[asterisk-users] MySQL question
Can someone point me to some info or provide a summary on how to allow ODBC access from other hosts to the Asterisk database? Wesley A. SchochetSenior Telecommunications EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[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
[Asterisk-Users] MWI
I am using an external voice mail system. I'd like to be able to light the message waiting light on SIP and SCCP phones. Can someone point me in the right direction? Is there a manager command or and AGI app that does this. If not, what would I have to do to interface with * and have the MWI light work? Wesley A. SchochetSenior Telecommunications EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[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
[Asterisk-Users] 7960-tones.xml
I am afraid I have the wrong version of this file that somehow got loaded. Does anyone have a US version? How about 7960-fonts.xml? Thanks, Wes Wesley A. SchochetSenior Telecommunications EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[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
[Asterisk-Users] Cisco 7960 dialing trouble
Hi- Ihave a 7960 using chan_sccp which gives me a busy signal as soon as I dial the 1 in my string of 91555222. Can't figure out why. I do have a dialplan.xml file: DIALTEMPLATE TEMPLATE MATCH="*" Timeout="5"/ TEMPLATE MATCH="#..." Timeout="5"//DIALTEMPLATE Anyone have any insights? Thanks Wesley A. SchochetSenior Telecommunications EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[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] Waiting for your help...
You have to make all of your manual changes in the _custom.conf files. [EMAIL PROTECTED] overwrites the xxx.conf files -I think this happens every time you restart the app. Log files are usually in /var/log/asterisk and you can see them in the maintenance screen on AMP From: yrving rivas [mailto:[EMAIL PROTECTED] Sent: Monday, February 13, 2006 8:37 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Waiting for your help... Thanks Tzafrir: I am apologize because of my language problems writing the subject. I am not good at english. So thanks for telling me I am doing it the wrong way, and I will be more carefully next time. Help me if it is possible to you. The Asterisk version is 2.1 wich I downloaded trhough http://asteriskathome.sourceforge.net/. To install de fax to email support I followed the instructions in http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+8 How do I trace and how do I post my configuration files?. I have some about asterisk programming, I know some in programming, and a good learner Plz. YrvingTzafrir Cohen [EMAIL PROTECTED] escribió: On Mon, Feb 13, 2006 at 05:25:37AM -0600, yrving rivas wrote: Hello every one. This is a question done by me, not yet answered. Please, help.How about a decent subject for your message? I: 1. Run install-pdf from linux to support faxes on my asterisk.Of what software package, exactly?What version?What version of Asterisk?What OS/distro? What version of it? 2. Made the configurations throuhg AMP in a. Setup-Inbound Routing-(the only route I have)-fax extension-System b. Setup-Inbound Routing-(the only route I have)-fax email-(my email) c. Setup-Inbound Routing-(the only route I have)-Immediate Answer- yes d. Setup-Inbound Routing-(the only route I have)-pause after answer- 2 e. Setup-General Settings-fax machine for receiving faxes-system f. Setup-General Settings-Email address to have-(my email) 3. as a good boy made a test call from a fax, and it reports that couldn´t send the fax ( what means the aste risk couldn´t receive it). I didn´t receive any fax. What can I do to receive them? Tips: 1- In my configuration I have a TDM04B. 2- I receive via email the voice mail messages left to any extension.Looks like a CLI trace would come in handy. In other hand (and not related to this case, as you will see):No, I'm not sure. AMP's dialplan is a mess, and there's no telling whata naive change to it will do. I made changes to the extensions.conf file through AMP to construct a call forward on no answer, but at the next day all programming was like at beggining. What should I do to make the changes for ever? amp normally does not override extensions.conf (except, maybe on upgradetime).Anyway, posting your modified extentions.conf may help. Yourextensions_additional.conf may help as well. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | bestICQ# 16849755 | | friend___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --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] chan_sccp .conf changes
It appears that every time I want to add a sccp phone to my system, I have to reinitialize (unload and load) the chan_sccp.so module. Is this the case - can I add a phone without taking down all of the rest? ___ --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 2620 as PRI gateway
I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway out there somewhere, I wouldn't be too proud to take a look at that either! Asterisk configs would be great too! Thanks, Wes ___ --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] AAH out bound routing problem
Ram-
You are three steps ahead of where you need to
be. You need to figure out what to send before you figure out how to send
it.
Add a test extension in your
extensions_custom.conf:
exten =
3852,1,Dial(sip/easycall/19197543700,30)
dial
3852 and aee if it works. If not, try:
exten =
3852,1,Dial(sip/easycall/9197543700,30)
If that doesn't work, then I'd get in touch with
easycall and ask them what they want to see in a dial
string.
Once you get the above to work, then you can mess with
the AAH settings to try and send the right dial string!
From: ram [mailto:[EMAIL PROTECTED]
Sent: Friday, January 27, 2006 10:06 PMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]
AAH out bound routing problem
Hi
i have added the Dial pattern as you said
1NXXNXX9|.NXXNXXNXX
but still i get all circuits are busy now call later
ram
On 1/28/06, J.Raborg
[EMAIL PROTECTED]
wrote:
Go
to the setup/ Outbound Routing/ pick
the one you use for out must be 0 9out by default and addin the dial
patterns 9|. -including the dotThen try to dial
919197543700JRram wrote:
Hi rajeev
i have posted the extension.conf before
now iam posting extension_additional.conf
[EMAIL PROTECTED] asterisk]# more
extensions_additional.conf[globals]#include
globals_custom.confVM_PREFIX = *RINGTIMER = 15REGTIME =
7:55-17:05REGDAYS = mon-friRECORDEXTEN = ""PARKNOTIFY =
SIP/200OUT_2 = SIP/easycallOUTPREFIX_2 =OUTMAXCHANS_2 = 1
OUTCID_2 = outside account
OPERATOR =NULL = ""IN_OVERRIDE = forcereghoursINCOMING =
group-allFAX_RX_EMAIL = [EMAIL PROTECTED] FAX_RX
= systemFAX =DIRECTORY_OPTS =DIRECTORY = last DIAL_OUT =
9DIAL_OPTIONS = trDIALOUTIDS = 2/CALLFILENAME =
""AFTER_INCOMING =
[ext-local]include = ext-local-customexten =
1000,1,Macro(exten-vm,1000,1000)exten =
${VM_PREFIX}1000,1,Macro(vm,1000)exten = 1000,hint,SIP/1000
[outbound-allroutes]include =
outbound-allroutes-custominclude = outrt-001-longdistance
[outrt-001-longdistance]include =
outrt-001-longdistance-customexten =
_1NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)exten =
_1NXXNXX,2,Macro(outisbusy) ;
No available circuitsexten =
_NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},) exten =
_NXXNXX,2,Macro(outisbusy) ; No available circuitsexten =
_NXX,1,Macro(dialout-trunk,2,${EXTEN},)exten =
_NXX,2,Macro(outisbusy) ; No available
circuits
ram
On 1/27/06, Rajeev
Natarajan [EMAIL PROTECTED]
wrote:
if
you are using AAH, please post
extensions.conf,extensions_additional.conf - also send us more info on
your phones. thanksrajeevram wrote:
Hi all of them thanks for the quick reply
i was tried adding 9 as well as 00 but i get number invalid if i
put any of the digits what kind of config files need to
post here to resolve the problem please
assists ram On 1/27/06, *Michael
Collins*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote: Ram,
On my AAH the
stock dial plan requires a 9 first.For kicks, try
dialing 919197543700 and see what you
get.
-MC
*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED] ] *On
Behalf Of *ram *Sent:* Friday, January 27,
2006 6:14 AM *To:* Asterisk Users Mailing
List - Non-Commercial Discussion
*Subject:* [Asterisk-Users] AAH out bound routing problem
Hi
all I have
installed AAH 2.2 in my P4
PC following AAH
handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as
per the guide says
and downloaded SJ Phone, and registered
user and when i
try to dial the 19197543700
i get
message that, all circuits are busy now, please try your
call
later and when i
see in the console i get this mesage
any
help Called
easycall/19197543700
-- Got SIP response 488 "Not acceptable here" back from
(PeerIP) --
SIP/easycall-838e is circuit-busy
ram
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[Asterisk-Users] Meetmee weirdness
I have several instances where conference calls are not being torn down appropriately. My CDR shows 3000 minute calls, which are coming in on PRI. I know that the calls aren't really lasting thatlong. What could be causing this? IN fact, here is what shows now: asterisk*CLI meetmeConf Num Parties Marked Activity Creation138 N/A 11:40:40 Static8472 N/A 12:21:53 Static8477 N/A 88:32:19 Static8478 N/A 108:39:05 Static Why are these calls still active if there are no Parties? I also see too many (?) dummy channels? asterisk*CLI zap show channels Chan Extension Context Language MusicOnHoldpseudo from-internalpseudo from-internalpseudo from-internalpseudo from-internalpseudo from-internal 1 from-internal 2 from-internal 3 from-internal 4 from-internal 5 from-internal 6 from-internal 7 from-internal 8 from-internal 9 from-internal 10 from-internal 11 from-internal 12 from-internal 13 from-internal 14 from-internal 15 from-internal 16 from-internal 17 from-internal 18 from-internal 19 from-internal 20 from-internal 21 from-internal 22 from-internal 23 from-internal P.S. I spend about 3 hours last week trying to figure out why I could get a new PIN to take for a conference room. Turned out that there was still an "active" call up, so it wouldn't re-read the meetme.conf file! ___ --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] trunk to trunk forwarding
Your carrier may offer a take back and transfer service - for a fee. That's the dtmf tones you sometimes hear during IVR sessions to large faceless companies. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 25, 2006 4:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] trunk to trunk forwarding -Original Message- From: Nic Hughes [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 25, 2006 4:49 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] trunk to trunk forwarding Hi all, Has anyone implemented trunk to trunk forwarding with an asterisk PBX. For the purpose I have in mind its quite important that once the call has been sent onwards to the new desination the lines into the PBX are no longer held. If anyone has UK-specific experience of getting this up and running that would be incredibly useful! Nic This can easily be done using VoIP but I don't think it will work using TDM (unless this is a feature that your Telco offers). With TDM you will utilizing two channels (one coming in bridged with one going out) for the entirety of the call. Thanks, Steve ___ --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] Nortel Meridian Opt 81C and PRI
Your trunk is set for pulse, should probably be set for tone - that may help? CLS DTN instead of the (archaic default) DIP -Original Message- From: Greg Camp [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 25, 2006 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI This is on an MSDL card. What is your individual trunk member config? Our zapata.conf / zaptel.conf are basically the same. No, they are exactly the same Interesting side note: we located an old Option 11 (release 19 with a cartridge!) and were able to get the below config to work just fine. Obviously there are some subtle differences since there is no MSDL on an Option 11, but other than the missing prompts due to an older OS on the Nortel, the config is the same. We don't really want to leave the Option 11 running just to support Asterisk. Do you get name display on your Asterisk phones when someone from the Nortel calls? That's one thing that doesn't happen on the Opt11. Names do get displayed on the Nortel phone when called from an Asterisk phone, however. REQ issp VERSION 2611 RELEASE 25 ISSUE 30 + MDCS01 NA00 PSWV VERSION: PSWV 53 cequ DLOP NUM DCH FRM LCMT YALM TRSH PRI 088 23 ESF B8S FDL 00 ADAN DCH 88 CTYP MSDL GRP 0 DNUM 12 PORT 0 DES asterisk USR PRI DCHL 88 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 T200 3 T203 10 N200 3 N201 260 K7 TN 088 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 38 1 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DIP CND WTA LPR APN THFD HKD P10 VNL TKID DATE 24 JAN 2006 TYPE RDB CUST 00 ROUT 38 DES ASTERISK TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT DTRK YES DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 0 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC NO 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 8138 TCPP NO PII NO TARG 01 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 NBL 4096 IENB 5 PAGE 002 TFD 0 DRNG NO CDR 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 PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Schochet, Wes Sent: Tuesday, January 24, 2006 8:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI I have a 61c running steady and reliable. ENLL in 60 will always bring up the DCHannel, whether it's disabled in 96 or not. Lets see some M1 configs - I'll show you mine if... LD 22 REQ prt TYPE cequ blah blah DLOP NUM DCH FRM TMDI LCMT YALM TRSH TRK 020 24 ESF NO B8S FDL 00 PRI 002 24 ESF NO B8S FDL 00 blah blah 019 23 ESF NO B8S FDL 00 blah blah REQ PRT TYPE ADAN DCH 5 ADAN DCH 5 CTYP MSDL DNUM 10 PORT 2 DES Asterisk USR PRI DCHL 19 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 ld 21 PT1000 REQ: PRT TYPE: RDB CUST 0 ROUT 19 TYPE RDB CUST 00 ROUT 19 DES ASTERISK 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 0 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC NO 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 2019 TCPP NO PII NO TARG 01 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 Dchannel daughter card on a NT5D12 or an actual MSDL? Jumpers can kill! Asterisk Files: zaptel.conf: loadzone = us defaultzone = us #span definitions span = 1,1,0,esf,b8zs #channel definitions bchan = 1-23 dchan = 24
RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI
I believe it's now called 3.0... REQ iss VERSION 2521 RELEASE 3 ISSUE 00 + DepList x210300_cpt Yup 3.0 -Original Message- From: Greg Camp [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 25, 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI We had been using DTN, but when DIP worked on the Option 11 we gave it a try. (We were following the instructions in asterisk-meridian-a1.pdf, and it didn't specify to use DTN on CLS so we were just taking the defaults.) What release is your 61? Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Schochet, Wes Sent: Wednesday, January 25, 2006 10:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI Your trunk is set for pulse, should probably be set for tone - that may help? CLS DTN instead of the (archaic default) DIP -Original Message- From: Greg Camp [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 25, 2006 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI This is on an MSDL card. What is your individual trunk member config? Our zapata.conf / zaptel.conf are basically the same. No, they are exactly the same Interesting side note: we located an old Option 11 (release 19 with a cartridge!) and were able to get the below config to work just fine. Obviously there are some subtle differences since there is no MSDL on an Option 11, but other than the missing prompts due to an older OS on the Nortel, the config is the same. We don't really want to leave the Option 11 running just to support Asterisk. Do you get name display on your Asterisk phones when someone from the Nortel calls? That's one thing that doesn't happen on the Opt11. Names do get displayed on the Nortel phone when called from an Asterisk phone, however. REQ issp VERSION 2611 RELEASE 25 ISSUE 30 + MDCS01 NA00 PSWV VERSION: PSWV 53 cequ DLOP NUM DCH FRM LCMT YALM TRSH PRI 088 23 ESF B8S FDL 00 ADAN DCH 88 CTYP MSDL GRP 0 DNUM 12 PORT 0 DES asterisk USR PRI DCHL 88 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 T200 3 T203 10 N200 3 N201 260 K7 TN 088 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 38 1 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DIP CND WTA LPR APN THFD HKD P10 VNL TKID DATE 24 JAN 2006 TYPE RDB CUST 00 ROUT 38 DES ASTERISK TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT DTRK YES DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 0 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC NO 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 8138 TCPP NO PII NO TARG 01 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 NBL 4096 IENB 5 PAGE 002 TFD 0 DRNG NO CDR 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 PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Schochet, Wes Sent: Tuesday, January 24, 2006 8:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI I have a 61c running steady and reliable. ENLL in 60 will always bring up the DCHannel, whether it's disabled in 96 or not. Lets see some M1 configs - I'll show you mine if... LD 22 REQ prt TYPE cequ blah blah DLOP NUM DCH FRM TMDI LCMT YALM TRSH TRK 020 24 ESF NO B8S FDL 00 PRI 002 24 ESF NO B8S FDL 00 blah blah 019 23 ESF NO B8S FDL 00 blah blah REQ PRT TYPE ADAN DCH 5 ADAN DCH 5 CTYP MSDL DNUM 10 PORT 2 DES Asterisk USR PRI DCHL 19 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 ld 21 PT1000 REQ: PRT TYPE: RDB CUST 0 ROUT 19 TYPE RDB CUST 00 ROUT 19 DES ASTERISK TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES
RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI
As I think about it, this sounds a lot like an issue I had at one point when I had the tip and ring of one of the T1 pairs reversed. Sad, but true. -Original Message- From: Schochet, Wes Sent: Wednesday, January 25, 2006 1:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI I believe it's now called 3.0... REQ iss VERSION 2521 RELEASE 3 ISSUE 00 + DepList x210300_cpt Yup 3.0 -Original Message- From: Greg Camp [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 25, 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI We had been using DTN, but when DIP worked on the Option 11 we gave it a try. (We were following the instructions in asterisk-meridian-a1.pdf, and it didn't specify to use DTN on CLS so we were just taking the defaults.) What release is your 61? Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Schochet, Wes Sent: Wednesday, January 25, 2006 10:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI Your trunk is set for pulse, should probably be set for tone - that may help? CLS DTN instead of the (archaic default) DIP -Original Message- From: Greg Camp [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 25, 2006 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI This is on an MSDL card. What is your individual trunk member config? Our zapata.conf / zaptel.conf are basically the same. No, they are exactly the same Interesting side note: we located an old Option 11 (release 19 with a cartridge!) and were able to get the below config to work just fine. Obviously there are some subtle differences since there is no MSDL on an Option 11, but other than the missing prompts due to an older OS on the Nortel, the config is the same. We don't really want to leave the Option 11 running just to support Asterisk. Do you get name display on your Asterisk phones when someone from the Nortel calls? That's one thing that doesn't happen on the Opt11. Names do get displayed on the Nortel phone when called from an Asterisk phone, however. REQ issp VERSION 2611 RELEASE 25 ISSUE 30 + MDCS01 NA00 PSWV VERSION: PSWV 53 cequ DLOP NUM DCH FRM LCMT YALM TRSH PRI 088 23 ESF B8S FDL 00 ADAN DCH 88 CTYP MSDL GRP 0 DNUM 12 PORT 0 DES asterisk USR PRI DCHL 88 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 T200 3 T203 10 N200 3 N201 260 K7 TN 088 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 38 1 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DIP CND WTA LPR APN THFD HKD P10 VNL TKID DATE 24 JAN 2006 TYPE RDB CUST 00 ROUT 38 DES ASTERISK TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT DTRK YES DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 0 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC NO 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 8138 TCPP NO PII NO TARG 01 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 NBL 4096 IENB 5 PAGE 002 TFD 0 DRNG NO CDR 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 PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Schochet, Wes Sent: Tuesday, January 24, 2006 8:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI I have a 61c running steady and reliable. ENLL in 60 will always bring up the DCHannel, whether it's disabled in 96 or not. Lets see some M1 configs - I'll show you mine if... LD 22 REQ prt TYPE cequ blah blah DLOP NUM DCH FRM TMDI LCMT YALM TRSH TRK 020 24 ESF NO B8S FDL 00 PRI 002 24 ESF NO B8S FDL 00 blah blah 019 23 ESF NO B8S FDL 00 blah blah REQ PRT TYPE ADAN DCH 5 ADAN DCH 5 CTYP MSDL DNUM 10 PORT 2 DES Asterisk USR PRI DCHL 19 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1
[Asterisk-Users] Dial String Questions
Hi all- My TDM long distance is provided by MCI. We use account codes where MCI sends a challenge tone after receiving 1NXXNXX. Anyone have any suggestions of how to accomplish this? I can't get the soft phones to send the DTMF (the other digits go down the d-channel of our PRI). I also have not bee able to get the dial or the outgoing queue command to work. Anyone run into this? Wesley A. SchochetSenior Telecommunications EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[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] Nortel Meridian Opt 81C and PRI
I have a 61c running steady and reliable. ENLL in 60 will always bring up the DCHannel, whether it's disabled in 96 or not. Lets see some M1 configs - I'll show you mine if... LD 22 REQ prt TYPE cequ blah blah DLOP NUM DCH FRM TMDI LCMT YALM TRSH TRK 020 24 ESF NO B8S FDL 00 PRI 002 24 ESF NO B8S FDL 00 blah blah 019 23 ESF NO B8S FDL 00 blah blah REQ PRT TYPE ADAN DCH 5 ADAN DCH 5 CTYP MSDL DNUM 10 PORT 2 DES Asterisk USR PRI DCHL 19 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 ld 21 PT1000 REQ: PRT TYPE: RDB CUST 0 ROUT 19 TYPE RDB CUST 00 ROUT 19 DES ASTERISK 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 0 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC NO 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 2019 TCPP NO PII NO TARG 01 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 Dchannel daughter card on a NT5D12 or an actual MSDL? Jumpers can kill! Asterisk Files: zaptel.conf: loadzone = us defaultzone = us #span definitions span = 1,1,0,esf,b8zs #channel definitions bchan = 1-23 dchan = 24 zapata.conf [channels] context = from-internal switchtype = 5ess usecallerid = yes echocancel = yes echocancelwhenbridged = yes rxgain = 0.0 txgain = 0.0 signalling = pri_net group = 1 channel = 1-23 I am using a sangoma A101 (which rocks, btw). Should be the same diff... -Original Message- From: Greg Camp [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 24, 2006 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI Yes, we have disabled the dch and loop several times (both on Nortel and Asterisk sides). We also disabled the entire MSDL card and issued a full download to both the MSDL and the DCH. Neither got us any further. Have you had success getting PRI between a Nortel 81C and Asterisk to work? Thanks, Greg OK, but did you do the sequence as I mentioned, probably I guess. 1)dis dch, 2)dis loop, 3)enable loop, 4)enable dch as below - Unfortunately, although I've supported multiple 81's I've only had a requirement to connect an Asterisk to a little Avaya INDeX. Steve Yes, we followed that sequence. We've had to follow that sequence with some of our carriers as well to make the b-channels come up properly. Greg ___ --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] Web Conferencing
Is there a good Web Conferencing add-on or a compatible package for Asterisk? I know there are web based controls for the audio, but I am looking for PowerPoint or desktop sharing functionality similar to WebEx. Anyone using a package they like? ___ --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] Exited non-zero
I am working on this app to dial two external numbers the second after the
first hangs up. I have simplified things down to:
exten = 3852,1,Dial(zap/g1/3964,10,g)
exten = 3852,2,Wait(2)
exten = 3852,3,Dial(zap/g1/7757,10,g)
exten = 3852,4,Hangup
Here is the debug:
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/3964|10|g) in new stack
-- Called g1/3964
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
Everything is OK here when someone picks up x3964. Then when x3964 is hung
up, the first Dial command executes fine, returning -1 (as per the docs
since it is disconnected by the far end). This causes a debug message of:
-- Hungup 'Zap/1-1'
== Spawn extension (from-internal, 3852, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
And execution halts.
I need to figure out either how to get the dial command to return 0 or get
the system to continue execution of the script despite the non-zero return.
Has anyone dealt with this before?
Thanks in advance,
Wes
-Original Message-
From: Schochet, Wes
Sent: Friday, January 13, 2006 2:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridging app
OK - this is great! However, I'm showing my lack of depth / newness here.
Calls from internal SIP phones work perfectly. Calls from external sources
(my PBX) fail. Obviously, I have a dialplan / context problem, but I'd
appreciate a brief explanation and some direction from the group!
In extensions.conf, I have [from-pstn]. Under that section, I have included
[ext-postcall]. Then I have the following in an included file:
[ext-postcall]
exten = 3852,1,Answer
exten = 3852,2,Dial(zap/g1/8030,10,g)
exten = 3852,3,wait(5)
exten = 3852,4,Dial(zap/g1/8041,10,g)
exten = 3852,5,wait(5)
exten = 3852,6,NoOp(${DIALSTATUS})
exten = 3852,7,Hangup
The trunk group g1 is a T1 to my PBX and in the zapata.conf file I have the
entry
context = from-pstn
..
..
Group = g1
Here is the trace from both an internal extension (205) and an external
extension.
From SIP/205 ( Context=from-internal in sip.conf ) asterisk*CLI
-- Executing Answer(SIP/205-1d7b, ) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8041|10|g) in new stack
-- Called g1/8041
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing NoOp(SIP/205-1d7b, ANSWER) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (from-internal, 3852, 7) exited non-zero on
'SIP/205-1d7b'
-- Executing Macro(SIP/205-1d7b, hangupcall) in new stack
-- Executing ResetCDR(SIP/205-1d7b, w) in new stack
-- Executing NoCDR(SIP/205-1d7b, ) in new stack
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/205-1d7b' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/205-1d7b'
asterisk*CLI
From External coming in Zap/g1 (from-pstn) :
asterisk*CLI
-- Executing Answer(Zap/23-1, ) in new stack
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (from-pstn, 3852, 2) exited non-zero on 'Zap/23-1'
-- Executing Macro(Zap/23-1, hangupcall) in new stack
-- Executing ResetCDR(Zap/23-1, w) in new stack
-- Executing NoCDR(Zap/23-1, ) in new stack
-- Executing Wait(Zap/23-1, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Zap/23-1'
in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
asterisk*CLI
-Original Message-
From: trixter aka Bret McDanel [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bridging app
There might be a simplier way. a channel variable that holds the users
response, and a gotoif. You should be able to pass 'g' to dial which
according to http://www.voip-info.org/wiki-Asterisk+cmd+Dial
g: When the called party hangs up, exit to execute more commands in the
current context.
So the agent just hangs up and the IVR will continue with the caller into
your survey
[Asterisk-Users] FW: Exited non-zero
I am working on this app to dial two external numbers. The second is dialed
after the first hangs up. I have simplified things down to:
exten = 3852,1,Dial(zap/g1/3964,10,g)
exten = 3852,2,Wait(2)
exten = 3852,3,Dial(zap/g1/7757,10,g)
exten = 3852,4,Hangup
Here is the debug:
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/3964|10|g) in new stack
-- Called g1/3964
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
Everything is OK here when someone picks up x3964. Then when x3964 is hung
up, the first Dial command executes fine, returning -1 (as per the docs
since it is disconnected by the far end). This causes a debug message of:
-- Hungup 'Zap/1-1'
== Spawn extension (from-internal, 3852, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
And execution halts.
I need to figure out either how to get the dial command to return 0 or get
the system to continue execution of the script despite the non-zero return.
Is there like an error handler / trap type routine that I can use in a
dialplan? I am going to try and dig through the source code of the dial
command, but there has got to be a better way...
Has anyone dealt with this before?
Thanks in advance,
Wes
-Original Message-
From: Schochet, Wes
Sent: Friday, January 13, 2006 2:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridging app
OK - this is great! However, I'm showing my lack of depth / newness here.
Calls from internal SIP phones work perfectly. Calls from external sources
(my PBX) fail. Obviously, I have a dialplan / context problem, but I'd
appreciate a brief explanation and some direction from the group!
In extensions.conf, I have [from-pstn]. Under that section, I have included
[ext-postcall]. Then I have the following in an included file:
[ext-postcall]
exten = 3852,1,Answer
exten = 3852,2,Dial(zap/g1/8030,10,g)
exten = 3852,3,wait(5)
exten = 3852,4,Dial(zap/g1/8041,10,g)
exten = 3852,5,wait(5)
exten = 3852,6,NoOp(${DIALSTATUS})
exten = 3852,7,Hangup
The trunk group g1 is a T1 to my PBX and in the zapata.conf file I have the
entry
context = from-pstn
..
..
Group = g1
Here is the trace from both an internal extension (205) and an external
extension.
From SIP/205 ( Context=from-internal in sip.conf ) asterisk*CLI
-- Executing Answer(SIP/205-1d7b, ) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8041|10|g) in new stack
-- Called g1/8041
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing NoOp(SIP/205-1d7b, ANSWER) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (from-internal, 3852, 7) exited non-zero on
'SIP/205-1d7b'
-- Executing Macro(SIP/205-1d7b, hangupcall) in new stack
-- Executing ResetCDR(SIP/205-1d7b, w) in new stack
-- Executing NoCDR(SIP/205-1d7b, ) in new stack
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/205-1d7b' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/205-1d7b'
asterisk*CLI
From External coming in Zap/g1 (from-pstn) :
asterisk*CLI
-- Executing Answer(Zap/23-1, ) in new stack
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (from-pstn, 3852, 2) exited non-zero on 'Zap/23-1'
-- Executing Macro(Zap/23-1, hangupcall) in new stack
-- Executing ResetCDR(Zap/23-1, w) in new stack
-- Executing NoCDR(Zap/23-1, ) in new stack
-- Executing Wait(Zap/23-1, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Zap/23-1'
in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
asterisk*CLI
-Original Message-
From: trixter aka Bret McDanel [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bridging app
There might be a simplier way. a channel variable that holds the users
response, and a gotoif. You should be able to pass 'g' to dial which
according to http://www.voip-info.org
RE: [Asterisk-Users] Bridging app
Hmm. I'll give this a shot. Thanks - BTW, the agent doesn't know which calls will get surveyed - that kinda wrecks the survey! -Original Message- From: trixter aka Bret McDanel [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bridging app There might be a simplier way. a channel variable that holds the users response, and a gotoif. You should be able to pass 'g' to dial which according to http://www.voip-info.org/wiki-Asterisk+cmd+Dial g: When the called party hangs up, exit to execute more commands in the current context. So the agent just hangs up and the IVR will continue with the caller into your survey if they so selected, if not it just hangs up. That might be the easiest way to do this. You could even have the agent instructed based on that channel var (depending on your CRM integration) to tell the caller that they will be connected to the survey they opted to do so they dont forget and hangup too. On Thu, 2006-01-12 at 13:48 -0600, Schochet, Wes wrote: Hi All- I am trying to create a post call survey application. I would like to: 1. ask the caller if they want to take a survey after their call completes 2. If no, just transfer the call 3. if yes, 4. bridge up another extension 5. wait for that extension to hang-up 6. have the system (not the user) transfer the call to different extension that administers an IVR based survey. ___ --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] Bridging app
OK - this is great! However, I'm showing my lack of depth / newness here.
Calls from internal SIP phones work perfectly. Calls from external sources
(my PBX) fail. Obviously, I have a dialplan / context problem, but I'd
appreciate a brief explanation and some direction from the group!
In extensions.conf, I have [from-pstn]. Under that section, I have included
[ext-postcall]. Then I have the following in an included file:
[ext-postcall]
exten = 3852,1,Answer
exten = 3852,2,Dial(zap/g1/8030,10,g)
exten = 3852,3,wait(5)
exten = 3852,4,Dial(zap/g1/8041,10,g)
exten = 3852,5,wait(5)
exten = 3852,6,NoOp(${DIALSTATUS})
exten = 3852,7,Hangup
The trunk group g1 is a T1 to my PBX and in the zapata.conf file I have the
entry
context = from-pstn
..
..
Group = g1
Here is the trace from both an internal extension (205) and an external
extension.
From SIP/205 ( Context=from-internal in sip.conf )
asterisk*CLI
-- Executing Answer(SIP/205-1d7b, ) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8041|10|g) in new stack
-- Called g1/8041
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing NoOp(SIP/205-1d7b, ANSWER) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (from-internal, 3852, 7) exited non-zero on
'SIP/205-1d7b'
-- Executing Macro(SIP/205-1d7b, hangupcall) in new stack
-- Executing ResetCDR(SIP/205-1d7b, w) in new stack
-- Executing NoCDR(SIP/205-1d7b, ) in new stack
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/205-1d7b' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/205-1d7b'
asterisk*CLI
From External coming in Zap/g1 (from-pstn) :
asterisk*CLI
-- Executing Answer(Zap/23-1, ) in new stack
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (from-pstn, 3852, 2) exited non-zero on 'Zap/23-1'
-- Executing Macro(Zap/23-1, hangupcall) in new stack
-- Executing ResetCDR(Zap/23-1, w) in new stack
-- Executing NoCDR(Zap/23-1, ) in new stack
-- Executing Wait(Zap/23-1, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Zap/23-1'
in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
asterisk*CLI
-Original Message-
From: trixter aka Bret McDanel [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bridging app
There might be a simplier way. a channel variable that holds the users
response, and a gotoif. You should be able to pass 'g' to dial which
according to http://www.voip-info.org/wiki-Asterisk+cmd+Dial
g: When the called party hangs up, exit to execute more commands in the
current context.
So the agent just hangs up and the IVR will continue with the caller into
your survey if they so selected, if not it just hangs up. That might be the
easiest way to do this.
You could even have the agent instructed based on that channel var
(depending on your CRM integration) to tell the caller that they will be
connected to the survey they opted to do so they dont forget and hangup too.
On Thu, 2006-01-12 at 13:48 -0600, Schochet, Wes wrote:
Hi All-
I am trying to create a post call survey application. I would like
to:
1. ask the caller if they want to take a survey after their call
completes 2. If no, just transfer the call 3. if yes,
4. bridge up another extension
5. wait for that extension to hang-up
6. have the system (not the user) transfer the call to different
extension
that administers an IVR based survey.
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RE: [Asterisk-Users] Second edition of my * book has been release d
But for us? From: William Boehlke [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 2:24 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Second edition of my * book has been released $39.95 retail. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel SartorSent: Tuesday, January 10, 2006 6:27 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Second edition of my * book has been released How much is this book ?? 2006/1/10, Randy Williams [EMAIL PROTECTED]: Greetings All,I have found that Paul's book is just right for rounding out the edgeswhen getting started.I managed to temporarily migrate our T-1 Asterisk system to a Analog asterisk system on information in Paul's book alone.Nicely done and a neat bit of help in a pinch.Just my $0.02 USD you understand. :)RandyWPaul Mahler wrote:Hi Greg, My book is a good place for a beginner to get started. I also find it to beuseful as a reference for Asterisk. It's not an advanced book, there areadvanced features it doesn't cover, for example AGI or the management interface.It should be very helpful for your customers. It should be helpful for abeginning to intermediate administrator. I still frequently refer to itmyself when I'm having a senior moment. :) There isn't anything in the book that would make it less useful for the CVSor stable branches.The O'Reilly book is excellent. I think my book complements the O'Reillybook. If I were just starting I would buy both. I think my book may be a bit more useful as a reference. I think I cover a bit more beginner's territory.Hope This Helps,Paul-Original Message-From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED] ] On Behalf Of [EMAIL PROTECTED]Sent: Monday, January 09, 2006 9:10 PMTo: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Second edition of my * book has beenreleasedHow does it compare with the O'Rielly book?Does it include information on CVS, or primarily on stable? Can it be provided to customers, or is it more sysadmin oriented?Regards,Greg-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of PaulMahlerSent: Thursday, January 05, 2006 9:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Second edition of my * book has been releasedThe second edition of my Asterisk book "VoIP Telephony with Asterisk" is now in print. It's reorganized and expanded.TKSPaul MahlerPaul Mahler[EMAIL PROTECTED] www.signate.com___--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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--No virus found in this incoming message. Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006___--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 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] Bridging app
Hi All- I am trying to create a post call survey application. I would like to: 1. ask the caller if they want to take a survey after their call completes 2. If no, just transfer the call 3. if yes, 4. bridge up another extension 5. wait for that extension to hang-up 6. have the system (not the user) transfer the call to different extension that administers an IVR based survey. Anyone have any ideas how to do this. I can envision the whole thing except the bridging up the second user. Any assistance, input, or code would be appreciated! Thanks, Wes ___ --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] Unable to connect to Asterisk
Check manager.conf and manager.custom.conf (installed by amp) for access lists which may be preventing you from reaching it. -Original Message- From: Nitesh Divecha [mailto:[EMAIL PROTECTED] Sent: Monday, January 09, 2006 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to Asterisk Hello All Everything was working OK, and decided to install AMP 1.10.010... and problem started. AMP took control of Asterisk... For some odd reasons I can not connect to Asterisk CLI any more. I get the following error: - [EMAIL PROTECTED] ~]$ sudo /usr/sbin/asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) [EMAIL PROTECTED] ~]$ But if I check the process, I do see Asterisk is running. I am running Asterisk 1.2 Any ideas? Thanking in advance... Thanks, Neal ___ --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] Problem getting D channel up on Sangoma A102
Also, make sure the Asterisk application is running. The span will be clean without it, but the application itself generates the d-channel messages. -Original Message- From: David Yat Sin [mailto:[EMAIL PROTECTED] Sent: Friday, December 30, 2005 8:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] Problem getting D channel up on Sangoma A102 Try to recompile/reinstall (make clean; make install) zaptel after the wanpipe installation to have the new 'patched' zaptel modules installed on your system. David Yat Sin Sangoma Technologies (905) 474-1990 x119 (800) 388-2475 x119 Fax: (905) 474 9223 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] Website: www.sangoma.com Message: 10 Date: Thu, 29 Dec 2005 13:35:19 -0500 From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem getting D channel up on Sangoma A102 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hi all, I am installing an Asterisk box equipped with the Sangoma A102 card. The telco just tested the PRI interface and it is ll ok. I now connect my Asterisk box and I can't get the D-Channel up. If I enable intense pri debug I see messages like the following: --SNIP START-- [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter == Primary D-Channel on span 1 up pbx*CLI [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter == Primary D-Channel on span 1 up T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (0) [ 00 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter -- Retrying poll with f-bit Sending Receiver Ready (0) [ 00 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter Stopping T_203 timer T_200 timer already going (3) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 86] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 6 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- T200 counter expired, What to do... -- Retransmitting 17 bytes [ 00 01 00 01 08 02 00 00 46 18 03 a9 83 86 79 01 80 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 000 0: 0 N(R): 000 P: 1 13 bytes of data -- Rescheduling retransmission (2) -- T200 counter expired, What to do... -- Timeout occured, restarting PRI Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data == Primary D-Channel on span 1 down --SNIP END-- Config is the following: zaptel.conf: span=1,1,2,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf [channels] language=fr context=from-pstn switchtype=national resetinterval=never signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=1 channel=1-23 Any hints appreciated Andre ___ --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] @Home overwrites configs on startup
Hi Al- It appears that my config files are being overwritten on restart and/or reboot wit my [EMAIL PROTECTED] distribution. Anyone familiar with this behavior? I'd like to be able to set some of the reboot defaults, but can't quite figure out what is going on here. Any help would be appreciated! Wes ___ --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] @Home overwrites configs on startup
OK - makes sense. Thanks. My problem is that I need to remove some defaults - specifically the 82xx meetme extensions which conflict with my existing dial plan. Any ideas on this. -Original Message- From: Tom Vile [mailto:[EMAIL PROTECTED] Sent: Monday, January 02, 2006 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] @Home overwrites configs on startup you need to add your custom mods to the _custom.conf files and then they wont get over written On 1/2/06, Schochet, Wes [EMAIL PROTECTED] wrote: Hi Al- It appears that my config files are being overwritten on restart and/or reboot wit my [EMAIL PROTECTED] distribution. Anyone familiar with this behavior? I'd like to be able to set some of the reboot defaults, but can't quite figure out what is going on here. Any help would be appreciated! Wes ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony - AAH support www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Asterisk Dynamic DNS
Sounds like a DNS caching problem. Can you tell if the machine is actually going out to look up the address each time, or is it cached locally for some period of time? -Original Message- From: Branko Samardzic [mailto:[EMAIL PROTECTED] Sent: Sunday, December 11, 2005 12:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Dynamic DNS Hi everyone, I am running two Asterisk servers on two machines that have dynamic DNS due to ISP changing IP address daily. Both servers are registered on DynDns.org and IP update scripts work fine on both machines. However, if one machine changes IP address, other one (that has trunk pointing to machine that changed address) starts displaying that trunk host is not reachable. O.k. I thought, it is DNS propagation problem, but it is NOT! Even one hour after IP change, machine A still points to old IP address and says that it is not reachable. Is there any solution? Regards, Branko ___ --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] MeetMe questions
Hi- I have seen several different explanations of how MeetMe is supposed to function. I am having a tough time figuring out which is correct. If I put the room number in the extensions.conf file, I never get prompted for a PIN. When I leave it out of the extensions.conf file, I get prompted for a room number and a PIN. What I want, is to have a room number based on the DID extension that asks the user to enter his/her PIN. I can't make that happen. Here is my current files: extensions.conf: [ext-meetme] exten = 5570,1,Answer exten = 5570,2,wait(1) exten = 5570,3,MeetMe(|M) Meetme.conf: conf = 100,2321 conf = 101,2331 conf = 102,2231 1. How can I get 5570 always go to room 100 and just prompt the caller for a pin? 2. Ideally, I'd like to have a leader passcode and a participant passcode where the participants can't talk to each other until the leader joins. Any way to do that? Thanks, Wes ___ --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] Nortel Meridian Option81C to TE405P
Yes, I have other PRIs. problem is that there are 100 different fields to fill in on the M1, but only 20 on the zap/asterisk side! I was able to get this going.Afewpoints: 1. I am using the Sangoma Card - works great. 2. I have the M1 set for USR and the Asterisk set for NET 3. The D-Channel messages on the Asterisk side are generated by the Asterisk application (in hindsight - duh!). The d-channel won't come up unless the application is running. I had the clocking on the M1 set for "External" on the ADAN / D-Channel. The clocking on the Sangoma has two choices: "Master" and "Normal". I picked "Master" because I figured the M1 would be the "Slave" from a clocking standpoint. This generated a ton of errors and took me a long time to figure out. (Why the hell can't Nortel give you an inline description of an error message?After all of these years, you still get DTA0021 as an error message!) I changed the clock on the Sangoma to Normal and the span came right up. I may have also had a physical layer problem - maybe a bad DB15 to RJ48 converter. Now I'm in dial plan and MeetMe hell - but I'll get by that too! From: Joe Pukepail [mailto:[EMAIL PROTECTED] Sent: Thursday, December 08, 2005 9:44 PMTo: Schochet, WesSubject: Re: [Asterisk-Users] Nortel Meridian Option81C to TE405P yup, is this the only PRI you have coming into your nortel? we already had 2 other PRI's coming in so pretty much just copied the config settings from one of the other PRIs. What does it show if you stat the d-channel. (I think it is in load 96: stat dch d channel number). Everything else setup on the nortel? Have a clock source setup and everything? On 12/8/05, Schochet, Wes [EMAIL PROTECTED] wrote: Joe, are you running PRI to your Opt 11? I have a 61 and I can;t get my d-channel to come up to save my life! From: Joe Pukepail [mailto:[EMAIL PROTECTED]] Sent: Thursday, December 08, 2005 9:21 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Nortel Meridian Option81C to TE405P You can't use a ethernet crossover cable, make sure you are using a T1 crossover cable. (you will definately need to use a T1 crossover cable). I'm running a Nortel Option 11 and Asterisk connected in this manner. On 12/8/05, Steve Totaro [EMAIL PROTECTED] wrote: He said that he is using a crossover but for some reason I think thecrossover may be the problem.Try making a new one.Cross pin one with four and two with five.Also try a straight through cable.Your configs look fine on the asterisk side although I am not real cluefullon the Meridian.One question, was the Meridian ever hooked up to the PSTN? Thanks,Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX tomy Asterisk 1.20 server. We are using the TE405P card with onecrossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If wecannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks forany help in advance. zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 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
RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel
The other thing I'll say about my PBX is that there is no comparison between my Nortel i2004 and any SIP phone I've seen. Yes, the cost is slightly more, but for an instrument that I interact with constantly - there is no SIP device to compare. I know there will be eventually, but not now! -Original Message- From: O'Connor, Jonathan [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel Not sure I completely agree with all of these. Looking at it objectively, Asterisk has many benefits over traditional PBX systems, yet you should be aware of some of the limitations. Benefits: 1. Open source / low-cost of ownership / operates on cheap PC hardware. You get voicemail, IVR, hunt-groups etc. without additional fees. Last I checked those are all expensive add-ons in the Nortel world. There aren't expensive licenses per user/handset either. You get what you pay for, yes it operates on cheap hardware, but if you go that route you risk loss whatever system you run. 2. Flexibility - you can configure Asterisk to handle calls to a microscopic degree of precision. This is just not possible with traditional PBX systems which are inherently proprietary. Asterisk also makes it easier to present data to callers from CRM, Billing, Order Tracking systems etc. using text-to-speech, automated-speech recognition and/or DTMF recognition. I would have to agree mostly... The Definity ECS we have also has a level of detail and ability that is close to (and in some areas exceeds) Asterisk. Of course that's a 24000 handset capable system so I would hope it does :) 3. Flexibility again - It really is much more flexible than anything else!! If you consider cost yes, otherwise you have to take a strong look at some of the VoIP offerings out there. I don't want to sound like a huge Avaya fan, but their newer IP stuff is being designed from a whole new perspective. 4. Supports multiple VoIP protocols - SIP, IAX, H323, (and skinny to a degree) and supports connection of a broad spectrum of third party handsets - e.g. Cisco, Siemens, Sipura, etc. IAX is a proprietary protocol for Asterisk but it has some benefits over SIP (supposedly - my experience has been a little different) and perhaps more importantly is gaining popularity among VoIP service providers. This I love about it. I use Atcom AT320 phones here for home users with cable/DSL and only have to have one firewall port open for them, its beautiful in its simplicity. Internally we use Polycoms running SIP and Ciscos Plus a few ATAs and softphones running whatever the user prefers! 1. Digium PSTN interface boards are not as cheap as they could be and haven't been around long enough for us to have meaningful data on how reliable they are. I agree they havent been around that long, however I have never spent more then $600 on a single port T1 card and that's both cheaper then the ones for my traditional PBX and other manifacturers I have seen. They have to make a profit, and I cant see that sort of card with this small a market compared to other devices being able to come down much more... 2. Complexity. Asterisk is powerful but it is complicated - which is it You will need to spend a few weeks solidly learning about Asterisk and playing with it in a test environment before even thinking about trying to install it in a production environment. Clearly your time has a cost to your employer - thus this may be perceived as problem with Asterisk. You can of course buy in the services of an Asterisk consultant to help set things up - but ideally you want to have someone on site with some degree of knowledge about Asterisk's capabilities. If your business has basic telephony requirements, doesn't need fancy features and wants to minimize the need for on-site technical expertise to support Asterisk, then a Mitel/Nortel solution MIGHT make sense. IMHO - the present level of complexity/flexibility is the biggest strength and weakness to Asterisk. Agree 100%, however its not alone here I have an Avaya Definity, a Nortel and a Vodavi switch in this company to run... In the end the Avaya is slightly easier to manage then Asterisk but not much, and both are FAR easier then the other two. That said, Asterisk is the glue that bonds them, in that each one is connected to an Asterisk server with a T1 card and we have 4 digit dialing throughout our enterprise because of it, over IAX trunks. 3. Asterisk is a work in progress. Yes it's pretty stables and yes it's being used in very large production systems from what one hears on this list. However it's a moving target with new releases appearing frequently. On a positive note that's great if you want new features and bug fixes - but it can also be a pain if you want a nice stable,
RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P
Joe, are you running PRI to your Opt 11? I have a 61 and I can;t get my d-channel to come up to save my life! From: Joe Pukepail [mailto:[EMAIL PROTECTED] Sent: Thursday, December 08, 2005 9:21 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Nortel Meridian Option81C to TE405P You can't use a ethernet crossover cable, make sure you are using a T1 crossover cable. (you will definately need to use a T1 crossover cable). I'm running a Nortel Option 11 and Asterisk connected in this manner. On 12/8/05, Steve Totaro [EMAIL PROTECTED] wrote: He said that he is using a crossover but for some reason I think thecrossover may be the problem.Try making a new one.Cross pin one with four and two with five.Also try a straight through cable.Yourconfigs look fine on the asterisk side although I am not real cluefullon the Meridian.One question, was the Meridian ever hooked up to the PSTN? Thanks,Steve This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX tomy Asterisk 1.20 server. We are using the TE405P card with onecrossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If wecannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks forany help in advance. zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 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 ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT 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 8901 TCPP NO PII NO TARG 01 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 DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 1 73 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DTN WTA LPR APN THFD HKD P10 VNL TKID DATE 5 DEC 2005 Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --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 listTo UNSUBSCRIBE
RE: [Asterisk-Users] Nortel Meridian Option81C to TE405P
Can you do a ISDN message trace in LD 96 on the M1 when you try to bring up the D-Channel? LD 96 enl msgo 10 enl msgi 10 Make sure you later do a dis msgi 10 dis msgo 10 To shut it off. You should see good info there. -Original Message- From: Anthony Rodgers [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 07, 2005 10:41 AM To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nortel Meridian Option81C to TE405P This might be an obvious question, but should you be using a crossover cable? Information on setting up Nortel to TDM card links can be found at: http://www.pham.org/asterisk/asterisk-meridian-a1.pdf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Dec 6, 2005, at 2:59 PM, Anish Basu wrote: Hi, I am having problems connecting a Nortel Meridian Option 81C PBX to my Asterisk 1.20 server. We are using the TE405P card with one crossover PRI T1 cable connecting the two systems. The lights on the back of the TE405P are green and zttool shows that the span is okay. Calls cannot be made and 'pri show span 1' shows the d-channel as down. If anyone has any experience with this, suggestions and tips are greatly appreciatd. If we cannot get this resolved within the next few days, we are willing to pay consulting fees for help. The config files are as listed below. Thanks for any help in advance. zaptel.conf --- loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] language=en switchtype=5ess context=from-pbx signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-23 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both musiconhold=default Nortel configuration: b-channel,d-channel, and route data block --- REQ prt TYPE adan dch 10 ADAN DCH 10 CTYP MSDL GRP 3 DNUM 2 PORT 0 DES VresaBridge USR PRI DCHL 101 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 ROUT 1 TYPE RDB CUST 00 ROUT 1 DES VERSA TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS EXT 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 8901 TCPP NO PII NO TARG 01 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 DES VERSA TN 101 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 0 RTMB 1 73 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS UNR DTN WTA LPR APN THFD HKD P10 VNL TKID DATE 5 DEC 2005 Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --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
