Re: [Asterisk-Users] Where are chan_capi bug reports and bugfixes sent?
chan_capi is not part of asterisk due to license issues (chan_capi is pure GPL, while to have code included in asterisk you have to sign a disclaimer to digium to let them use your code for commercial, closed source asterisk as well). (if I'm wrong here forgive me and don't start a flame ;)) So you should report chan_capi bugs to it's creator at www.junghanns.net, where you downloaded it (or maybe you've found it in a package?). Since I've the feeling that the creator of chan_capi don't belive that the ISDN future is chan_capi anymore, but his bristuff, I also suggest you: a) use his bristuff (same site as above) b) if you have kernel 2.6, use mISDN kernel patches and chan_mISDN, that is, seems, well supported and developed (and works, with a compilation flag, with asterisk stable and asterisk head as well): http://www.beronet.com/?PageID=3017 Best regards Marco Menardi Luis Vazquez wrote: Hello all I found a bug in the chan_capi driver (really a not implemented message handling and then a false error condition) and I guess I have wrote a patch to fix it (basically I searched the internet for other capi open source implementation an borrowed the code snippet) but I don't know where to send the report and bugfix. I also found some miss-behaviours that I would like to share with other asterisk+chan_capi users. I went to Asterisk bugtracker but I didn't find a capi (or related) section. I also looked at Junghanns.net and I didn't find an asterisk capi users mailing list or a way to report bugfixes to chan_capi. Does anybody knows the best way to submit the report so it's available to anyone? Thanks a lot Luis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New native assisted transfer (atxfer) usage info required
Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My features.conf are these ones: [general] language=it parkext = 521 ; What ext. to dial to park parkpos = 522-525 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 120 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] ; there are different from the default in features.conf.SAMPLE, ; and faster to type when apropriate, OMHO blindxfer = #7; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *7 ; Attended transfer And in the Dial() command I use the tT flag. Is there any other special Dial() flag required in the dialplan? If I hang-up the call is not transferred, nor if I presso *0 (disconnect). I would also like to know how to go back to the caller, if the other extension is busy or doesn't answer or doesn't want to talk with the caller. Could someone provide me the exact settings required, and the keystrokes needed to make it work (successful transfer and aborted transfer, going back to the caller)? A sort of atxfer for dummies :) I'm using the more recent CVS Head. Btw, of course I know that I can have a assisted transfer enabled SIP phone, or use the 3 way calling of my TDM400, but I want to make this feature of asterisk working without any client implementation (that is the goal of atxfer). Thanks a lot Marco Menardi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk QSIG
Hi all, reading around and surfing the net I've found some informations about QSIG PRI protocol, that seems a good choice to integrate 2 PBX systems with PRI interfaces. The question is: which is the state of Asterisk support for that protocol ? I was wondering if I could link a traditional PBX system to Asterisk with a QSIG PRI interface ... Thanks a lot. marco -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.302 / Virus Database: 265.7.1 - Release Date: 19/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bri stuff and unknown signalling type
I've downloaded and compiled zaphfc and libpri. To do that i've downloaded bri-stuff and commented out the asterisk related stuff because i've installed it from a debian package. Does this means that i have to rebuild the whole asterisk thing to support zaphfc? thanks ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hfc card and isdn error E001B
In data Thu, 9 Dec 2004 13:43:05 +0100, hai scritto:
I'm trying to use an hfc based pci card with asterisk but every call fails
falling in the congestion extension.
More info.
This is the output from asterisk:
-- Registered SIP 'marco' at 192.168.0.5 port 5060 expires 120
-- Saved useragent SJLabs-SJphone/1.30.252 for peer marco
-- Executing Dial(SIP/marco-c6d7, Modem/g1:05xx) in new stack
-- Called g1:05xx
-- Saved useragent SJLabs-SJphone/1.30.252 for peer massimo
-- Modem[i4l]/ttyI1 is busy
-- Hungup 'Modem[i4l]/ttyI1'
== Everyone is busy/congested at this time
-- Executing Congestion(SIP/marco-c6d7, ) in new stack
== Spawn extension (default, 005xx, 2) exited non-zero on
'SIP/marco-c6d7'
here are my config files:
**
modem.conf
[interfaces]
context=remote
driver=i4l ; isdn4linux - an alternative to i4l is to use chan_capi
language=en
type=autodetect
stripmsd=0
dialtype=tone
;mode=answer
;mode=ring
mode=immediate
group=1
msn=059465066
incomingmsn=*
outgoingmsn=059465066
device = /dev/ttyI0
device = /dev/ttyI1
***+
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
;TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
TRUNK=Modem/g1
[default]
include = echotest
exten=_0.,1,Dial(Modem/g1:${EXTEN:1})
exten = _0.,2,Congestion
exten = 101,1,dial(SIP/marco)
exten = marco,1,goto(101,1) ; To be able to dial with text
exten = 102,1,dial(SIP/fabio)
exten = fabio,1,goto(102,1) ; To be able to dial with text
exten = 103,1,dial(SIP/stefano)
exten = stefano,1,goto(103,1) ; To be able to dial with text
exten = 104,1,dial(SIP/massimo)
exten = massimo,1,goto(104,1) ; To be able to dial with text
[echotest]
exten = 600,1,Playback(demo-echotest) ; Let them know what's going on
exten = 600,2,Echo ; Do the echo test
exten = 600,3,Playback(demo-echodone) ; Let them know it's over
exten = 600,4,Hangup
++
modules.conf
[modules]
autoload=yes
noload = pbx_gtkconsole.so
;load = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_musiconhold.so
noload = chan_alsa.so
;noload = chan_oss.so
[global]
chan_modem.so=yes
ciao
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[Asterisk-Users] hfc card and isdn error E001B
I'm trying to use an hfc based pci card with asterisk but every call fails
falling in the congestion extension.
exten = _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr)
exten = _0.,2,Congestion
Looking in the syslog i can see:
isdn: HiSax,ch0 cause: E001B
it seems that this is a terrible error when arrives... hard to tell what is
the cause. Also terrible is finding a lot of material about this error in
german and not knowing german.
The only thing that keeps me running is the fact that using the red hat
isdn config found in Knoppix (BTW i'm using a debian testing now and i
booted knoppix to try something about this error) it's possible to make a
connection to the provider... so the line is ok and the card works...
...
So:
OS: Debian testing
asterisk 1.0.2
isdn card: HFC based, type 35
what else... let me know if you need some other information.
Thanks for any help
ciao
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[Asterisk-Users] Snom 220 Problem
Hi all, I'experiencing problems with my Snom 220 telephones. I bought 2 of them, and as I receivedthe phones I started them up and I used the web interface to configure them. Both telephones, after first boot, are network unreachable: they don't answer to ping requests and they, obviously, don't work. After talking with Snom support I tried to move then in a completely new network environment and I had the same behaviour, so I askedto my reseller to replace my telephones. I just received my two new devices, and I plugged the firs one, configured it with web interface and, after first reboot the same orrible behaviour !!! I can say that I am not configuring advanced options, nothing more than language and SIP settings, but maybe I'm doing something wrong. I can say that I own also Cisco 7940 and Budgetone phones, I got them working in a couple ofhours.Anyone experienced the same problem ? Thanks Marco Vescovi --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.793 / Virus Database: 537 - Release Date: 10/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue and Agent functionality
Chris,
I agree with your assessment of asterisk's queues. I took Robert's reply to
my original post, and came up with a way to tackle your first scenario (no
agents in queue=caller in limbo) with his idea of setting variables. My idea
deals with setting global variable states for each agent. I only have 4
agents, so it should work for me fairly easily. In the extensions.conf file
I would have something like this:
[globals]
GCSR1=off
GCSR2=off
GCSR3=off
GCSR4=off
Then, in the context where my agents log in/out of queue, I set the global
variable to on/off depending on their action. When the agent dials 800,
GCSR1 becomes 'on'. When they dial 801##, GCSR1 becomes 'off'.
[fromcsr1]
exten= 800,1,AgentCallbackLogin(101|[EMAIL PROTECTED])
exten= 800,2,SetGlobalVar(GCSR1=on)
exten= 800,3,Hangup
exten= 801,1,AgentCallBackLogin(101)
exten= 801,2,SetGlobalVar(GCSR1=off)
exten= 801,3,Hangup
Then, in my queue, I check for the value of GCSR1 before dumping them to the
queue. Otherwise, dump them to VM. Obviously, the GotoIf would have to check
if GCSR1 = on | GCSR2 = on | GCSR3 = on | etc... For my testing, I was just
using GCSR1.
[queue]
exten = 1,1,DigitTimeout,1
exten = 1,2,ResponseTimeout,1
exten = 1,3,GotoIf($[${GCSR1} = on]?4:5)
exten = 1,4,Queue(order|tT)
exten = 1,5,Goto(generalvm|s|1)
While this idea seems to make sense (in my head), I am unable to make it
work. For example, my GotoIF command does work, so the value of GCSR1 will
determine which path the caller takes. The part that doesn't work is in the
[fromcsr] context. My SetGlobalVar(GCSR1=on) seems to have no effect,
therefore, making my solution not work.
Does anyone have any ideas?
Thanks,
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Icide
Sent: Saturday, September 25, 2004 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Queue and Agent functionality
I've seen alot of posts lately on Queue and Agent functionality, and alot of
hacks to make them do different things that most call center managers want.
In the sake of doing this one time, I'd like to develop a single list of
request so we can consolidate a feature request for the Queue/Agent system.
Here are the ones that I run into the most:
1. Queue should know the status of agents assigned to a queue and act
accordingly.
Here are a couple examples of the problem.
A queue has no agents logged in and handling the queue, a call comes in for
the queue, the call remains in the queue until either an agent logs in, or
the queue reaches it's timeout. What it should do is immediately time out
setting priority +101. Normal timeout (caller in busy queue with agents
active) should exit with priority set +1.
A Queue has active agents in a prioritized fashion. Agent 1 is priority 1,
2 is 2, 3 is 3, and 4 is 4. Agent 1 needs to make an outbound call as does
agent 2. Both are now 'busy'. The Queue still attempts to call agent 1,
gets 'busy' back from the sip device (i've only tried this with sip), and
then the system appears to wait for something like 7-8 seconds before trying
the next agent in line.
2. The queue system should allow a set of messages to be played at specific
times. For example, a message that is played upon entry into the queue and
no other time, the current set of messages played every frequency=XX, a
message played to the caller when the call is accepted by an agent (eg
transfering), finally, a set of messages played to the user based upon a
predefined period int he config file.. see example below
message1-time=time in seconds
message1-frequency=never|once|always
message1=message1-file-loc
message2-time=time in seconds
message2-frequency=never|once|always
message2=message2-file-loc
Where a message messageX-file-loc is played never|once|always every
time in seconds.
if time is set to 0, or freqency is set to never, the message is not played.
If time is set to 0, and frequency is set to once, message is played at
messagex-time, and never again.
if time is set to 0 and frequency is set to always, message is played every
messagex-time in seconds.
3. Agent timeout (logs the agent off if they do not respond to a ring in a
defined about of time) does not track across calls. For example, if an
agent steps away and forgets to log out, then thier phone will ring based
upon whatever call strategy is used. If the agent timeout is set higher
than the time the queue polls a set of agents they will never be logged out.
The timer needs to increment per agent across multiple polls. So if my
queue poll timer is 20 secons, but the agent timeout is set to 60 seconds,
the preferred function would be to log the agent out of the queue if they
completely miss three poll events.
4. If a caller empties a handled queue (active agents) with no callers, the
caller will still hear messages (you are first in queue, etc.). This should
not occur. Someone posted a 2-line patch
RE: [Asterisk-Users] Queue and Agent functionality
Philipp... Good point. I totally missed the auto-logoff angle. At this time, I'm not using auto-logoff, but I really should. Are you saying that the reason my code was not working is because AgentCallbackLogin() invalidates the lines afterwards? In my example, the global variable was not being set--or at least that's my best guess as to why it was not working. I can try setting the global variable FIRST, and the follow by using the AgentCallbackLogin(). Ok, I'm going to follow on the path that Robert mentioned a few e-mails back re: using the monestary script. I've experimented by telneting into the manager CLI, and was able to sucessfully get the 'show agents' command to work. Besides telneting into the manager CLI (or writing a script to do this), is there any other way to run this command and capture the results? Asuming I have to write a perl script similar to monestary one, what should I do once I determine if there is an agent logged in? Am I able to set an * global variable from within this external perl script? If not, what do you suggest? Thanks, Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Saturday, September 25, 2004 5:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Queue and Agent functionality Hi! [fromcsr1] exten= 800,1,AgentCallbackLogin(101|[EMAIL PROTECTED]) exten= 800,2,SetGlobalVar(GCSR1=on) exten= 800,3,Hangup determine which path the caller takes. The part that doesn't work is in the [fromcsr] context. My SetGlobalVar(GCSR1=on) seems to have no effect, therefore, making my solution not work. Does anyone have any ideas? Unfortunately AgentCallbackLogin() _itself_ initiates the hangup, which means that any following priorities in your dialplan are useless. Besides your approach isn't yet perfect, what if an agent gets auto-logged out because he/she hasn't answered within the time limit? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4155ea5e217701945313673! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agents and queues
Hello all, I am currently using asterisk in a call center configuration. I have created a queue where our customers listen to music while an agent picks up. Pretty standard stuff. I have a total of 5 agents who are able to sucessfully sign-in and sign-out of the queue by using setting AgentCallbackLogin. Here's my problem. Say there are no agents signed-on. The caller is still able to enter the queue by hitting the appropriate menu option. Once in the queue, they will listen to hold music indefinately...or until the first agent logs-in to the queue. This is obviously a bad experience for the user. Basically, they wont know they have an indefinate hold time. How can i determine if there are any agents signed-in, and if not, take them straight to voice mail with a message like "Sorry, we are unable to take you call now, please leave a message..."?? Any help with this would be appreciated. thanks, Marco -- Marco Nicolayevsky Chief Technology Officer MisterArt.com LP http://www.misterart.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with ISDN (NT-Mode) - Error Messages inside
Hello everyone, I try to connect the Asterisk PBX Server on my NTBA (NT1) - actually its hard-connected on the box, but I've tried a crosslink-ISDN cable, too. My ISDN-Phone is connected on one port (of two) S0-Ports on NTBA. When I start Asterisk and want to make a call I'll geht a lot of this messages -- 040827-034639 WARNING[213005]: chan_zap.c:6902 zt_pri_error: PRI: !! Got I-frame while link state 0 040827-034639 WARNING[213005]: chan_zap.c:6902 zt_pri_error: PRI: !! Got I-frame while link state 0 040827-034643 WARNING[213005]: chan_zap.c:6902 zt_pri_error: PRI: !! Received short unnumbered frame 040827-034644 WARNING[213005]: chan_zap.c:6902 zt_pri_error: PRI: !! Received short unnumbered frame 040827-034646 WARNING[213005]: chan_zap.c:6902 zt_pri_error: PRI: !! Received short unnumbered frame And in loop -- zaphfc: empty HDLC Frame received Any ideas what it could be? My Zaptel.conf -- -- loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 -- My Zapata.conf -- [channels] switchtype=euroisdn signalling=bri_net_ptmp pridialplan=local echocancel=yes immediate=no overlapdial=yes group=1 context=default channel=1-2 -- If you need additional infos plz tell me. :) Thanks, Marco ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls
Hello everyone, I bought 2 HFC-ISDN Cards and want to run the first card in NT-Mode an the second one in TE-Mode. Everything looks ok under SuSE 9.1, but I can't dial out. I removed one card, for testing purposes and want to run this one card in TE-Mode. I only want to make a call with my Grandstream BT-101 over Asterisk via ISDN. When I try to make a call I get: - Executing Dial(SIP/11-3ef2, Zap/g1/00MY-NUMBER) in new stack Aug 25 18:11:00 NOTICE[1117453232]: app_dial.c:727 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time - zap show channels says: - Chan Extension Context Language MusicOnHold pseudodefault 1default 2default - ztcfg -vvv tells: - Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. - Only one point in zttool I don't really understand: - Current Alarms: No alarms. â â â âSync Source:Internally clocked â a â â âIRQ Misses: 0 â a â â âBipolar Viol: 0 â a â â âTx/Rx Levels: 0/ 0 â a â â âTotal/Conf/Act: 3/ 3/ 0 - Conf = configured or conflicted? I try a Loop but nothing happend. TxA, TxB etc. are empty, too. Can someone help me? - I really need some sample configs, too. Which linux distribution runs smoothest with Asterisk? Thanks! Marco Czudej ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and AVM FritzBox Fon (Germany)
Hello everyone, I've some stange problems with my AVM FritzBox Fon (http://www.avm.de/de/Presse/Informationen/2004/2004_06_22.php3) and Asterisk. I want to build up an internal SIP Network for testing purposes. I can phone with my Grandstream BT-101 to other BT-101 or SIP-Software, but not to my AVM FritzBox Fon. It looks like the Box connects successfully to Asterisk, but you can't call other internal SIP Accounts. You also can't call to the Box. The Phone on the Box rings, but I you want to start the call you hear nothing. And after a few seconds, the Grandstream responds a 403-Error. Here a snapshot from my sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = default [10] type=friend username=10 fromuser=10 host=dynamic secret=SECRET canreinvide=no [11] type=friend username=11 fromuser=11 host=dynamic secret=SECRET Here my extensions.conf [general] static=yes writeprotect=no [default] include = 10 include = 11 [10] exten = 10,1,Dial(SIP/10,45) exten = 10,2,Hangup [11] exten = 11,1,Dial(SIP/11,45) exten = 11,2,Hangup I've tried a qualify=yes in the sip.conf, but then the Box will loose connection after a few seconds: Aug 21 13:16:10 NOTICE[98310]: chan_sip.c:7653 sip_poke_noanswer: Peer '10' is now UNREACHABLE! I would be really pleased for any help :) Regards, Marco canreinvide=no ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integrating an old PBX with Asterisk
Hi all, I was thinking about integrating an old PBX with Asterisk and I was wondering some possible configurations. The question is: which is the best way to let the 2 systems interact ? I can imagine some possible scenarios: - scenario 1: I want to use other then old PBX terminations (ie I have to link the 2 systems with some internal number line) In this scenario I could think to give each user a dedicated old line number from old PBX to a 'dedicated' port of a TDM card. Pros: easy configuration (one - to - one mapping), no old PBX configuration changes, users with new SIP phone can still mantain their old extension. Dis: expensive (one TDM card each 4 ext), not scalable (2 limits:free extension on the old PBX and PCI slots in the * server to add TDM cards), when I receive a call from a old extension and I want to forward it to another old PBX extension I am actually using 2 lines between * and the old PBX. - scenario 2:I want to link the 2 PBX with a trunk of n linesnd use an arbitrary number of SIP phones being able to have # of SIP phones then # of lines. Pros: less expensive then scenario 1 because the number of lines I have to use between * and old PBX is based onblock probability I choose to have, more scalable for the same reason, virtually no limit to SIPextension number Dis: same call transfer problem of above, if the old PBX doesn't support some sort of DID between its extension I have to tell * to answer the line and then to ask the required extension, configuration changes to old PBX... I know that probably the best way should be to add a digital card to old PBX and havea trunk between two systems, but the PBX is really old and I'm not sure I can still find an expansion card. Any suggestion or tip ??? thanks marco ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.732 / Virus Database: 486 - Release Date: 29/07/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 29/07/2004
Re: [Asterisk-Users] Newbie voiceplus + asterisk
On Wed, May 12, 2004 at 10:25:54PM -0400, [EMAIL PROTECTED] wrote: *CLI dial 17185551212 No such extension '17185551212' in context 'local' ok I added : [local] include = default to extensions.conf and I get : May 12 22:31:43 WARNING[311316]: chan_iax2.c:1732 create_addr: No such host: voicepulse May 12 22:31:43 NOTICE[311316]: app_dial.c:527 dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy at this time -- Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie voiceplus + asterisk
ok, I got it voicepulse vs voiceplus more questions coming I am sure... -- Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Variable digit length in national dial plan
Dear All,
we are integrating Asterisk between PSTN and a Panasonic PBX using 2 E100P
cards
and PRI lines.
--PSTN--PRI---Asterisk-PRI---Panasonic PBX
Everything works fine except for outgoing calls.
In Italy the local call may have a variable lenght (9 or 10 digit). How
can we
instruct Asterisk to wait the possibly 10th digit?
We tried to configure the extension.con as follow:
;emergency number
exten = _1xx,1,Dial(Zap/g2/${EXTEN})
;international
exten = _00XX.,1,Dial(Zap/g2/${EXTEN})
;Local 9 Digit
exten = _0,1,Dial(Zap/g2/${EXTEN})
;Local 10 Digit
exten = _0X,1,Dial(Zap/g2/${EXTEN})
But it does not work while attempting a number of ten digit asterisk
sends out a 9 lenght digit number, ignoring the 10th digit.
We have a work around that consists in terminating the dialed number
with a # character... But it is very unprofessional solution.
May someone help us?
Best regards,
Marco
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[Asterisk-Users] RTP channel
Hi all, i would like to know if it is possible to bridging the rtp traffic over Asterisk... I would like that the RTP flow is not controlled by * but by the endpoint. Is it possible??? Any suggestion to do this??? Thanks Marco Ë^®+$R²f¢)à+-Ë^®+$R²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
[Asterisk-Users] * with external sip proxy
Hi all, i'm tring ro use sip with an external sip proxy as vocal or ser. My scenario is Vocal or SER Asterisk with cnah_oh323 - Gatekeeper I would like that sip termial register themself to Vocal or ser and the h.323 terminal to gatekeeper. When i place a call from h323 side to sip side all work When a try to place a call form sip to h323 nothing happen Does someone try this??? Any suggestion will be appreciate Tnx Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error chan_oh323.so
Hi all, I want to install h.323 support for *, but when I launch * from shell command asterisk -vvvc I have the next error screen: [chan_oh323.so]WARNING[1024]: File loader.c, Line 226 (ast_load_resource): liboh323wrap.so: cannot open shared object file: No such file or directory WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so failed! It can't loading chan_oh323.so, I have this module in the /usr/lib/asterisk/modules directory, but it does not recognize this library, and at the same time does not recognize liboh323wrap.so Someone has installed and using with success this oh323 package from inaccess networks ??? thanks in advance, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error chan_oh323.so
Hi all, I want to install h.323 support for *, but when I launch * from shell command asterisk -vvvc I have the next error screen: [chan_oh323.so]WARNING[1024]: File loader.c, Line 226 (ast_load_resource): liboh323wrap.so: cannot open shared object file: No such file or directory WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so failed! It can't loading chan_oh323.so, I have this module in the /usr/lib/asterisk/modules directory, but it does not recognize this library, and at the same time does not recognize liboh323wrap.so Someone has installed and using with success this oh323 package from inaccess networks ??? thanks in advance, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info sip/h.323 interoperability
Hi all, I'm a student (my thesis work consist in testing interopearbility SIP/H.323) and I begin to work with asterisk in this days. I have to testing to SIP/H.323, since today I have used Vocal system, but there are some problem for this features. In the asterisk mailing list, in the next message I've seen an e-mail [Asterisk-Users] Cisco 7960 with Asterisk H.323 Shaun Ewing [EMAIL PROTECTED] Mon, 26 May 2003 21:56:42 +1000 I've red in that mail youhave over a hundred 7960s using Asterisk and chan_h323 , so my question is: Asterisk supports this interoperability ? I have done some test with Vocal to make calls from IP cisco Phone via sip/h323 translator to connect with Netmeeting or other h.323 end points... If this interoparability is supported which module I needs ? Thanks for attention, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting up fax on *
Hello All, I am using an E100P card on a PRI line. I need to setup a FAX extension. Can somebody help me please? Marco
