Re: [Asterisk-Users] SIP vs SCCP vs XML
On Mon, 25 Aug 2003 18:45:22 -0400 Ray Burkholder [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* No, this is not the case currently with any of the Cisco SIP software loads that I am aware of. If you find this to be incorrect, please let the list know. Cisco has not deployed much of the featureset in their SCCP phones (such as paging/intercom) into the SIP phones due to lack of standards/interest/political capital. JT Ok, after further research in the 7960 administrators guide for SIP 5.1 (current is 5.3 and probably not changed much), they do state that support is not provided for CiscoIPPhoneExecute in the current SIP load, which is needed to make streaming channel 1 work. Bummer. So, in looking around at HotDispatch.com, I see a number of companies charging outrageous dollars for their own SCCP versions of a softphone. Also, a while back, for $1000, a person could join Cisco's developer program and gain access to SCCP docs. Perhaps an Asterisk group member has the funds available to attempt joining? Then we could finish up on some of the aborted attempts at SCCP integration, if the license agreement allows this sort of development. Perhaps, through a little creativity, it might be possible to use a SCCP 796x phone and not worry about SCCP. With XML, screens could be programmed to send responses back to *. Then * could drive streaming channel 1 directly and simulate the phone call. So, on a SCCP phone, you don't use SCCP, nor SIP. You use XML. Would that work? Hopefully soft button presses don't interfere with the streaming media. Oh, and if it does work, then you can use multicasting to intercom a number of phones simultaneously. The thing I miss on SIP phones that was available on the Callmanager version of 796x, is the ability to go off hook, dial some numbers, and callmanager automatically dials the call. The SIP version requires you to go off hook, dial the digits, then press dial. Any way around this for 4, 7, 10 or 11 digit dialling? Good question.. Does * support overlap dialing with SIP? I have a feeling it does, I do vaguely remember getting an Address Incomplete response when not dialing enough digits. I guess all you have to do is set your cisco phone for overlap dialing. Hopefully there is an option for it in is config. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP vs SCCP vs XML
On Mon, 2003-08-25 at 18:40, Adam Roach wrote: As a side note, I'll point out that the Pingtel phones let you provision client-side digitmaps. Based on asterisk-like pattern matching, you get to say how long a digit string should be matched, and the phone will automatically dial when it matches (no need to hit send!). You can even make different patterns go different places, like: 972xxx : sip:[EMAIL PROTECTED] 214xxx : sip:[EMAIL PROTECTED] 489xxx : sip:[EMAIL PROTECTED] 1xx : sip:[EMAIL PROTECTED] (To clarify: Dallas has three local area codes and 10 digit local dialing) The Cisco ATA-186 also has a Dial Plan config option. http://1.2.3.4/dev/ to access the ATA's config screen. HOWEVER, I've never been able to figure out the damn thing. Even with reading the docs. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP vs SCCP vs XML
As to prior comments about SCCP documentation: if you'd like to help contribute to the SCCP channel project, it seems far from 'aborted' at the moment. Check out http://sourceforge.net/projects/sccp/ and download the channel. Compile, test, send bugs, submit code. The web site indicates that This Project Has Not Released Any Files. Am I not seeing something? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP vs SCCP vs XML
Use the dialplan.xml file on the SIP phone. It will let you set how many digits to accept before beginning the dialing process, (you don't need to hit the DIAL button) Lee Goodman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Roach Sent: Monday, August 25, 2003 7:41 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] SIP vs SCCP vs XML I'll start by mentioning that the newer Cisco SIP dumps let you hit # instead of Dial when you're done dialing, which I find to be much more intuitive than the Dial softbutton. Good question.. Does * support overlap dialing with SIP? I have a feeling it does, I do vaguely remember getting an Address Incomplete response when not dialing enough digits. I guess all you have to do is set your Cisco phone for overlap dialing. Hopefully there is an option for it in is config. Even if Asterisk does the overlap stuff defined in RFC 3578, I seriously doubt you'll see the Cisco phones (or any hardware phones, for that matter) doing it. The overlap stuff is really designed for gateways from the PSTN, not end terminals. That said, there's nothing that would *prevent* implementing it in end devices. I note that it would cause an awful lot of signaling traffic if you did so, though. As a side note, I'll point out that the Pingtel phones let you provision client-side digitmaps. Based on asterisk-like pattern matching, you get to say how long a digit string should be matched, and the phone will automatically dial when it matches (no need to hit send!). You can even make different patterns go different places, like: 972xxx : sip:[EMAIL PROTECTED] 214xxx : sip:[EMAIL PROTECTED] 489xxx : sip:[EMAIL PROTECTED] 1xx : sip:[EMAIL PROTECTED] (To clarify: Dallas has three local area codes and 10 digit local dialing) /a ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intercom with Cisco SIP 796x phones?
What about the option of auto-answer that is available on Cisco 7960 version 5.x? Could you not setup a second line instance on the phone and set that line instance to auto answer? Lee Goodman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Laur Sent: Monday, August 25, 2003 11:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intercom with Cisco SIP 796x phones? BTXML support for client applications is necessary to achieve this. The SIP images state that they support BTXML; however, they only use it for their internal screens and internal navigation. CMXML is the only language supported for client applications with the SIP loads currently. A little bird at Cisco told me that a future version of the SIP loads will support BTXML client applications. (and a less reliable bird told me that the next version of call mangler will be SIP-based) This will support all of the good stuff I really want to be able to do with this phone. For instance, we could forward a URL with the call to a BTXML app that causes the phone to display extended information about the caller, or the voicemail Services button could show the users a menu of their voicemails and choosing one would play the message directly over the speaker. Also, intercom would be supported. All this is documented in the BTXML guide and ready to go whenever they open it up.. I suppose it might also be possible to find an exploit in the current firmware that causes the phone to execute some BTXML from a remote location.. we might be able to get in that way, too :) John -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Todd Sent: Monday, August 25, 2003 7:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intercom with Cisco SIP 796x phones? If you find a way to make the phone request that second audio stream without user intervention, I'm all ears. :-) JT At 5:15 PM -0400 8/25/03, Ray Burkholder wrote: From: Ray Burkholder [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Intercom with Cisco SIP 796x phones? Reply-To: [EMAIL PROTECTED] Date: Mon, 25 Aug 2003 17:15:01 -0400 I read about this intercom stuff on page 62 63 of the book Developing Cisco IP Phone Services isbn 1-58705-060-9. Primary calls take place on streaming channel 0. When streaming channel 0 is not in use, streaming channel 1 can be used for asynchronously streaming (in and out) stuff like voicemail, email, and, yep the one we want, intercom. Page 87-88 of the book talks about CiscoIPPhoneExecute to push the commands to the phone. On the last two pages of an addendum found at http://services.dogma.net/errata.doc, more details are provided for connecting to streaming port 1. http://cisco.evolvis.net/ivision/pdfs/Jukka_Nurmi_iVision2003.pdf provide some background on Cisco's IP Phone Services. Title is foreign language, but text is English. http://www.loligo.com/asterisk/Cisco/79xx/2003-06-20.from-ftpeng.cisco. c om/CMXML_App_Guide.pdf provides additional program details. From what I see, basic functionality should be a piece of cake. The fun will be in the Asterisk call control integration. All this hinges on the fact that all the XML functionality built into the CallManager phone load is also built into the recent SIP phone loads. I guess trial and error is the best way to find this out. Good Luck! Ray Burkholder One Unified 519 570 0689 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: August 25, 2003 15:11 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Is Asterisk ready for real use? Oh really?!? Can you give us more information... On Mon, 2003-08-25 at 12:30, Ray Burkholder wrote: The Cisco SIP phones have a second voice channel available for a paging type of implementation. Now the problem is simply of finding someone and some time to see if it can be made to work with Asterisk. Ray Burkholder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] Intercom with Cisco SIP 796x phones?
At 9:59 PM -0500 8/25/03, John Laur wrote: BTXML support for client applications is necessary to achieve this. The SIP images state that they support BTXML; however, they only use it for their internal screens and internal navigation. CMXML is the only language supported for client applications with the SIP loads currently. A little bird at Cisco told me that a future version of the SIP loads will support BTXML client applications. (and a less reliable bird told me that the next version of call mangler will be SIP-based) Yes, the same bird has told me similar stories. However, Cisco is a large organization, and there are lots of birds there, some of them singing songs that will never be heard by the rest of the world. Cisco's phones are great (IMHO) and are already full of nice features. However, a few more features in the SIP load would make them much more market-able. Much of this delay has to do with internal Cisco politics (despite what they might admit or realize) which is somewhat disheartening. This will support all of the good stuff I really want to be able to do with this phone. For instance, we could forward a URL with the call to a BTXML app that causes the phone to display extended information about the caller, or the voicemail Services button could show the users a menu of their voicemails and choosing one would play the message directly over the speaker. Also, intercom would be supported. All this is documented in the BTXML guide and ready to go whenever they open it up.. That would be really useful. I suppose it might also be possible to find an exploit in the current firmware that causes the phone to execute some BTXML from a remote location.. we might be able to get in that way, too :) My idea is to accept SIP NOTIFY messages, password-protected, that do certain things. It's already part of the SIP spec. An XML URL, as an example, would be perfect payload. As would an INVITE to an auto-answer meta-extension (configurable from the phone) for intercom or paging. I've made these suggestions through a feature request system within Cisco, and through what hopefully will be a clueful channel. We'll see. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gnophone connection
Title: Message hello everybody well...while trying to make gnophone to gnophone call using IAXtel's PBX server..i am not being able to establish a connetion possibly whats happening is, we are not being able to transmit our message packets properly and result is its not being able to establish connection n is resending frmaes again and gain here is message we are getting Tx-Frame Retry[000] Type IAX Subclass LAGRQ repeatedly pls see to it if somebody can help me out also interested in knowing something about configuring these GNOPHONE with our own ASTERISK server..which we have already downloaded and installed...we are still clueless about what all configuration files need to be modified and what else...pls help us if some body can thanks regards Ashish AGrawal Engineer HFCL(RD) Gurgaon ,India
Re: [Asterisk-Users] Chan_h323 and a Cisco Gateway
Well depends.. what kind of problem are you having? http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml http://www.cisco.com/en/US/tech/tk652/tk701/technologies_problem_troubleshooting09186a00800c5e33.shtml Check those... I suspect one of those has nailed ya. If you have PRI and you try to terminate outbound via chan_h323 you must have bearer-cap speech on your voice-ports. Because chan_h323 isn't sending the appropriate bearer cap in the H.225 SETUP message. Hours of beating head on desk and searching... Hope this helps. Thanks, Brian On Tue, 26 Aug 2003, Steven Thomas wrote: Hi, Can anyone tell me what should be included in h323.conf to get asterisk to talk to a Cisco 2600 gateway? Any statement examples for extensions.conf would also be appreciated. Thanks. Will chan_h323 use the Cisco as a gateway anyway? Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alias limitation in asterisk-oh323.0.5.5
Hi, Is there limitation of alias creation in the file oh323.conf with asterisk-oh323.0.5.5 ? In my config file I have 32 alias and when i call someone in h323 Asterisk do a segmentation fault. I try to delete alias and have 9, everything is OK. It is normal ? Regards Rattana
[Asterisk-Users] TDM10M Siemens Euroset 2015
Hi all,
I have installed a TDM400 with one active FXS port (TDM10B) an connected
it to a Siemens Euroset 2015 analogue phone.
I have installed some smom IP phones to the network as well and
configured them as usual (sip.conf). For configuring the TDM10B I have
used FXO signalling in /etc/zaptel.conf and in
/etc/asterisk/zapata.conf. I definded the TDM channel and the Snom
phones to the same context (local) and created a dialplan for the local
dialing in extensions.conf.
I can dial from every Snom IP phone the any other Snom IP Phone. I can
dial from any Snom IP phone to the Siemens Euroset but I cannot dial
from the Siemens Euroset to any IP phone. While pressing the first
button of the Euroset's keyboard a busy tone is heared and Asterisk
console shows a hangup signal. Do you have any idea to solve this
problem ?
here are my configuration files:
/etc/zaptel.conf
---
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
#
fxoks=1
loadzone = us
#loadzone=fr
#loadzone=fr
#loadzone=de
#loadzone=uk
defaultzone=us
-
/etc/asterisk/zapata.conf
-
;
; Zapata telephony interface
;
; Configuration file
[channels]
;
; Default language
;
language=en
;
; Default context
;
callerid=PBX Operator 2000
signalling=fxo_ks
relaxdtmf=yes
channel=1
context=local
-
/etc/asterisk/sip.conf
---
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
disallow=all
allow=ulaw
allow=alaw
;tos=lowdelay
;tos=184
;maxexpirey=3600; Max length of incoming registration we
allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY
;
;register = [EMAIL PROTECTED] ; Register with a SIP provider
;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider
as 1234 here.
;
[snom1]
type=friend
host=129.26.10.121
dtmfmode=rfc2833
mailbox=2101
reinvite=no
context=local
callerid=Studenten 1 2101
[snom2]
type=friend
host=129.26.10.122
dtmfmode=rfc2833
mailbox=2102
reinvite=no
context=local
callerid=Studenten 2 2102
[snom3]
type=friend
host=129.26.10.108
dtmfmode=rfc2833
mailbox=2103
reinvite=no
context=local
callerid=Olaf 2103
[snom4]
type=friend
host=129.26.10.109
dtmfmode=rfc2833
mailbox=2104
reinvite=no
context=local
callerid=Samba 2104
[zaurus1]
type=friend
host=dynamic
dtmfmode=rfc2833
mailbox=2105
context=local
callerid=Zaurus1 2105
---
/etc/asterisk/extensions.conf
-
;
; Static extension configuration files, used by
; the pbx_config module.
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; The Globals category contains global variables that can be
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=/dev/dsp
[local]
; Operator/ Console
exten = 2000,1,Dial,Zap/1|30
exten = 2000,2,Voicemail,u2000
exten = 2000,102,Voicemail,b2000
; SIP Phones
exten = 2101,1,Dial,SIP/snom1|30
exten = 2001,2,Voicemail,u2001
exten = 2001,102,Voicemail,b2001
exten = 2102,1,Dial,SIP/snom2|30
exten = 2002,2,Voicemail,u2002
exten = 2002,102,Voicemail,b2002
exten = 2103,1,Dial,SIP/snom3|30
exten = 2003,2,Voicemail,u2003
exten = 2003,102,Voicemail,b2003
exten = 2104,1,Dial,SIP/snom4|30
exten = 2004,2,Voicemail,u2004
exten = 2004,102,Voicemail,b2004
-
Thank you for your help
Olaf
--
Dipl. Ing. Olaf Menzel - System Engineer
FOKUS - Fraunhofer Institute for Open Communication Systems:
- Competence Center for Advanced Satellite Communication
Schloss Birlinghoven, 53754 Sankt Augustin, Germany
Phone: +49-2241-14-3494 Mobile: +49-175-2616161 Fax: +49-2241-14-43494
email: [EMAIL PROTECTED] Internet:http://www.fokus.fhg.de/satcom
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free World Dialup.
G'day Asteriskers, I've been looking through google to see how to get asterisk and free world dialup working together - I'm not having much luck. When I dial a FWD number (via my SIP FWD definition), I hear the first 1/4 second (or so) of sound, then nothing (but I still stay connected)... Does this make sense to anybody? I know my FWD connection is working, because when I call myself via iaxtel, using the 170099 (FWD) number, I hear my voicemail. Appreciate any tips... ...deon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Decent DECT cordless compatible with Asterisk/ATA?
Hi, Can anyone recommend me a decent DECT cordless phone with the following minimal features: - Caller ID with graphical display for both Name and Number (compatible with */ATA) - min.14 days autonomy (without calls) - backlight for display and keys - good quality keyboard (I have two models of Philips Onis phones now and the quality of the rubber keyboard is unacceptable) - no answering machine included (use Asterisk Voicemail System) - graphical Message Waiting indicator (compatible with */ATA), in order not to be forced to use the different dialtone type of signaling - keep a list of the lost calls (min. 10) with the callerid information Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * server based Phonebook
Hi, There is any phonebook type application/script available, server based? I want to keep the list of names/phone numbers centralized on the server (in a database) for all the phones connected to Asterisk. The caller ID extracted from the list must be passed then to the phones when a call is received through the X100P card from the PSTN line. Thanks, Da ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Secondary gatekeeper support by asterisk h323drivers
Michael Ulitskiy wrote: Hi, I'm wondering if there are any plans on adding secondary gatekeeper support to asterisk h323 channel drivers. Nice to have something like this. I 'll add it to the TODO features of asterisk-oh323. Also I've noticed that chan_h323 is crashing asterisk at startup if primary gatekeeper is not available. Wouldn't it be a more correct behavior if it doesn't crashing but continue registration attempts in the background? Didn't test it with chan_oh323. There is no such problem with chan_oh323. Thank you. Michael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bug report: whitespaces in uris
FYI: Asterisk puts URIs in messages which violates the SIP spec and can't be accepted by URI parsers: username includes a whitespace. See for example the From header field. Attached is example of an incorrect message and related parts of RFC3261 specification. (Who doesn't want to dig into parser details may want to realize that whitespaces are used as uri delimitors in first request line and can't thus be a uri part.) I would recommend that the stack generally validates URIs for such glitches and uses other word for no callId. anonymous is in frequent use by other software. -jiri OPTIONS sip:195.37.77.101 SIP/2.0 Via: SIP/2.0/UDP 24.172.18.166:5060;branch=z9hG4bK03be4cf3 From: No CallID sip:No [EMAIL PROTECTED];tag=as2746f4f3 To: sip:195.37.77.101 Contact: sip:No [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 3261: From-name_addr|addr_spec addr_spec-SIP_URI SIP_URI-userinfo user_info-user user-1*( unreserved / escaped / user-unreserved user-unreserved = / = / + / $ / , / ; / ? / / unreserved = alphanum / mark mark= - / _ / . / ! / ~ / * / ' / ( / ) -- Jiri Kuthan http://iptel.org/~jiri/ iptel.org -- creaters of the fastest SIP server ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * server based Phonebook
we run such at iptel.org -- just get an account there and see how it works. Limitation: it takes phones with REFER support. We tested against Cisco 7960, Mitel, Pingtel (and maybe some more whose name I no longer remember). Microsfot Messenger is known not to work, as it is REFER-ignorant. -jiri On Tue, 26 Aug 2003, Dan wrote: Hi, There is any phonebook type application/script available, server based? I want to keep the list of names/phone numbers centralized on the server (in a database) for all the phones connected to Asterisk. The caller ID extracted from the list must be passed then to the phones when a call is received through the X100P card from the PSTN line. Thanks, Da ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * server based Phonebook
Hi Jiri, Why is this phone dependant? It is jut about to query a database for a specific phone number and then extract the Name and set the variable CALLERID variable accordingly. How can I see how it works? I can see just the result... Thanks, Dan - Original Message - From: Jiri Kuthan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 26, 2003 3:06 PM Subject: Re: [Asterisk-Users] * server based Phonebook we run such at iptel.org -- just get an account there and see how it works. Limitation: it takes phones with REFER support. We tested against Cisco 7960, Mitel, Pingtel (and maybe some more whose name I no longer remember). Microsfot Messenger is known not to work, as it is REFER-ignorant. -jiri On Tue, 26 Aug 2003, Dan wrote: Hi, There is any phonebook type application/script available, server based? I want to keep the list of names/phone numbers centralized on the server (in a database) for all the phones connected to Asterisk. The caller ID extracted from the list must be passed then to the phones when a call is received through the X100P card from the PSTN line. Thanks, Da ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * server based Phonebook
I think your are looking for something like this tool: LookupCIDName it uses the internal database of asterisk and allows you to hookup the CallerIDName based on a list in the database. show application LookupCIDName in * for more info. Greetings, Tj - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 26, 2003 1:22 PM Subject: [Asterisk-Users] * server based Phonebook Hi, There is any phonebook type application/script available, server based? I want to keep the list of names/phone numbers centralized on the server (in a database) for all the phones connected to Asterisk. The caller ID extracted from the list must be passed then to the phones when a call is received through the X100P card from the PSTN line. Thanks, Da ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: T100P/ TSU 600 installation problem
I am using a crossover cable. My channel definitions are: fxoks=1-22 fxsks=23-24 in zaptel.conf --- Alex Lopez [EMAIL PROTECTED] wrote: What cable are you using, The SU600 to Digium cards need a crossover cable. 1 to 4 2 to 5 4 to 1 5 to 2 That would stop it from not working, also make sure that you have a span definition on the zaptel.con file. Message: 7 Date: Mon, 25 Aug 2003 12:52:12 -0700 (PDT) From: jerk face [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Each port is set to the proper signalling type (FXO, FXS). I can't find any other options for the individual ports. As for the timing and configuration of NI, I have tried NI: Timing Mode as both DTE and NI (my only choices) Where else should I be checking? (Before this morning, I hadn't even seen a channel bank before, so I'm a little lost at the moment). Thanks for your time. --- Wade Weppler [EMAIL PROTECTED] wrote: Have you configured the TSU600 properly? You have to allocate each FXO/FXS channel to a timeslot before it will work. This is not automatically done (like the Adtran Total Access series). Mind you, you should still have a sync light on the T1 card... -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of jerk face Sent: Monday, August 25, 2003 3:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem My zapata.conf is located in /etc/asterisk and my zaptel.conf is located in the /etc directory. --- Adams, Gavin [EMAIL PROTECTED] wrote: -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel = 1-22 ... signalling=fxs_ks ... channel = 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In /proc/zaptel/1 all of the configured channels are listed. Crazy question, is zaptel.conf is /etc or /etc/asterisk? If the latter, try: ztcfg -c /etc/asterisk/zaptel.conf -vvv --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: T100P/ TSU 600 installation problem
Oops ... I found out my problem span= --- jerk face [EMAIL PROTECTED] wrote: I am using a crossover cable. My channel definitions are: fxoks=1-22 fxsks=23-24 in zaptel.conf --- Alex Lopez [EMAIL PROTECTED] wrote: What cable are you using, The SU600 to Digium cards need a crossover cable. 1 to 4 2 to 5 4 to 1 5 to 2 That would stop it from not working, also make sure that you have a span definition on the zaptel.con file. Message: 7 Date: Mon, 25 Aug 2003 12:52:12 -0700 (PDT) From: jerk face [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Each port is set to the proper signalling type (FXO, FXS). I can't find any other options for the individual ports. As for the timing and configuration of NI, I have tried NI: Timing Mode as both DTE and NI (my only choices) Where else should I be checking? (Before this morning, I hadn't even seen a channel bank before, so I'm a little lost at the moment). Thanks for your time. --- Wade Weppler [EMAIL PROTECTED] wrote: Have you configured the TSU600 properly? You have to allocate each FXO/FXS channel to a timeslot before it will work. This is not automatically done (like the Adtran Total Access series). Mind you, you should still have a sync light on the T1 card... -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of jerk face Sent: Monday, August 25, 2003 3:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem My zapata.conf is located in /etc/asterisk and my zaptel.conf is located in the /etc directory. --- Adams, Gavin [EMAIL PROTECTED] wrote: -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel = 1-22 ... signalling=fxs_ks ... channel = 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In /proc/zaptel/1 all of the configured channels are listed. Crazy question, is zaptel.conf is /etc or /etc/asterisk? If the latter, try: ztcfg -c /etc/asterisk/zaptel.conf -vvv --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * server based Phonebook
Thanks, This is what I looking for. Best regards, Dan - Original Message - From: Tjardick van der Kraan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 26, 2003 3:56 PM Subject: Re: [Asterisk-Users] * server based Phonebook I think your are looking for something like this tool: LookupCIDName it uses the internal database of asterisk and allows you to hookup the CallerIDName based on a list in the database. show application LookupCIDName in * for more info. Greetings, Tj - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 26, 2003 1:22 PM Subject: [Asterisk-Users] * server based Phonebook Hi, There is any phonebook type application/script available, server based? I want to keep the list of names/phone numbers centralized on the server (in a database) for all the phones connected to Asterisk. The caller ID extracted from the list must be passed then to the phones when a call is received through the X100P card from the PSTN line. Thanks, Da ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unsubscribe
unsubscribe - Original Message - From: jerk face [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 26, 2003 9:57 AM Subject: Re: [Asterisk-Users] RE: T100P/ TSU 600 installation problem Oops ... I found out my problem span= --- jerk face [EMAIL PROTECTED] wrote: I am using a crossover cable. My channel definitions are: fxoks=1-22 fxsks=23-24 in zaptel.conf --- Alex Lopez [EMAIL PROTECTED] wrote: What cable are you using, The SU600 to Digium cards need a crossover cable. 1 to 4 2 to 5 4 to 1 5 to 2 That would stop it from not working, also make sure that you have a span definition on the zaptel.con file. Message: 7 Date: Mon, 25 Aug 2003 12:52:12 -0700 (PDT) From: jerk face [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Each port is set to the proper signalling type (FXO, FXS). I can't find any other options for the individual ports. As for the timing and configuration of NI, I have tried NI: Timing Mode as both DTE and NI (my only choices) Where else should I be checking? (Before this morning, I hadn't even seen a channel bank before, so I'm a little lost at the moment). Thanks for your time. --- Wade Weppler [EMAIL PROTECTED] wrote: Have you configured the TSU600 properly? You have to allocate each FXO/FXS channel to a timeslot before it will work. This is not automatically done (like the Adtran Total Access series). Mind you, you should still have a sync light on the T1 card... -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of jerk face Sent: Monday, August 25, 2003 3:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem My zapata.conf is located in /etc/asterisk and my zaptel.conf is located in the /etc directory. --- Adams, Gavin [EMAIL PROTECTED] wrote: -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel = 1-22 ... signalling=fxs_ks ... channel = 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In /proc/zaptel/1 all of the configured channels are listed. Crazy question, is zaptel.conf is /etc or /etc/asterisk? If the latter, try: ztcfg -c /etc/asterisk/zaptel.conf -vvv --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Accountcode and cdr-csv
Hello, Why does not accountcode apeer in cdr-csv? Could anyone help? ,710,01332213334,striped,710,SIP/-0810ee00,Zap/1-1,Dial,Zap/g1/01332213334,2003-08-26 10:01:59,2003-08-26 10:02:08,2003-08-26 10:02:11,12,3,ANSWERED,DOCUMENTATION ===sip.conf== [ata] type=friend context=striped host=10.0.11.160 dtmfmode=inband accountcode=pd thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unsubscribe
Please read either the headers of this message, or the footer. DO NOT SEND ANOTHER UNSUBSCRIBE TO THE LIST. On Tue, 2003-08-26 at 08:39, FRANCISCO PEREZ-LANDAETA wrote: unsubscribe From: Adam Roach [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP vs SCCP vs XML Date: Mon, 25 Aug 2003 18:40:59 -0500 I'll start by mentioning that the newer Cisco SIP dumps let you hit # instead of Dial when you're done dialing, which I find to be much more intuitive than the Dial softbutton. Good question.. Does * support overlap dialing with SIP? I have a feeling it does, I do vaguely remember getting an Address Incomplete response when not dialing enough digits. I guess all you have to do is set your Cisco phone for overlap dialing. Hopefully there is an option for it in is config. Even if Asterisk does the overlap stuff defined in RFC 3578, I seriously doubt you'll see the Cisco phones (or any hardware phones, for that matter) doing it. The overlap stuff is really designed for gateways from the PSTN, not end terminals. That said, there's nothing that would *prevent* implementing it in end devices. I note that it would cause an awful lot of signaling traffic if you did so, though. As a side note, I'll point out that the Pingtel phones let you provision client-side digitmaps. Based on asterisk-like pattern matching, you get to say how long a digit string should be matched, and the phone will automatically dial when it matches (no need to hit send!). You can even make different patterns go different places, like: 972xxx : sip:[EMAIL PROTECTED] 214xxx : sip:[EMAIL PROTECTED] 489xxx : sip:[EMAIL PROTECTED] 1xx : sip:[EMAIL PROTECTED] (To clarify: Dallas has three local area codes and 10 digit local dialing) /a ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN 8: Get 6 months for $9.95/month. http://join.msn.com/?page=dept/dialup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem starting Asterisk after abnormal shutdown
I've seen this happen a few times and I think it's when the system that Asterisk is running on crashes due to a power failure (or for some other reason that causes a non-planned shutdown). While Linux comes up fine, Asterisk won't start because the drivers are loadingin the wrong order. fixed by: 1) sh /usr/src/fix-asterisk-modules.sh 2) sh /etc/init.d/asterisk start Is this a known problem? Is there an existing bug on this or should I open one up? Anyone else seen this problem? Thanks Lee Goodman
[Asterisk-Users] Asterisk internal database access
Hi, There is any simple way to have external access to the Asterisk database? I want to be able to edit a specific family only, which is a phonebook. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream firmware update DMTF Payload Type
marrandy wrote: On Monday 25 August 2003 03:37 pm, Ian Blenke wrote: What NTP issue? My 1.0.3.78 phones are NATting out just fine. Ian, I also had no NTP issues with NAT, but from what I remember when I perused the archives back in June/July, there were issues if you changed from the default NTP server. Perhap's that's what they mean. Elaborations welcomed. Regards...Martin The phones came pre-configured with sipphone.com's NTP server. I've been using our own NTP servers for testing with 1.0.3.78, behind a linux iptables NATting firewall, with no apparent issues. I've just verified with ethereal that the NTP packets do indeed hit our NTP servers, and the reply does set the time correctly on the phone's display. Count me as someone who has not experienced this problem. YMMV. -- - Ian C. Blenke [EMAIL PROTECTED] (This message bound by the following: http://www.nks.net/email_disclaimer.html) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP vs SCCP vs XML
Checkout the dialplan.xml file... Jared Smith On Mon, 2003-08-25 at 16:45, Ray Burkholder wrote: No, this is not the case currently with any of the Cisco SIP software loads that I am aware of. If you find this to be incorrect, please let the list know. Cisco has not deployed much of the featureset in their SCCP phones (such as paging/intercom) into the SIP phones due to lack of standards/interest/political capital. JT Ok, after further research in the 7960 administrators guide for SIP 5.1 (current is 5.3 and probably not changed much), they do state that support is not provided for CiscoIPPhoneExecute in the current SIP load, which is needed to make streaming channel 1 work. Bummer. So, in looking around at HotDispatch.com, I see a number of companies charging outrageous dollars for their own SCCP versions of a softphone. Also, a while back, for $1000, a person could join Cisco's developer program and gain access to SCCP docs. Perhaps an Asterisk group member has the funds available to attempt joining? Then we could finish up on some of the aborted attempts at SCCP integration, if the license agreement allows this sort of development. Perhaps, through a little creativity, it might be possible to use a SCCP 796x phone and not worry about SCCP. With XML, screens could be programmed to send responses back to *. Then * could drive streaming channel 1 directly and simulate the phone call. So, on a SCCP phone, you don't use SCCP, nor SIP. You use XML. Would that work? Hopefully soft button presses don't interfere with the streaming media. Oh, and if it does work, then you can use multicasting to intercom a number of phones simultaneously. The thing I miss on SIP phones that was available on the Callmanager version of 796x, is the ability to go off hook, dial some numbers, and callmanager automatically dials the call. The SIP version requires you to go off hook, dial the digits, then press dial. Any way around this for 4, 7, 10 or 11 digit dialling? Ray Burkholder 519 570 0689 x2002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More questions. Call Waiting and Threeway
I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking something (plus, I'm still not up to speed on the hardware/telco end of the setup - all of the work I'm doing is with software). Is there any way to control whether three-way and caller ID are enabled per-call or per-SIP-phone? What I'd like to do is, for example, be able to dial *70 from my SIP phone to turn off call waiting, or be able to enable three-way on a per-phone basis. I don't know what's on the other side of the Zap channels (i.e. PRI, CT-1, whatever), but if it makes a difference, I can find out. -- JustThe.net Internet Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex Problem
Hey all, I'm current experiencing a problem when attempting to make a speex call from X-Lite to Asterisk... When I attempt to call an extension with the demo configuration, I get no audio (only with speex), and after awhile my /var/log/asterisk/messages fills up with this: Aug 26 12:18:53 WARNING[122894]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Here is a brief cut from the messages log for the entire transaction.. Aug 26 12:22:08 DEBUG[57352]: File chan_sip.c, Line 3783 (check_user): Setting NAT on RTP to 0 Aug 26 12:22:08 DEBUG[57352]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 30335: Found Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 3783 (check_user): Setting NAT on RTP to 0 Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 4807 (handle_request): Check for res Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 952 (find_user): Call from user 'nettwerk' is 1 out of 0 Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 3249 (build_route): build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Aug 26 12:22:09 DEBUG[139278]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from UNKN to SPEEX Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 30336: Found Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space I *know* speex is supported by asterisk, so I must be doing something dumb.. lol Any help would be appreciated -san ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers
Great! Thanks, Michael. Jeremy, what do you think? Michael On Tuesday 26 August 2003 07:53 am, Michael Manousos wrote: Michael Ulitskiy wrote: Hi, I'm wondering if there are any plans on adding secondary gatekeeper support to asterisk h323 channel drivers. Nice to have something like this. I 'll add it to the TODO features of asterisk-oh323. Also I've noticed that chan_h323 is crashing asterisk at startup if primary gatekeeper is not available. Wouldn't it be a more correct behavior if it doesn't crashing but continue registration attempts in the background? Didn't test it with chan_oh323. There is no such problem with chan_oh323. Thank you. Michael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk internal database access
Hi, There is any simple way to have external access to the Asterisk database? I want to be able to edit a specific family only, which is a phonebook. Thanks, Dan A search thorough the archives of only 17 days ago found this: JT Subject: RE: [Asterisk-Users] UNIX command-line interaction with astdb From: Benjamin Miller [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Sat, 9 Aug 2003 12:54:08 -0500 As far as I understand you do not want to open this DB while Asterisk has it open. Even though it is in standard db1 format. A project _way_ down my list was to write some perl DBI commands that allowed access to the asterisk db via the manager interface. However, for now, I am looking at implementing most of my DB needs in an external postgres database. This allows for mulitple asterisk boxes to access it via an asterisk native app or an agi app, as well as external programs. If you really need to access the astdb, you might want to pursue doing it through the manager interface. Ben -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Friday, August 08, 2003 8:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] UNIX command-line interaction with astdb I'm wondering if there is any command-line interface available for working with values stored in astdb. Of course, I can run asterisk -rx database show or other commands like that, but I was hoping for a local command that would allow manipulation or output in some other form. Is astdb in a standard db format? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 0 out of voicemail to different secretaries
On Monday, August 25, 2003 4:55 PM, Brad Bergman [SMTP:[EMAIL PROTECTED] wrote: I certainly contemplated that very thing... but somehow it escaped implementation. Even as things are now, the PBX administrator can set something like this up by putting engineering, accounting, etc in different contexts, and setting different o extensions for them. This does not appear to be an alternative since a channel is assigned a context before the call is made, not after the number is dialed so the dialed number does not establish the context. So putting phones (or channels) in different contexts would only help if the context can be changed during a call. I was thinking of a couple of relevant features, one giving the individual mailbox user the ability to store a different extension (i.e., the target attendant) in the database that would override the o extension in the dialplan when a caller presses '0'. The other is to allow another number(s) to be stored in the DB so that a caller could press, say, 4, 5, or 6, and be transferred to whatever number the mailbox owner has stored there. Of course, all of this subject to whatever restrictions are imposed on the use of these features. I will look into this. I hope your schedule and motivation leads you to look into this further as it will be important for us. We will be converting fully to * as soon as we can satisfy internal users that * can handle our requirements. Please let me know if I can help with testing. Don Pobanz Cheers, Brad On Mon 25 Aug 2003 14:25, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=156 patients grass hopper! bkw On Mon, 25 Aug 2003, Don Pobanz wrote: Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pickup groups with SIP
I upgraded to the latest and greatest from CVS today, and now SIP pickup groups appear to be broken. Can anyone else tell me whether or not they're seeing the same problem. If anyone out there can verify the problem, I'll submit it as a bug. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100P: Ring/off-hook in strange state 6 on channel1
All of a sudden I am getting the following warning "Ring/off-hook in strange state 6 on channel1" from chan_zap.c and I cannot answer calls. I can place calls out without a problem though. Any ideas what can be the problem. I have checked zapata.conf and zaptel.conf and they both seem fine. Thanks in advance. Dan
RE: [Asterisk-Users] Dialed Number Identification in analog hunt group
On Tuesday, August 26, 2003 4:55 PM, Stephen R. Besch [SMTP:[EMAIL PROTECTED] wrote: Does anyone out there know if it is possible to discover the dialed number when a line in an analog hunt group rings? I can't get a straight answer from our IT folks. We have a 5ess switch delivering 4 analog lines which are in a simple hunt group servicing our lab. I would like to have a different call attendant based on which number is dialed so that I can route the calls to the appropriate group. I know that Asterisk can easily do this once I have the information to pass into the dial plan. The problem is getting the information. While I know that this is possible with T1, it is, unfortunately, a bit overkill for 4 lines. Anyone have any suggestions? If they are pots lines in a hunt group, you won't be able to. If they are analog DID trunks then the dialed number would be passed. My guess is you have pots lines and there is no way to find out the dialed number. Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers
On Tuesday 26 August 2003 05:26 pm, Jeremy McNamara wrote: H.323 is no longer part of my own network, so I no longer have a built in method to burn lots of time beating up a dying protocol and unfortunately, I have to dedicate myself to projects that directly generate revenue. I'll see what I can do, but I can make no promises. I understand. I hope it doesn't mean that chan_h323 is now unsupported. Sorry, Jeremy McNamara P.S. I could not duplicate any crash using my own (proprietary) gatekeeper. Plus, I cannot do a thing without debug information. This is output of asterisk startup with chan_h323 when gatekeeper is unavailable or reject registration: [EMAIL PROTECTED]:/etc/asterisk# asterisk -cd DEBUG[1024]: File config.c, Line 744 (__ast_load): No file to parse: /usr/local/asterisk/etc/asterisk/asterisk.conf Asterisk CVS-08/05/03-16:15:37, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] ... [chan_h323.so] = (The NuFone Network's Open H.323 Channel Driver) == Creating H.323 Endpoint == Setting default context to h323 == Adding alias pbxb to endpoint == Adding Prefix 1212555 to endpoint == H.323 listener started *** Error registering with gatekeeper 192.168.0.7. ERROR[1024]: File chan_h323.c, Line 1673 (load_module): Gatekeeper registration failed. WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_h323.so: load_module failed, returning -1 == PWLib proces going down. WARNING[1024]: File loader.c, Line 345 (load_modules): Loading module chan_h323.so failed! Segmentation fault Michael Ulitskiy wrote: Great! Thanks, Michael. Jeremy, what do you think? Michael On Tuesday 26 August 2003 07:53 am, Michael Manousos wrote: Michael Ulitskiy wrote: Hi, I'm wondering if there are any plans on adding secondary gatekeeper support to asterisk h323 channel drivers. Nice to have something like this. I 'll add it to the TODO features of asterisk-oh323. Also I've noticed that chan_h323 is crashing asterisk at startup if primary gatekeeper is not available. Wouldn't it be a more correct behavior if it doesn't crashing but continue registration attempts in the background? Didn't test it with chan_oh323. There is no such problem with chan_oh323. Thank you. Michael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Secondary gatekeeper support by asterisk h323drivers
H.323 is no longer part of my own network, so I no longer have a built in method to burn lots of time beating up a dying protocol and unfortunately, I have to dedicate myself to projects that directly generate revenue. I'll see what I can do, but I can make no promises. Sorry, Jeremy McNamara P.S. I could not duplicate any crash using my own (proprietary) gatekeeper. Plus, I cannot do a thing without debug information. Michael Ulitskiy wrote: Great! Thanks, Michael. Jeremy, what do you think? Michael On Tuesday 26 August 2003 07:53 am, Michael Manousos wrote: Michael Ulitskiy wrote: Hi, I'm wondering if there are any plans on adding secondary gatekeeper support to asterisk h323 channel drivers. Nice to have something like this. I 'll add it to the TODO features of asterisk-oh323. Also I've noticed that chan_h323 is crashing asterisk at startup if primary gatekeeper is not available. Wouldn't it be a more correct behavior if it doesn't crashing but continue registration attempts in the background? Didn't test it with chan_oh323. There is no such problem with chan_oh323. Thank you. Michael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware Requirement for Asterisk PBX
Hello All, I am newbie to Asterisk IP PBX Community. We are planning to use Asterisk IP PBX at our university campus. I would like to know what's the ideal hardware requirements for setting up Asterisk. At present we have 12000 analog telphone lines, so goal is to use IP telephony on campus in future. Thanks, Tarun ___ Art meets Army ; Swapna Weds Capt. Rajsekhar. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
