side, but will comment on the timing thingie. The asterisk T1 card (port going to the Nortel) will always generate T1 timing on the transmit side of the T1. There is no way to turn it off (by T1 Spec's). So, letting the Nortel use CLOK = EXT is perfect. The sync parameter in /etc/zaptel.conf for that same T1 port should probably be set to zero, but that statement is somewhat dependent on what the other ports on the Asterisk T1 card are used for. If there are no other Asterisk T1 card ports in use, then I'd suggest setting the sync parameter to 1. If at least one other Asterisk T1 port is in use and goes to a central office, then turn that port's sync to 1 and the Nortel port sync to 0. (Keep in mind the digium T1 cards only have one clock on board, and syncing that clock to a T1 coming from a central office is the right thing to do. Once that clock is in sync, then the Nortel will sync to asterisk.) I'm a little confused with your last paragraph when you say the circuit does establish and pass calls but resets frequently due to slips. Are those calls to/from asterisk talking to the Nortel? Yes that is correct. The Nortel switch connects to the PSTN but not the Asterisk. It connects to the Nortel. While the circuit is up I can call extensions on the Nortel from the Asterisk and visa versa. Or, are you routing incoming pstn calls from the central office through asterisk to the Nortel? No Also, have you tried any of the pri show ... commands in asterisk, or any of the pri debug items? Yes. When the circuit is up I can pri show span 1 and it show partitioned up and active. Ronnie Jones Engineer - ICT Clay Electric Cooperative, Inc 352-473-8000 ext. 8272 352-473-1929(F) 352-745-0910(C) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER with multiple asterisk deployment
Hi Zac, Thank you so much for your sincere answer. What you brought up is exactly what I encountered when I tried to find a solution for this, the documentation is inconsistent and ambiguous, and everywhere I look I end up with outdated examples that make little or no sense in the good case, or just don't compile due to being so old in the bad case. This is very frustrating but just by reading what you wrotewas very uplifting for me. Thanks again, Adi. On 9/27/06, Zac Amsler [EMAIL PROTECTED] wrote: Adi,It is possible to do what you are looking for. It is actually easy.There is a problem that I have found with ser/openser.. Documentation is difficult to read and some things are just not there, so you get peoplethat spend many hours trying to get these functions to work. In thesedays time is money, so the people that know how to do what you are seeking.. charge large amounts of money for a simple 50 line config file.I will tell you that everything you are looking for is documented inexamples. You will have to piece them together and make them work in harmony like the rest of us have.I suggest you look at voip user and piece the config together fromexamples there. It may also help you to read the source code of themodules that handle routing in ser. There are a few tricks that are hidden in the code.I am sorry for my vagueness. I am not able to share the configinformation due to an IP agreement with my company.(They think it is atrade secret)I wish you the best. Cheers,Zac Amsler, Network OperationsSur-Tel Communications, Inc. NetIQ Systems, LLC* US48, Canada, A-Z Wholesale Termination.* US48 Origination, Toll Free DIDs.* Toll Free Termination (FREE). Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large http://www.voip-info.org/wiki-Asterisk+at+large or the asterisk_integration http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration page at openser.org http://openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk skills in the philippines
hi thereAdvanced Science and Technology Institute uses Asterisk. On 9/21/06, tubongpeyups [EMAIL PROTECTED] wrote:hi all,my apologies for posting it here in a technical mailing list. i need some info on companies that support asterisk deployment in the Philippines. Please send me a note offline. thanks Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA941 - Asterisk - Voip provider - PSTN - ShoreTel garble
On 9/22/06, Rich Adamson [EMAIL PROTECTED] wrote: So, it seems there is some type of weird interaction between my system and the ShoreTel system if I use the SPA941 IP phone. Does anyone have suggestions as to how I can start debugging this? Check the RTP Packet Size (under the Sip tab). Set it to .020 (20 milliseconds) and place another test call. For whatever reason, the Linksys/Sipura products default to 30 milliseconds and has impacted the quality of audio on some systems. Setting the RTP packet size to 20ms seems to have fixed it. Thanks for the suggestion. Cliff -- === Cliff Brake http://bec-systems.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RPID
Has anyone successfully gotten rpid working between two phones through asterisk? Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spurious hangups on zaptel interface
We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of flash (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1 Barry D. Hassler President HCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED] +1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED] Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER with multiple asterisk deployment
How do you plan on choosing which Asterisk server to send the SIP requests? Truly random? Based on some sort of LCR methodology? Have you tried using the LCR module for SER to send the requests to asterisk? Not sure it would work, but it might be worth looking at. N. On Wed, 27 Sep 2006 21:34:33 +0200, Adi Simon wrote Hi Zac, Thank you so much for your sincere answer. What you brought up is exactly what I encountered when I tried to find a solution for this, the documentation is inconsistent and ambiguous, and everywhere I look I end up with outdated examples that make little or no sense in the good case, or just don't compile due to being so old in the bad case. This is very frustrating but just by reading what you wrote was very uplifting for me. Thanks again, Adi. On 9/27/06, Zac Amsler [EMAIL PROTECTED] wrote: Adi, It is possible to do what you are looking for. It is actually easy. There is a problem that I have found with ser/openser.. Documentation is difficult to read and some things are just not there, so you get people that spend many hours trying to get these functions to work. In these days time is money, so the people that know how to do what you are seeking.. charge large amounts of money for a simple 50 line config file. I will tell you that everything you are looking for is documented in examples. You will have to piece them together and make them work in harmony like the rest of us have. I suggest you look at voip user and piece the config together from examples there. It may also help you to read the source code of the modules that handle routing in ser. There are a few tricks that are hidden in the code. I am sorry for my vagueness. I am not able to share the config information due to an IP agreement with my company.(They think it is a trade secret) I wish you the best. Cheers, Zac Amsler, Network Operations Sur-Tel Communications, Inc. NetIQ Systems, LLC * US48, Canada, A-Z Wholesale Termination. * US48 Origination, Toll Free DIDs. * Toll Free Termination (FREE). Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large http://www.voip-info.org/wiki-Asterisk+at+large or the asterisk_integration http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration page at openser.org http://openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Are you using app_meetme or app_conference
Hi, for call centers with voip phones and calls coming in via SIP and Zap, what app_ are people using to do: -conference -listening to conversation of agents Is app_meetme or app_conference? Does app_meetme still suffers from the need to transcode to slin? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UK Colocation services
Can anyone direct me to a colo provider in the UKwhere I can park an asterisk server and buy UK toll free inbound services over SIP? Thanks Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPID
DANIEL, AARON MATTHEW wrote: Has anyone successfully gotten rpid working between two phones through asterisk? Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 Aaron, RPID is supported in Asterisk but many phones do not support it. Try adding the following to sip.conf: sendrpid=yes trustrpid=yes If it is going to work with your phones, it will just work. If not, chances are your phone does not support RPID. You can always look at a SIP debug to make sure it is getting sent. Even if your phones do not support RPID, From: usually works just fine :). -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER with multiple asterisk deployment
It won't work, unless you make sure that transfers go through the same asterisk server as the orignal call went through. Using the SER dispatcher won't fix that. -Original Message-From: sip [mailto:[EMAIL PROTECTED]Sent: Wednesday, September 27, 2006 2:25 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: [EMAIL PROTECTED]Subject: Re: [asterisk-users] SER with multiple asterisk deploymentHow do you plan on choosing which Asterisk server to send the SIP requests? Truly random? Based on some sort of LCR methodology? Have you tried using the LCR module for SER to send the requests to asterisk? Not sure it would work, but it might be worth looking at. N. On Wed, 27 Sep 2006 21:34:33 +0200, Adi Simon wrote Hi Zac,Thank you so much for your sincere answer. What you brought up is exactly what I encountered when I tried to find a solution for this, the documentation is inconsistent and ambiguous, and everywhere I look I end up with outdated examples that make little or no sense in the good case, or just don't compile due to being so old in the bad case. This is very frustrating but just by reading what you wrotewas very uplifting for me. Thanks again,Adi. On 9/27/06, Zac Amsler [EMAIL PROTECTED] wrote: Adi, It is possible to do what you are looking for. It is actually easy. There is a problem that I have found with ser/openser.. Documentation is difficult to read and some things are just not there, so you get people that spend many hours trying to get these functions to work. In these days time is money, so the people that know how to do what you are seeking.. charge large amounts of money for a simple 50 line config file. I will tell you that everything you are looking for is documented in examples. You will have to piece them together and make them work in harmony like the rest of us have. I suggest you look at voip user and piece the config together from examples there. It may also help you to read the source code of the modules that handle routing in ser. There are a few tricks that are hidden in the code. I am sorry for my vagueness. I am not able to share the config information due to an IP agreement with my company.(They think it is a trade secret) I wish you the best. Cheers, Zac Amsler, Network Operations Sur-Tel Communications, Inc. NetIQ Systems, LLC * US48, Canada, A-Z Wholesale Termination. * US48 Origination, Toll Free DIDs. * Toll Free Termination (FREE). Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large http://www.voip-info.org/wiki-Asterisk+at+large or the asterisk_integration http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration page at openser.org http://openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER with multiple asterisk deployment
Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large http://www.voip-info.org/wiki-Asterisk+at+large or the asterisk_integration http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration page at openser.org http://openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. Adi, The dispatcher module should do what you want to do. Check it out here: http://www.openser.org/docs/modules/1.1.x/dispatcher.html They claim it is stateless but it should be possible to use the AVPs it sets to direct INVITEs, ACKs, and BYEs to the proper Asterisk (or whatever) boxes. However, you can also load balance based on source/destination URIs with the lcr module. P.S. - This is really more of an OpenSER/SER question. Did you try those mailing lists? I'd be happy to help you more there :). -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P Problem or Not?
With a TDM 400 card in the system, WHY do you even need ztdummy?? I thought that was a substitute when there was no other timing source The only time I have had to compile ztdummy is when there was NO card present. Of course, I could be wrong. Please enlighten John Novack Eddie Johnson Jr wrote: Hello, I am having a problem. I have aterisk1.2.11 installed on ubuntu server 6.06 dapper and I have the ztdummy module listed in /etc/modules directory however upon startup it will not load. I type lsmod | grep ztdummy and there is no output. I then ran a config file for asterisk to create startup config files and it installs. I restarted the server with the newly installed files, and I type asterisk and asterisk starts. I do this issue a command to connect remotely and receive the connection and I type zap show status and I have the following: Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board1 OK 0 0 ZTDUMMY/1 1 UNCONFUR 0 0 0 I type stop now, go back to the prompt type mdprobe ztdummy and it loads. I repeat the above procedures above and I get the same message. Any suggestsions? Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
Ronnie I have 4 non-PRIs connected to a Nortel 11C and I had played with PRI connections before and got them working. If you want to call me we can go over your set up and compare with mine. Kevin Savoy 701-774-4023 Novo1 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronnie Jones Sent: Wednesday, September 27, 2006 2:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P I have no experience on the Nortel side, but will comment on the timing thingie. The asterisk T1 card (port going to the Nortel) will always generate T1 timing on the transmit side of the T1. There is no way to turn it off (by T1 Spec's). So, letting the Nortel use CLOK = EXT is perfect. The sync parameter in /etc/zaptel.conf for that same T1 port should probably be set to zero, but that statement is somewhat dependent on what the other ports on the Asterisk T1 card are used for. If there are no other Asterisk T1 card ports in use, then I'd suggest setting the sync parameter to 1. If at least one other Asterisk T1 port is in use and goes to a central office, then turn that port's sync to 1 and the Nortel port sync to 0. (Keep in mind the digium T1 cards only have one clock on board, and syncing that clock to a T1 coming from a central office is the right thing to do. Once that clock is in sync, then the Nortel will sync to asterisk.) I'm a little confused with your last paragraph when you say the circuit does establish and pass calls but resets frequently due to slips. Are those calls to/from asterisk talking to the Nortel? Yes that is correct. The Nortel switch connects to the PSTN but not the Asterisk. It connects to the Nortel. While the circuit is up I can call extensions on the Nortel from the Asterisk and visa versa. Or, are you routing incoming pstn calls from the central office through asterisk to the Nortel? No Also, have you tried any of the pri show ... commands in asterisk, or any of the pri debug items? Yes. When the circuit is up I can pri show span 1 and it show partitioned up and active. Ronnie Jones Engineer - ICT Clay Electric Cooperative, Inc 352-473-8000 ext. 8272 352-473-1929(F) 352-745-0910(C) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are you using app_meetme or app_conference
On 15:27, Wed 27 Sep 06, Erick Perez wrote: Hi, for call centers with voip phones and calls coming in via SIP and Zap, what app_ are people using to do: We use SIP and IAX2 and SCCP (chan_sccp). Zap is not possible for us because we want to run it on OpenBSD and the zaptel is not ported to it yet. -conference -listening to conversation of agents ChanSpy works for this. Is app_meetme or app_conference? We use app_conference. We have to since there's no timer zaptel stuff for OpenBSD. Does app_meetme still suffers from the need to transcode to slin? I have no idea. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK Colocation services
On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Can anyone direct me to a colo provider in the UK where I can park an asterisk server and buy UK toll free inbound services over SIP? Thanks Probably more relevant on the asterisk-biz list. However I'd be interested to know what replies you get. I'm considering renting another 11 root server and installing my own asterisk soon. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
On 27 Sep 2006, at 20:16, Ronnie Jones wrote: Also, have you tried any of the pri show ... commands in asterisk, or any of the pri debug items? Yes. When the circuit is up I can pri show span 1 and it show partitioned up and active. One thing to note - changes to the timing parameter in zaptel.conf do not take effect on an asterisk 'reload' , you need to unload and load the zaptel driver. I've found it useful (on occasion) to power cycle the asterisk box too, as this _forces_ the far end of the E1 (T1 in your case) to start afresh. Tim. Tim Panton www.mexuar.com/cards.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SER with multiple asterisk deployment
Yeah... I wasn't really sure. I'm trying to think of a way and nothing comes to mind. The problem is that SER is sort of part stateful and part not, and isn't as concerned with a constant dialog as simply passing the SIP packets effectively. You might be able to couch some logic somehow that searched for a particular message tag on incoming packets and assigned messages with the same tag an identical flag in the DB (using an AVP), then checked the AVP later to determine the proper direction to route the SIP message. It would be easier, I imagine, to write your own SER module to handle the dispatching details and tag searching, though. All around, it sounds like it could be a mess. Something to play with, though, if you have time. N. On Wed, 27 Sep 2006 15:25:04 -0600, Douglas Garstang wrote It won't work, unless you make sure that transfers go through the same asterisk server as the orignal call went through. Using the SER dispatcher won't fix that. -Original Message- From: sip [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 27, 2006 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [asterisk-users] SER with multiple asterisk deployment How do you plan on choosing which Asterisk server to send the SIP requests? Truly random? Based on some sort of LCR methodology? Have you tried using the LCR module for SER to send the requests to asterisk? Not sure it would work, but it might be worth looking at. N. On Wed, 27 Sep 2006 21:34:33 +0200, Adi Simon wrote Hi Zac, Thank you so much for your sincere answer. What you brought up is exactly what I encountered when I tried to find a solution for this, the documentation is inconsistent and ambiguous, and everywhere I look I end up with outdated examples that make little or no sense in the good case, or just don't compile due to being so old in the bad case. This is very frustrating but just by reading what you wrote was very uplifting for me. Thanks again, Adi. On 9/27/06, Zac Amsler [EMAIL PROTECTED] wrote: Adi, It is possible to do what you are looking for. It is actually easy. There is a problem that I have found with ser/openser.. Documentation is difficult to read and some things are just not there, so you get people that spend many hours trying to get these functions to work. In these days time is money, so the people that know how to do what you are seeking.. charge large amounts of money for a simple 50 line config file. I will tell you that everything you are looking for is documented in examples. You will have to piece them together and make them work in harmony like the rest of us have. I suggest you look at voip user and piece the config together from examples there. It may also help you to read the source code of the modules that handle routing in ser. There are a few tricks that are hidden in the code. I am sorry for my vagueness. I am not able to share the config information due to an IP agreement with my company.(They think it is a trade secret) I wish you the best. Cheers, Zac Amsler, Network Operations Sur-Tel Communications, Inc. NetIQ Systems, LLC * US48, Canada, A-Z Wholesale Termination. * US48 Origination, Toll Free DIDs. * Toll Free Termination (FREE). Adi Simon wrote: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). Please don't direct me to Asterisk+At+Large http://www.voip-info.org/wiki-Asterisk+at+large or the asterisk_integration http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration page at openser.org http://openser.org as they are quite old and useless. What I seek are examples of ser.cfg or some advice from someone who actually managed to accomplish this. Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Got SIP response 415 Unacceptable Content-Type back from 192.168.1.209
I'm still getting these errors if anyone has any ideas I'd be truly appreciative. On 9/25/06, Mr. Jones [EMAIL PROTECTED] wrote: Could the problem is this: Content-Type: unknown? Reliably Transmitting (NAT) to 192.168.1.228:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK4bf724c9;rport From: sip:[EMAIL PROTECTED];user=phone;tag=as744e33c0 To: test guy sip:[EMAIL PROTECTED];tag=6583e0d3a15652bd Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 109 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: - Content-Type: unknown Subscription-State: active Content-Length: 0 --- asterisk*CLI -- SIP read from 192.168.1.228:5060: SIP/2.0 415 Unacceptable Content-Type Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK77316e5c;rport From: sip:[EMAIL PROTECTED];user=phone;tag=as744e33c0 To: test guy sip:[EMAIL PROTECTED];tag=6583e0d3a15652bd Call-ID: [EMAIL PROTECTED] CSeq: 108 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.9 Accept: application/sdp, application/simple-message-summary, application/octet-stream, application/pidf+xml, message/sipfrag;version=2.0 Content-Length: 0 On 9/25/06, Anthony Cennami [EMAIL PROTECTED] wrote: Bidirectional SIP trace usually helps in these situations. On 9/25/06, Mr. Jones [EMAIL PROTECTED] wrote: Hi Folks, Has anyone seen these errors repeatedly in the CLI? Incoming call: Got SIP response 415 Unacceptable Content-Type back from 192.168.1.209 We're using GXP-2000s. TIA, Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony D Cennami ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to detect dial tone on ZAP channel before dialling using TDM2400P
Hello allI have an asterisk box running Asterisk 1.2.8 and I installed a digium TDM2400 with 8 FXO ports. When I amke a call to the PSTN, the zap channel answers, and teh call goes through if a PSTN is connected to the answered port. However, if there is no dial tone in the answered channel, or if no POTS line is connected, the user gets no indication until the call time outs. I want * to be able to detect if there is a dialtone on the channel, before it dials, if not, to send a busy signal or choose another available channel. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are you using app_meetme or app_conference
Be careful when using heavily ChanSpy. We did couple of weeks ago and the result was having Asterisk crashing almost once every day. How heavy? around 4 people using it 8 hours a day, each one using ChanSpy every 3-5 mins. we were not able to find the exact reason, so just stop using it.Alyed Return-Path: [EMAIL PROTECTED] Wed Sep 27 15:06:03 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 27 Sep 2006 15:06:03 -0700 On 15:27, Wed 27 Sep 06, Erick Perez wrote: Hi, for call centers with voip phones and calls coming in via SIP and Zap, what app_ are people using to do:We use SIP and IAX2 and SCCP (chan_sccp). Zap is notpossible for us because we want to run it on OpenBSD and thezaptel is not ported to it yet. -conference -listening to conversation of agentsChanSpy works for this. Is app_meetme or app_conference?We use app_conference. We have to since there's no timerzaptel stuff for OpenBSD. Does app_meetme still suffers from the need to transcode to slin?I have no idea.-- Michiel van Baak[EMAIL PROTECTED]http://michiel.vanbaak.euGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD"Why is it drug addicts and computer afficionados are both called users?"___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS Text Send working with BT Text in the UK??
Hi all- In 2004, I set up a sms texting process for a UK customer, using the asterisk SMS command and BT's BT Text SMS facility. This has been running fine up until recently. A couple of weeks ago, I upgraded them from their old asterisk version (CVS 2004.07.25) to 1.2.9.1, and have been having trouble getting the SMS feature to work on this newer version. I'm connecting to BT via a BRI, running an updated bristuff. (was also running this configuration previously) I do note the differences called out in the documentation, mainly that smsq is used to set up parameters for the text to be sent, and I've changed my code appropriately. Here is what I try: smsq --motx-channel=Zap/g3/17094001 --motx-retries=0 0111222 Hello! This seems to start things happening, as I observe the following on the asterisk console: --- -- Attempting call on Zap/g3/17094001 for application SMS(0) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH Sep 27 20:47:51 NOTICE[13661]: channel.c:2455 __ast_request_and_dial: Don't know what to do with control frame 15 Channel Zap/7-1 was answered. Launching SMS(0) on Zap/7-1 -- SMS RX 93 00 6D -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- Channel 0/1, span 3 received AOC-E charging 0 units -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- Hungup 'Zap/7-1' Sep 27 20:47:59 NOTICE[13661]: pbx_spool.c:279 attempt_thread: Call completed to Zap/g3/17094001 --- From looking at the app_sms.c code, I seem to be connecting to BT ok, but it appears that the 92 code received from them indicates an error in the format. As other posts have suggested,I have tried the following: (a) going back to version 1.2.7.1 (same symptoms) and (b) increasing the wait for response delay (h-opause) -no effect either. I've also tried reverting to my 2 year old app_sms.c, which no longer compiles (as expected) Does anyone have asterisk SMS texting via BT working in the UK, using a recent asterisk version, and if so, can you please shed some light on this? Many thanks Scott Stingel www.evtmedia.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI on 1.4 Beta
Anyone else having trouble with MWI on 1.4 Beta? The messages are getting stored and I'm getting the emails but no stutter tone or MWI as far as I can tell. MARK. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
re: [asterisk-users] Spurious hangups on zaptel interface
I'm experiencing the same problems, but unfortunatelly haven't been able to associate them with any number since they appear to be random. But maybe we can do a little research about it, and hopefully find teh solution for both: are your PSTN lines POTS or E1/T1? can you make a couple of calls and post here the logs? would be nice if you can enable the full Asterisk log for a single call and post that one.Alyed Return-Path: [EMAIL PROTECTED] Wed Sep 27 14:00:14 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 27 Sep 2006 14:00:14 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 364712FCB46; Wed, 27 Sep 2006 13:21:04 -0700 (MST) We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of "flash" (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1Barry D. HasslerPresidentHCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED]+1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED]Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spurious hangups on zaptel interface
Barry D. Hassler wrote: We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of flash (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1 Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to detect dial tone on ZAP channel before dialling using TDM2400P
Naija Man wrote: Hello all I have an asterisk box running Asterisk 1.2.8 and I installed a digium TDM2400 with 8 FXO ports. When I amke a call to the PSTN, the zap channel answers, and teh call goes through if a PSTN is connected to the answered port. However, if there is no dial tone in the answered channel, or if no POTS line is connected, the user gets no indication until the call time outs. I want * to be able to detect if there is a dialtone on the channel, before it dials, if not, to send a busy signal or choose another available channel. Asterisk does not support this on analog ports. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
Likewise, Ronnie, we have 2 PRIs going to an 11C - let me know if I can help. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Sep 27, 2006, at 2:42 PM, Savoy, Kevin - Williston, ND wrote: Ronnie I have 4 non-PRI’s connected to a Nortel 11C and I had played with PRI connections before and got them working. If you want to call me we can go over your set up and compare with mine. Kevin Savoy 701-774-4023 Novo1 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronnie Jones Sent: Wednesday, September 27, 2006 2:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P I have no experience on the Nortel side, but will comment on the timing thingie. The asterisk T1 card (port going to the Nortel) will always generate T1 timing on the transmit side of the T1. There is no way to turn it off (by T1 Spec's). So, letting the Nortel use CLOK = EXT is perfect. The sync parameter in /etc/zaptel.conf for that same T1 port should probably be set to zero, but that statement is somewhat dependent on what the other ports on the Asterisk T1 card are used for. If there are no other Asterisk T1 card ports in use, then I'd suggest setting the sync parameter to 1. If at least one other Asterisk T1 port is in use and goes to a central office, then turn that port's sync to 1 and the Nortel port sync to 0. (Keep in mind the digium T1 cards only have one clock on board, and syncing that clock to a T1 coming from a central office is the right thing to do. Once that clock is in sync, then the Nortel will sync to asterisk.) I'm a little confused with your last paragraph when you say the circuit does establish and pass calls but resets frequently due to slips. Are those calls to/from asterisk talking to the Nortel? Yes that is correct. The Nortel switch connects to the PSTN but not the Asterisk. It connects to the Nortel. While the circuit is up I can call extensions on the Nortel from the Asterisk and visa versa. Or, are you routing incoming pstn calls from the central office through asterisk to the Nortel? No Also, have you tried any of the pri show ... commands in asterisk, or any of the pri debug items? Yes. When the circuit is up I can pri show span 1 and it show partitioned up and active. Ronnie Jones Engineer - ICT Clay Electric Cooperative, Inc 352-473-8000 ext. 8272 352-473-1929(F) 352-745-0910(C) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP on Asterisk, new install
I've managed to get asterisk going. For the moment, I simply wish to get a couple of SIP phones functional. One is a x-lite softphone, the other a generic hard (sip) phone. Each connects to asterisk and will give me a dial tone, and accept key input. But neither can speak to the other, call never completes. A push in the right direction? Better, a gentle nudge. joe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spurious hangups on zaptel interface
I'm curious... why will this work?? busydetect will just cut the line if there are 4 tones (les or more depending the busycount param), and call progress will in fact try not to cut the call due to false hangups.Alyed Return-Path: [EMAIL PROTECTED] Wed Sep 27 16:12:13 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 27 Sep 2006 16:12:13 -0700 Barry D. Hassler wrote: We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of "flash" (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spurious hangups on zaptel interface
Both can cause random hangups. This is a well known issue. It even says in the sample configs that these features are prone to false positives. Alyed Tzompa wrote: I'm curious... why will this work?? busydetect will just cut the line if there are 4 tones (les or more depending the busycount param), and call progress will in fact try not to cut the call due to false hangups. Alyed Return-Path: [EMAIL PROTECTED] Wed Sep 27 16:12:13 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 27 Sep 2006 16:12:13 -0700 Barry D. Hassler wrote: We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of flash (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1 Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP on Asterisk, new install
Well, never mind. I seem to have found some docs that may assist. joe joe, at j4computers[EMAIL PROTECTED] Wrote on: 9/27/2006 7:22 PM: I've managed to get asterisk going. For the moment, I simply wish to get a couple of SIP phones functional. One is a x-lite softphone, the other a generic hard (sip) phone. Each connects to asterisk and will give me a dial tone, and accept key input. But neither can speak to the other, call never completes. A push in the right direction? Better, a gentle nudge. joe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with trying to use two extensions for different announcements
Hello Folks, First post. I am using a Trixbox 1.1.1 version and have been working with it for a few weeks, experimenting and trying to learn. I have decided to set-up the box as a phone system for a community organization/club in our area. I have tried to use FreePBX to make all the changes to Asterisk. I know that you die-hard linux guys like to do it all from a command line, but I am really a Windows person, trying to learn Linux, but who like graphical interfaces ;-) Anyway I have set up several extensions - one (ext 10) for an announcement for the club's next meeting (plays a recorded message), an extension (ext 11) for the club president (which goes to voicemail and then gets emailed to the president), an extension (12) for the head committee chair (which goes to voicemail and gets emailed to her), an extension (ext 13) that will provide basic club information to a non-member (SUPPOSE to play a recorded announcement of club information), an extension (ext 15) that provides current area weather, an extension (ext 16) that provides the current time, and finally and extension (ext 20) which goes directly to me. To accomplish this, I have three autoattendants (digital receptionists) set up. The main one answers the phone and plays a recorded message of the extensions. The two other autoattendants are set-up to handle the two extensions (10 and 13) that provide two different announcements. There are two recordings made using the System Recording function of the FreePBX. Each recording is assigned to each of the two autoattendants. Extension 10 works fine and announces the next club meeting then hangs up. However, when I dial ext 13, it does NOT play the correct recorded .wav file - it plays the message the is associated with ext 10. On the Trixbox forum, someone suggested this: Create a custom script in your extensions_custom.conf under [from-internal-custom] exten = 121,1,Answer exten = 121,2,Playback(/tmp/announcement) exten = 121,3,Hangup exten = h,1,Hangup assuming that 121 is the extension that you want people to call. When the extension is called your recorded announcement will be played. After the announcement is played it will hang up. You can also create a custom entry for calling from IVR and the IVR choice pointing to custom apps. [custom-announcement] exten = 121,1,Answer exten = 121,2,Playback(/tmp/announcement) exten = 121,3,Hangup exten = h,1,Hangup So I tried the second method, and changed the directory structure to make it look in the correct directory where FreePBX stores the recorded .wav files. That still does not work. It still goes to the wrong .wav file used by the other autoattendant. I even rerecorded the announcements. It is puzzling because each extension uses its own autoattendant/IVR/digital assistant and each has a separate .wav file associated with it, yet ext. 13 wants to grab the other audio file. It must be something obvious that I am missing or maybe I am not naming things correctly so it goes to the first available .wav file - I just don't know. Anybody know what I am possibly doing wrong? Thanks, Duffy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Termination
We are looking at putting an asterisks box in place and I was curious to know who you guys recommend for termination DID's? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
On Wed, Sep 27, 2006 at 12:02:48PM -0600, Colin Anderson wrote: Were those people -- who, unlike me, had done it and had problems -- wrong? There are more variables than the Digium card itself. Things like bus design, chipset etc all come into play. I've noticed that there is a concerted effort with Asterisk implmentors to often roll out Asterisk in a white box clone with a $129 integrated motherboard in an effort to drive down the cost which (IMO) is foolish. Why would you (not you, but people) spec a DL380G4 for a database server then turn around and use an ECS brand motherboard for a telephony platform - a platform which by definition requires sub-millisecond response time? Tier 1 boxes are *designed* for this I concur with your approach, but Tier 1 means as little here as it does when evaluating Internet backbone carriers. could you expand on what evaluation criteria you use? I'm going to be pre-speccing some stuff myself this month... type of application and are engineered to conform to spec + safety margin, wheras Taiwan clone boards are usually designed to roughly confirm to spec, and that's all (notable exception: I have used ASUS motherboards for Asterisk installs, and they all work flawlessly) I've always liked ASUS boards, though some of *them* have problems too. I've said it many times before on the list: It's trivial to make a crappy Asterisk install. Anyone can do it. It's really, really, hard to make a *good* Asterisk install. You need cross-discipline experience, a lot of which is hard to come by in the closed, secretive telephony world. Tip o' the hat to SHSU. I wouldn't touch *that* install with a space tether. Has anyone *interviewed* those implementors? Should I go do it? Does anyone know what Computer Telephony is paying these days? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
which is hard to come by in the closed, secretive telephony world. Tip o' the hat to SHSU. I wouldn't touch *that* install with a space tether. Has anyone *interviewed* those implementors?Should I go do it? All you gotta do is say Hey Aaron, how'd you do such and such and I'm sure he'd be more thanhappy to tell you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination
Verizon. On 9/27/06, Duracom Lists [EMAIL PROTECTED] wrote: We are looking at putting an asterisks box in place and I was curious to know who you guys recommend for termination DID's? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
