[Asterisk-Users] Problem with OpenPhone-Asterisk
Hello all, I just installed Asterisk with H323 support (chan_h323 from Jeremy McNamara). But experience problem while connecting OpenPhone to Asterisk Here is h.323 trace: 5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720, handle=27 5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225 Started incoming call thread 5:37.445 H225 Answer:9cc1250 transports.cxx(1127) H225 Awaiting first PDU 5:37.470 H225 Answer:9cc1250 h323pdu.cxx(517) H225 Receiving PDU: setup 5:37.471 H225 Answer:9cc1250 transports.cxx(1136) H225 Incoming call, first PDU: callReference=27042 5:37.471 H225 Answer:9cc1250 h323caps.cxx(1942) H323 Added capability: UserInput/hookflash 1 5:37.472 H225 Answer:9cc1250 h323caps.cxx(1942) H323 Added capability: UserInput/RFC2833 2 5:37.472 H225 Answer:9cc1250 h323caps.cxx(2008) H323 Found capability: UserInput/hookflash 1 5:37.473 H225 Answer:9cc1250 h323caps.cxx(2008) H323 Found capability: UserInput/RFC2833 2 5:37.473 H225 Answer:9cc1250 rfc2833.cxx(81) RFC2833 Handler created 5:37.474 H225 Answer:9cc1250 h323ep.cxx(2227) H323 Created new connection: ip$10.120.160.15:3172/27042 5:37.474 H225 Answer:9cc1250 h323.cxx(1761) H225 Handling PDU: Setup callRef=27042 5:37.475 H225 Answer:9cc1250 h323ep.cxx(1898) H323 Clearing connection ip$10.120.160.15:3172/27042 reason=EndedByTransportFail 5:37.476 H225 Answer:9cc1250 h323.cxx(1540) H323 Call end reason for ip$10.120.160.15:3172/27042 set to EndedByTransportFail 5:37.476 H225 Answer:9cc1250 h323.cxx(1558) H225 Sending release complete PDU: callRef=27042 5:37.477 H225 Answer:9cc1250 h323pdu.cxx(517) H225 Sending PDU: releaseComplete 5:37.478 H225 Answer:9cc1250 transports.cxx(1166) H225 Signal channel stopped on first PDU. 5:37.479 H323 Cleaner h323ep.cxx(1955) H323 Cleaning up connections 5:37.479 H323 Cleaner h323.cxx(1595) H323 Connection ip$10.120.160.15:3172/27042 closing: connectionState=NoConnectionActive 5:37.479 H323 Cleaner h323neg.cxx(334) H245 Stopping MasterSlaveDetermination: state=Idle 5:37.479 H323 Cleaner h323neg.cxx(561) H245 Stopping TerminalCapabilitySet: state=Idle 5:37.479 H323 Cleaner transports.cxx(1109) H323 H323Transport::Close 5:37.480 H323 Cleaner transports.cxx(1191) H323 H323Transport::CleanUpOnTermination for H225 Answer:9cc1250 5:37.480 H323 Cleaner h323.cxx(1659) H323 Connection ip$10.120.160.15:3172/27042 terminated. 5:37.480 H323 Cleaner h323.cxx(1490) H323 Connection ip$10.120.160.15:3172/27042 deleted. I dont have firewall and both machines are in the same LAN. What does this reason EndedByTransportFail mean? Can anybody help? Thanks in advance! Regards, Vassil Kolarov ___ 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: Re: [Asterisk-Users] Re: Howto Setup TFTP server on Linux for Cisco
Thnx. Will try during the weekend. Michiel van Baak Terrazur - Originele Bericht - Van: Doug Lytle Aan: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion Datum: Wednesday, 26 January 2005, 18:54 Onderwerp: Re: [Asterisk-Users] Re: Howto Setup TFTP server on Linux for Cisco Michiel van Baak wrote: Hi, Do you happen to know if those image will work on a cisco 7905g ? I have chan_sccp now but SIP is what i want to do. Michael, SIP040406A is labeled on Ciscos website for the CP-7905G Doug ___ 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] Re: phone rings when I'm using it over VOIP - WHY?
In article [EMAIL PROTECTED], Joseph [EMAIL PROTECTED] wrote: When I use my phone to make VOIP call and another calls comes from POTS my phone rings to POTS caller. Why? Shouldn't it generate busy signal! Yes, but there are all sorts of configuration errors that could result in the behaviour described. Without knowing your particular setup, it is impossible to know what the cause could be. Perhaps you could describe in more detail. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] UPS for Asterisk
Indeed, I'm thinking of using 2 CompactFlash ATA disks. One fully read only with just a small partition writable that will keep /etc/asterisk (astlinux mounts read-only always and only mounts read-write if you need to change/save the config). No worries about unclean shutdown. The second disk I will use for voicemail, and I can swap it every year before it wears down. Better than that, mirror the disk. Then when one drive fails Linux will automatically use the other disk. You can go one step more and define a third disk as a hot spare then after Linux detects the drive failure and switches to the surviving twin it will also bring up the hot spare and begin building a replacement for the dead twin. You can then swap out the dead drive with no need to power down the server and declare the new drive as the new hot spare I would mirror the read-only patition also It you truely want 5 nines you have to set things up so that you can do normal maintanance (swapping out drives, power supplies and the like without powering down. I thought of that but that's not much use with flash disks. Flash can only be written to a number of times. If I would do raid1 on two flash drives and they reach that limit they might die shortly after each other. Raid1 is a good solution when doing real harddrives. I may consider doing raid1 with two laptop harddiscs. laptop drives do not consume a lot of power nor do they produce any heat. Alternatively I could consider hardware raid1 with one ATA flash drive and one laptop drive, chances of them dying both at the same time are slim. I will do some testing on the behaviour of * when the partition where voicemail is stored is failing. If * will just skip voicemail that would be good enough for me, I don't care about voicemail being unavailable, i just dont want it to bring the whole box down. ___ 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] IAX Softphone
Both applications work well and the sound quality seems to be identical between them. Firefly has limited features but it is well polished and looks nice. It gives the impression that it is easy to use so would be a good choice for users who are not very computer literate. Diax has more features such as being able to alter the headset volume during the call without having to go to the volume control setting in the mixer. It has the option to play the ringing tone out of a different sound source than the normal volume which is great if you use a bluetooth headset as you can set the speakers to ring and not wear the headset all the time. It is only really the address book that I dont like in Diax. You have buttons you can program but to search for a contact you have to open the address book. Firefly has an expanding list which is a lot simpler. If you have loads of contacts the Firefly approach will probably become limiting. Firefly does not allow you to enter the users name so all CLID is set to the number only. Diax allows you to specifically set the CLID name and number which is better. Diax supports showing the number of new and old voicemail messages although I have not managed to get that working yet. Diax supports 8 connections so you can switch between them easily and choose which system you wish to make the call over. Diax also has AGC support which enables me to leave the 'mic boost' turned off and still get acceptible microphone volume. This reduces the background noise picked up by the mic. In conclusion therefore I think Diax is the much better application and it is only its addressbook and the not so intuitive GUI for non computer literate users that lets it down compared to Firefly. On Wed, 2005-01-26 at 17:10, Dan wrote: Hi, - Original Message - From: Gareth Blades [EMAIL PROTECTED] On Wed, 2005-01-26 at 15:50, Germn Micale wrote: Hi, Does someone know an ActiveX IAX softphone? I need a free softphone to connect with Asterisk from a web page. Regards I use Firefly as a free IAX client and it works well. I have also used diax which has more features (multiple line support for example) but the quality does not seem quite as good. Please develop a little bit on this one. What do you mean by quality does not seem quite as good? It is about sound, interface, stability? I really need a feedback on this sentence. Thank you and best regards, Dan ___ 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] DUNDi on Asterisk
Hi, Has anybody tried DUNDi Enterprise Configuration using IAX on Asterisk? If yes, could you please explain in detail the configuration of this feature works and the importance of this feature. I went through the info. available at http://voip-info.org but could not understand how the configuration works. Thanks, Jagan ___ 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] Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250
hi,
thanks to peter i solved my problems with the asterisk server spliced
between the telco and our ericsson BP250.
the problem was solved by setting 'overlapdial=yes'
Peter Svensson wrote:
Am Dienstag, den 25.01.2005, 22:39 +0100 schrieb Frank Sautter:
the setup desired with asterisk spliced in:
Arcor TelCo PRI(E1) P1 asterisk P2--- Ericsson BP250 PRI(E1)
Extension '' in context 'pri-ericsson' from '123498765' does not
exist
It sounds like the Ericsson pbx uses overlap dialing. Try enabling
that on both links in the zapata.conf file and see if it works better.
For immediate=no you should not match the s context. I think
exten = _.,1,Dial(Zap/g2/${EXTEN})
is more correct. Or use _XXX for a three digit DID.
i had to modify my dialplan on some points (thanks again to peter) and
twiddle with the callerid, our trunk MSN and the extensions, but it
seems to work.
today is our first working-day with asterisk in-between - so far no
problems (i hope it keeps this state).
here are the essential parts of the configuration files.
/etc/zaptel.conf
# TDM40B quad fxs analog-modules
span=1,0,0,ccs,hdb3,crc4
fxoks = 1-4
# TE405P/TE410P quad E1
span=2,1,0,ccs,hdb3,crc4
bchan=5-19,21-35
dchan=20
span=3,0,0,ccs,hdb3,crc4
bchan=36-50,52-66
dchan=51
span=4,2,0,ccs,hdb3,crc4
bchan=67-81,83-97
dchan=82
span=5,0,0,ccs,hdb3,crc4
bchan=98-112,114-128
dchan=113
loadzone=nl ; there is no 'de' zone right now
defaultzone=nl
/etc/asterisk/extensions.conf
[pri-external]
exten = _5678.,1,SetCIDNum(0${CALLERIDNUM}) ; Add a leading zero
exten = _5678.,2,Goto(${EXTEN:4}|1) ; Strip trunk digits from the DDI
exten = h,HangUp()
include = durchwahl
include = pri-external-route
[pri-external-route]
exten = _.,1,Dial(Zap/g3/${EXTEN})
[pri-ericsson]
include = durchwahl
include = pri-ericsson-route
exten = h,HangUp()
[pri-ericsson-route]
exten = _XX.,1,SetCIDNum(${CALLERIDNUM:8})
exten = _XX.,2,SetCIDName('my name')
exten = _XX.,3,Dial(Zap/g2/${EXTEN})
/etc/asterisk/zapata.conf
[channels]
;### Quad FXS Card (TDM40B)
language=de
context=analog-lines
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
usecallingpres=yes
sendcalleridafter=1
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
signalling=fxo_ks
callerid=Harry Hirsch171
mailbox=171
accountcode=analog1
channel = 1
callerid=Hans Dampf172
mailbox=172
accountcode=analog2
channel = 2
callerid=Mork vom Ork173
mailbox=173
accountcode=analog3
channel = 3
callerid=Faxe179
mailbox=0
accountcode=analog4
channel = 4
;### Quad PRI(E1) Card (TP405P/TP410P)
language=de
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
musiconhold=default
callgroup=1
pickupgroup=1
immediate=no
overlapdial=yes
accountcode=pri
context=pri-external
group = 2
signalling=pri_cpe
channel = 5-19,21-35
context=pri-ericsson
group = 3
signalling=pri_net
channel = 36-50,52-66
context=pri-debug1
group = 4
signalling=pri_cpe
channel = 67-81,83-97
context=pri-debug2
group = 5
signalling=pri_net
channel = 98-112,114-128
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Re: [Asterisk-Users] Re: Polycom and call waiting again..
On Wed, 2005-01-26 at 17:58 -0500, Sean A. Newton wrote:
On Wed, 26 Jan 2005, Kevin P. Fleming wrote:
But you _can_ use SetGroup/CheckGroup/GetGroupCount if you don't put the
SIP peer directly into the queue, but instead add a Local/.. channel
that makes the Queue call out to the agent via a special context in your
dialplan. This special context can then do anything it wants, including
returning Busy/Congestion back to the Queue app if needed.
I understand the concept of what your saying, but I can't seem to
visualize how to implement it. Do you have an example of this?
I would very much appreciate it.
I haven't had time to actually do this yet, it's on my 'list' but
something like this:
[local-stuff]
; This is where we pretend a channel is an extension
exten = 1234,1,SetGroup(SIP1234)
exten = 1234,2,CheckGroup(1)
exten = 1234,3,Dial(SIP/1234,15)
exten = 1234,104,Busy
[queue-stuff]
exten = 6939,1,AddQueueMember(Local/${CALLERIDNUM})
Something like that Obviously requires certain configs in sip.conf
etc...
PS, This could be totally outrageously wrong, so don't blame me if it
breaks.
Regards,
Adam
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[Asterisk-Users] Moh in meetme doesn't work if I transfer to meetme
Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-: Jan 27 11:02:23 DEBUG[6133]: chan_iax2.c:5762 socket_read: We don't do requested format ilbc, falling back to peer capability 1550 -- Accepting AUTHENTICATED call from 192.168.0.101, requested format = 1024, actual format = 2 -- Executing MeetMe(IAX2/[EMAIL PROTECTED]/1, 81|pMs) in new stack == Parsing '/etc/asterisk/meetme.conf': Found == Parsing '/etc/asterisk/meetme_additional.conf': Found Jan 27 11:02:23 WARNING[6133]: channel.c:1901 ast_request: No channel type registered for 'zap' Jan 27 11:02:23 WARNING[6133]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '81' Jan 27 11:02:23 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'conf-onlyperson' (language 'si') Jan 27 11:02:23 DEBUG[6133]: chan_iax2.c:5346 socket_read: Ooh, voice format changed to 2 Jan 27 11:02:27 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 27 11:02:27 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 27 11:02:27 DEBUG[6133]: app_meetme.c:695 conf_run: Placed channel IAX2/[EMAIL PROTECTED]/1 in ZAP conf 1023 -- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/1 Jan 27 11:02:27 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Jan 27 11:02:27 DEBUG[6133]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Jan 27 11:02:27 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 27 11:02:37 DEBUG[6133]: chan_iax2.c:5528 socket_read: Immediately destroying 1, having received hangup Jan 27 11:02:37 WARNING[6133]: app_meetme.c:962 conf_run: Unable to write frame to channel: Resource temporarily unavailable -- Stopped music on hold on IAX2/[EMAIL PROTECTED]/1 -- Now if I dial another local extension (201) and transfer to conference from there, moh doesn't start. I get: --- == Channel 'IAX2/[EMAIL PROTECTED]/1' jumping out of macro 'dial' == Channel 'IAX2/[EMAIL PROTECTED]/1' jumping out of macro 'exten-vm' -- Executing MeetMe(IAX2/[EMAIL PROTECTED]/1, 81|pMs) in new stack == Parsing '/etc/asterisk/meetme.conf': Found == Parsing '/etc/asterisk/meetme_additional.conf': Found Jan 27 11:06:30 WARNING[6133]: channel.c:1901 ast_request: No channel type registered for 'zap' Jan 27 11:06:30 WARNING[6133]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '81' Jan 27 11:06:30 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'conf-onlyperson' (language 'si') Jan 27 11:06:33 DEBUG[6133]: acl.c:176 ast_apply_ha: # Testing 192.168.0.160 with 192.168.0.0 Jan 27 11:06:33 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 27 11:06:33 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 27 11:06:33 DEBUG[6133]: app_meetme.c:695 conf_run: Placed channel IAX2/[EMAIL PROTECTED]/1 in ZAP conf 1023 Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random Jan 27 11:06:37 DEBUG[6133]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 Jan 27 11:06:37 DEBUG[6133]: chan_sip.c:1313 create_addr: Setting NAT on VRTP to 0 Jan 27 11:06:37 DEBUG[6133]: acl.c:176 ast_apply_ha: # Testing 192.168.0.160 with 192.168.0.0 Jan 27 11:06:37 DEBUG[6133]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Jan 27 11:06:42 DEBUG[6133]: chan_iax2.c:5528 socket_read: Immediately destroying 1, having received hangup Jan 27 11:06:42 WARNING[6133]: app_meetme.c:962 conf_run: Unable to write frame to channel: No such file or directory Thanks in advance, Rob. ___ 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] How to check sip channel with smoething similar to ping ?
Hi, I saw Nagios plugin that can check if Asterisk IAX2 channels is alive. Can I do the same with SIP channel ? Regards, Rob. ___ 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] Re: New Firefly version
Hi Adam, Sory to say it, bu it still interupt the mouse if you have microsoft wireless mouse/keayboard. The mouse jumps around on the screen. Any news on this ? /HHA Adam Hart wrote: As always, I'm happy to announce a new version of Firefly. Firefly 1.9.8 has more of what you want and less of what you don't http://www.virbiage.com/firefly/download/firefly-thirdparty.exe There's a few bug fixes - notably fixed the Reject button and sending of audio before answering in some circumstances. -Adam ___ 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
RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Hi Alex, Concerning the web interface, in this version we need the register_globals = On I will try to change it in the next release... To find out the error on the agi, can you run the agi script manually. php areskicc.php You will get more details about the error! Regards, Areski On Thu, 2005-01-27 at 03:07, Alexander Romanov wrote: Hi, I've tried it and could not get to work any of them (webapp and agi). On webapp I do not get a full menu, just logout and disconnect With agi nothing happens when I execute the script. -- Executing Answer(SIP/2204-6221, ) in new stack -- Executing Wait(SIP/2204-6221, 2) in new stack -- Executing AGI(SIP/2204-6221, areskicc.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php -- AGI Script areskicc.php completed, returning 0 -- Executing Wait(SIP/2204-6221, 2) in new stack -- Executing Hangup(SIP/2204-6221, ) in new stack == Spawn extension (local, 40, 5) exited non-zero on 'SIP/2204-6221' I have followed instructions to the letter. Am I missing something? Alex. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Thursday, 27 January 2005 4:05 AM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc.. * take care of multiple calls using the same Cardnumber * Caller gets informed about his credit Announce the remaining credit * Caller is requested to enter a destination number * Announce the maximal call time for the given destination number It calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout * Interupt the call if the card balance gets zero Warn the caller about the call interupt 60 30 seconds before the call gets interupted * It connects the Caller to the destination through the configured trunk note : different trunks can be configured and associated by prefix * After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit FEATURES - WEB INTERFACE: * CARD/CUSTOMERS * List customers * Refill customer * CARD/CUSTOMERS * List customers/cards * Refill customer/card * Create customer/card * Generate customers/cards * BILLING * View money situation * View Payment * Add new Payment * RATECARD * List Tariffplan * Create new Tariffplan * Define Tariffplan * TRUNK * List Trunk * Add Trunk * CALL REPORT - BALANCE Last note : It's distributed under GNU GPL Licence. I hope there will have a big interest for the soft, I am waiting your feedbacks... Regards, /Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belad Arezqui www.areski.net E-mail : areski [EMAIL PROTECTED] gmail (.dot.) com ___ 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 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] IAX Softphone
Hi, - Original Message - From: Gareth Blades [EMAIL PROTECTED] Both applications work well and the sound quality seems to be identical between them. Firefly has limited features but it is well polished and looks nice. It gives the impression that it is easy to use so would be a good choice for users who are not very computer literate. Diax has more features such as being able to alter the headset volume during the call without having to go to the volume control setting in the mixer. It has the option to play the ringing tone out of a different sound source than the normal volume which is great if you use a bluetooth headset as you can set the speakers to ring and not wear the headset all the time. It is only really the address book that I dont like in Diax. You have buttons you can program but to search for a contact you have to open the address book. Firefly has an expanding list which is a lot simpler. If you have loads of contacts the Firefly approach will probably become limiting. The address book in 0.9.10a (available for download at the end of this week) is completely redesigned, same for Registration to make it a lot intuitive and easy to use it. Firefly does not allow you to enter the users name so all CLID is set to the number only. Diax allows you to specifically set the CLID name and number which is better. .. and the registration server from which you get that call, which can be very usefull in some circumstances.. Diax supports showing the number of new and old voicemail messages although I have not managed to get that working yet. There is a parameter in iax.conf which must be enabled in order to get this functionality. ; If mailboxdetail is set to yes, the user receives ; the actual new/old message counts, not just a yes/no ; as to whether they have messages. this can be set on ; a per-peer basis as well ; mailboxdetail=yes For the account you must have a line mailbox=xxx This is all you need to make it work. In conclusion therefore I think Diax is the much better application and it is only its addressbook and the not so intuitive GUI for non computer literate users that lets it down compared to Firefly. Address book is changed now. The GUI will be the next step in order to release 1.0 stable in February. If you want to play with the new pre 0.9.9i (not ready yet), you can download it from: http://www.cosmica.com/dante/diax/diax099i.zip Pls keep in mind that this is an intermediate version under heavy development, so do not use it for day to day calls. What's new here: ' - independent codec configuration for each registration server; ' - use control chars in the dial string to send some DTMF codes after dialing '(good for special services, banking, etc): ' - '#' dial separator ' - 'p' pause 1s (long press on '*' key) ' - 'h' hangup (long press on '#' key) ' - accept URLs during a call and open that page in the default browser ' - redesigned Registration and Phonebook forms and operation; ' - redesigned audio level display and volume adjustment ' - phonebook menu replaced with a button; ' - no other files in the basic package, just the exe, DLL and ring file. All ' other files are automatically generated if they do not exist; ' - the old phonebook file is automatically updated to the new format; ' - the old call list file is replaced with the new one (old list is lost); ' - right click on memory buttons to edit them directly, even empty; ' - right click on registration servers buttons to directly edit them ' - Lithuanian and Polish language added; ' 'solved bugs: ' - sometimes corrupted FORM parameter in config file make ' the application to crash as startup; ' - when using the FQDN for the server registration, the button does not goes to green Pls help me improove the Address Book and the rest of the phone interface, if you think is stil not so easy to be used. Thank you and best regards, Dan P.S. ___ 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] enter/leave sound with meetme adminmenu
Hi, I'm missing the enter/leave sound when I activate the adminmenu in a dynamic conference. Did anybody know, how to solve the problem? Klaus ___ 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] IAX Softphone
www.signate.co.uk There is an e-book version. I bought mine from the states, arrived very quickly to the UK - around 5 days, and no postage cost. I ordered the CD of Asterisk with it, but didnt use it, and dont see it as having much value. Book is quite good for getting * running from basics IMHO when used in conjuction with the Wiki. John On Wed, 26 Jan 2005 16:50:13 +0100, Germán Micale [EMAIL PROTECTED] wrote: Hi, Does someone know an ActiveX IAX softphone? I need a free softphone to connect with Asterisk from a web page. Regards ___ 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
Re: [Asterisk-Users] IAX Softphone
Hi, http://www.cosmica.com/dante/diax/diax099i.zip Sorry... the correct address is: http://www.cosmica.ro/dante/diax/diax099i.zip Best rregards, Dan ___ 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] TDM400 - channel out to lunch?
On Wed, Jan 26, 2005 at 06:11:31AM -0600, Rich Adamson wrote: For those of us that have had probems with the tdm dropping, it seems stopping *, stop and restart zaptel, restart * fixes what seems to be a software bug. No reboot necessary. If that doesn't fix the problem, then you might have a defective module. I found that I need to unload and reload the wcfxs module from the kernel and re-run ztcfg. Perhaps the former is no necessary, but it's in my script now. There was an issue with the first tdm cards shipped (ver e/f) where the first module slot had a problem. Those that received replacement cards found an added jumper wire on them suggesting a printed circuit board trace had been missed (or something like that). I have been having trouble with E/Fs (the H seems to be more stable), but it's not just with the first module. In my case it is the second one. And I initially had trouble because the FXO was on socket 1. Digium had me move it to socket 4 and that helped some. But only for a time. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk
Hello I got the similar error while trying a call. -- Executing Answer(SIP/8000104-86ef, ) in new stack -- Executing Wait(SIP/8000104-86ef, 2) in new stack -- Executing AGI(SIP/8000104-86ef, areskicc.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php: 'agi_request' = 'areskicc.php' areskicc.php: 'agi_channel' = 'SIP/8000104-86ef' areskicc.php: 'agi_language' = 'en' areskicc.php: 'agi_type' = 'SIP' areskicc.php: 'agi_uniqueid' = '1106824539.3' areskicc.php: 'agi_callerid' = 'DTA-310 8000104' areskicc.php: 'agi_dnid' = '002379511272' areskicc.php: 'agi_rdnis' = 'unknown' areskicc.php: 'agi_context' = 'prepaid' areskicc.php: 'agi_extension' = '002379511272' areskicc.php: 'agi_priority' = '3' areskicc.php: 'agi_enhanced' = '0.0' areskicc.php: 'agi_accountcode' = '' areskicc.php: areskicc.php: ANSWER areskicc.php: string(56) DTA-310 8000104 ; SIP/8000104-86ef ; 1106824539.3 ; n -- AGI Script areskicc.php completed, returning 0 -- Executing Wait(SIP/8000104-86ef, 2) in new stack -- Executing Hangup(SIP/8000104-86ef, ) in new stack == Spawn extension (prepaid, 002379511272, 5) exited non-zero on 'SIP/8000104-86ef' Need some help. Thanks Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: jeudi 27 janvier 2005 11:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk Hi Alex, Concerning the web interface, in this version we need the register_globals = On I will try to change it in the next release... To find out the error on the agi, can you run the agi script manually. php areskicc.php You will get more details about the error! Regards, Areski On Thu, 2005-01-27 at 03:07, Alexander Romanov wrote: Hi, I've tried it and could not get to work any of them (webapp and agi). On webapp I do not get a full menu, just logout and disconnect With agi nothing happens when I execute the script. -- Executing Answer(SIP/2204-6221, ) in new stack -- Executing Wait(SIP/2204-6221, 2) in new stack -- Executing AGI(SIP/2204-6221, areskicc.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php -- AGI Script areskicc.php completed, returning 0 -- Executing Wait(SIP/2204-6221, 2) in new stack -- Executing Hangup(SIP/2204-6221, ) in new stack == Spawn extension (local, 40, 5) exited non-zero on 'SIP/2204-6221' I have followed instructions to the letter. Am I missing something? Alex. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Thursday, 27 January 2005 4:05 AM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc.. * take care of multiple calls using the same Cardnumber * Caller gets informed about his credit Announce the remaining credit * Caller is requested to enter a destination number * Announce the maximal call time for the given destination number It calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout * Interupt the call if the card balance gets zero Warn the caller about the call interupt 60 30 seconds before the call gets interupted * It connects the Caller to the destination through the configured trunk note : different trunks can be configured and associated by prefix * After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit FEATURES - WEB INTERFACE: * CARD/CUSTOMERS * List customers * Refill customer * CARD/CUSTOMERS * List customers/cards * Refill customer/card * Create customer/card * Generate customers/cards * BILLING * View money situation * View Payment * Add new Payment * RATECARD * List Tariffplan * Create new Tariffplan * Define Tariffplan * TRUNK * List Trunk * Add
Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
just finish it if anyone like mysql go for it or someone love postgresql its ok but don't ruin the purpose of this list keep out these kind of mess sorry areski for that and thanks for your great work ___ 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] Com-on-Air - DECT card
I've received my Com-on-Air DECT card this morning and while I have configured it (both the DECT registration and SIP to Asterisk configuration), I am running into a Call missing Call ID from error message. I can call from a DECT handset to an extension, but not the reverse. Not being an expert, it seems to me that Asterisk is looking for a CallerID from the DECT card. I *believe* I've set everything up correctly, but if anyone has installed this solution and has an example sip.conf file I would be most grateful for a copy. Many thanks, George ___ 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] Re: TDM400 - channel out to lunch?
I have been having trouble with E/Fs (the H seems to be more stable), but it's not just with the first module. In my case it is the second one. And I initially had trouble because the FXO was on socket 1. Digium had me move it to socket 4 and that helped some. But only for a time. I'm having the same problem, random power alert on random module. Sometimes it freezes up and I have to reboot the machine. Maybe it is just a power problem. PSU or something similar. I have connected a fan on the same cable where the TDM400P is connected. I will try to remove it Sergio ___ 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] TDM400 - channel out to lunch?
For those of us that have had probems with the tdm dropping, it seems stopping *, stop and restart zaptel, restart * fixes what seems to be a software bug. No reboot necessary. If that doesn't fix the problem, then you might have a defective module. I found that I need to unload and reload the wcfxs module from the kernel and re-run ztcfg. Perhaps the former is no necessary, but it's in my script now. There was an issue with the first tdm cards shipped (ver e/f) where the first module slot had a problem. Those that received replacement cards found an added jumper wire on them suggesting a printed circuit board trace had been missed (or something like that). I have been having trouble with E/Fs (the H seems to be more stable), but it's not just with the first module. In my case it is the second one. And I initially had trouble because the FXO was on socket 1. Digium had me move it to socket 4 and that helped some. But only for a time. Depending on which distro your using, doing 'service zaptel stop' and 'service zaptel start' handles all of the required driver restarts. Usually when my tdm fails, all four ports fail at the same time. ___ 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] Grandstream setup woe and solution
Just added a new Grandstream BT102 to my network. Its running new firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to (SIP) register Gripe 1: The New Firmware does NOT show the current version of all the firmware. You have to ask the phone manually with its menu button. Gripe 2: It does not show '' in the the two password fields... This is what caught me - I had two browser (tabbed) sessions and was switching between them - looking for differences... obvious the password fields now being blank look the same.. I never typed in the Authenticate Password: Doing so fixed the problem. If anyone from Grandstream lurks - can they change this behaviour? - at least fake some '***' in the password fields... Asterisk also had me chasing my tail - it never mentioned anything such as 'SIP Registration password is incorrect' - I got one.. chan_sip.c:7231 handle_request: Failed to authenticate user Phone Five sip:[EMAIL PROTECTED];user=phone;tag=fjhgkjhgkhjlk (OK - failed authentication - but something about the password would have been better) and got lots of... chan_sip.c:7588 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.0.126' ... which had me greping around for the word phone (should this have not been phone5 ??) -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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@home and TDM400P cards...
Guys, Anyone know if the default [EMAIL PROTECTED] install supports TDM400P cards at all (Digium Fxo/Fxs port card) ?? Thx Paul ___ 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] Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi,
well, most of the things work right now due to the help of peter
svensson, but after heavy use of our ericsson BP250 today several
problems appeared.
i split into several mails as they are seperate problems.
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten = _.,102,Busy() with
no effect. this is the part of the extensions.conf i'm using:
[pri-external]
exten = _5678.,1,SetCIDNum(0${CALLERIDNUM}) ; Add a leading zero
exten = _5678.,2,Goto(${EXTEN:4}|1) ; Strip trunk digits from the DDI
exten = h,HangUp()
include = durchwahl
include = pri-external-route
[pri-external-route]
exten = _.,1,Dial(Zap/g3/${EXTEN})
exten = _.,2,Hangup()
exten = _.,102,Busy()
this is a excerpt from /var/log/asterisk/full
a call from a mobile phone (017212345678) to extension 134 which is busy:
-- Starting simple switch on 'Zap/35-1'
-- Executing SetCIDNum(Zap/35-1, 017212345678) in new stack
-- Executing Goto(Zap/35-1, 134|1) in new stack
-- Goto (pri-external,134,1)
-- Executing Dial(Zap/35-1, Zap/g3/134) in new stack
-- Called g3/134
-- Zap/38-1 is making progress passing it to Zap/35-1
Requested indication 14 on channel Zap/35-1
Received AST_CONTROL_PROGRESS on Zap/35-1
Dunno what to do with control type 15
-- Zap/38-1 is busy
Set option AUDIO MODE, value: ON(1) on Zap/38-1
Hangup: channel: 38 index = 0, normal = 63, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
disabled echo cancellation on channel 38
Set option TDD MODE, value: OFF(0) on Zap/38-1
Updated conferencing on 38, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/38-1
disabled echo cancellation on channel 38
-- Hungup 'Zap/38-1'
== Everyone is busy/congested at this time (1:1/0/0)
Exiting with DIALSTATUS=BUSY.
-- Executing Busy(Zap/35-1, ) in new stack
Requested indication 5 on channel Zap/35-1
== Spawn extension (pri-external, 134, 102) exited non-zero on 'Zap/35-1'
-- Executing Dial(Zap/35-1, Zap/g3/h) in new stack
-- Called g3/h
Set option AUDIO MODE, value: ON(1) on Zap/38-1
Hangup: channel: 38 index = 0, normal = 63, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
disabled echo cancellation on channel 38
Set option TDD MODE, value: OFF(0) on Zap/38-1
Updated conferencing on 38, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/38-1
disabled echo cancellation on channel 38
-- Hungup 'Zap/38-1'
Exiting with DIALSTATUS=CANCEL.
== Spawn extension (pri-external, h, 1) exited non-zero on 'Zap/35-1'
Set option AUDIO MODE, value: ON(1) on Zap/35-1
Hangup: channel: 35 index = 0, normal = 60, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
disabled echo cancellation on channel 35
Set option TDD MODE, value: OFF(0) on Zap/35-1
Updated conferencing on 35, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/35-1
disabled echo cancellation on channel 35
-- Hungup 'Zap/35-1'
regards
frank sautter
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[Asterisk-Users] Channel Restart - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * from time to time (sometime within a few minutes sometime after hours) a complete PRI line or several PRI lines are kind of resetting (none of my colleagues reported a call interruption though). could this be a problem of the length (around 4kilometres) of the line between the telco switch and the NT providing the E1-PRI? The PRI line itself is only 3 metres long. is this the line build-out parameter in /etc/zaptel.conf? or is this something with timing of the span? my current settings are: # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # TE405P/TE410P quad E1 span=2,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 span=3,0,0,ccs,hdb3,crc4 bchan=36-50,52-66 dchan=51 span=4,2,0,ccs,hdb3,crc4 bchan=67-81,83-97 dchan=82 span=5,0,0,ccs,hdb3,crc4 bchan=98-112,114-128 dchan=113 this is a excerpt from /var/log/asterisk/full -- B-channel 0/1 successfully restarted on span 2 -- B-channel 0/3 successfully restarted on span 2 -- B-channel 0/5 successfully restarted on span 2 -- B-channel 0/6 successfully restarted on span 2 -- B-channel 0/7 successfully restarted on span 2 -- B-channel 0/8 successfully restarted on span 2 -- B-channel 0/9 successfully restarted on span 2 -- B-channel 0/10 successfully restarted on span 2 -- B-channel 0/11 successfully restarted on span 2 -- B-channel 0/12 successfully restarted on span 2 -- B-channel 0/13 successfully restarted on span 2 -- B-channel 0/14 successfully restarted on span 2 -- B-channel 0/17 successfully restarted on span 2 -- B-channel 0/18 successfully restarted on span 2 -- B-channel 0/19 successfully restarted on span 2 -- B-channel 0/20 successfully restarted on span 2 -- B-channel 0/21 successfully restarted on span 2 -- B-channel 0/22 successfully restarted on span 2 -- B-channel 0/23 successfully restarted on span 2 -- B-channel 0/24 successfully restarted on span 2 -- B-channel 0/25 successfully restarted on span 2 -- B-channel 0/26 successfully restarted on span 2 -- B-channel 0/27 successfully restarted on span 2 -- B-channel 0/28 successfully restarted on span 2 -- B-channel 0/29 successfully restarted on span 2 -- B-channel 0/30 successfully restarted on span 2 -- B-channel 0/31 successfully restarted on span 2 regards frank sautter ___ 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] analog fax on ericsson BP250 - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * some faxes from our analog fax-machine on our ericsson BP250 do not get through or only after several tries. regards frank sautter ___ 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] Re: TDM400 - channel out to lunch?
I have been having trouble with E/Fs (the H seems to be more stable), but it's not just with the first module. In my case it is the second one. And I initially had trouble because the FXO was on socket 1. Digium had me move it to socket 4 and that helped some. But only for a time. I'm having the same problem, random power alert on random module. Sometimes it freezes up and I have to reboot the machine. Maybe it is just a power problem. PSU or something similar. I have connected a fan on the same cable where the TDM400P is connected. I will try to remove it Since several of the folks using the iaxy box (which uses the same fxs module) have complained about early failures, over heating, etc, might there be an overheating issue going on with the tdm card? I just felt the four fxo modules on my tdm card and all are running very cool. However, just simply touching the modules caused multiple ports to believe an incoming call was present. Repeated the process by touching them again, and wow, multiple ports started ringing phones again. If a board design is that sensitive to touching, it certainly implies a design problem. Twenty-plus years of doing electronic repair/diagnostic work says that is no where near normal. Very very very sensitive to the injection of electrical noise. I'm going to do a bunch more testing this weekend with that approach and try some decoupling caps, etc, to see if the card stablizes. No wonder the digium support folks are having problems nailing down the stability issue. ___ 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] Channel Restart - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
Hi, restarting the B channels is a normal process on PRIs. Nothing to worry about as long only idle B channels are restarted. best regards Klaus Am Donnerstag, den 27.01.2005, 13:10 +0100 schrieb Frank Sautter: hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * from time to time (sometime within a few minutes sometime after hours) a complete PRI line or several PRI lines are kind of resetting (none of my colleagues reported a call interruption though). could this be a problem of the length (around 4kilometres) of the line between the telco switch and the NT providing the E1-PRI? The PRI line itself is only 3 metres long. is this the line build-out parameter in /etc/zaptel.conf? or is this something with timing of the span? my current settings are: # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # TE405P/TE410P quad E1 span=2,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 span=3,0,0,ccs,hdb3,crc4 bchan=36-50,52-66 dchan=51 span=4,2,0,ccs,hdb3,crc4 bchan=67-81,83-97 dchan=82 span=5,0,0,ccs,hdb3,crc4 bchan=98-112,114-128 dchan=113 this is a excerpt from /var/log/asterisk/full -- B-channel 0/1 successfully restarted on span 2 -- B-channel 0/3 successfully restarted on span 2 -- B-channel 0/5 successfully restarted on span 2 -- B-channel 0/6 successfully restarted on span 2 -- B-channel 0/7 successfully restarted on span 2 -- B-channel 0/8 successfully restarted on span 2 -- B-channel 0/9 successfully restarted on span 2 -- B-channel 0/10 successfully restarted on span 2 -- B-channel 0/11 successfully restarted on span 2 -- B-channel 0/12 successfully restarted on span 2 -- B-channel 0/13 successfully restarted on span 2 -- B-channel 0/14 successfully restarted on span 2 -- B-channel 0/17 successfully restarted on span 2 -- B-channel 0/18 successfully restarted on span 2 -- B-channel 0/19 successfully restarted on span 2 -- B-channel 0/20 successfully restarted on span 2 -- B-channel 0/21 successfully restarted on span 2 -- B-channel 0/22 successfully restarted on span 2 -- B-channel 0/23 successfully restarted on span 2 -- B-channel 0/24 successfully restarted on span 2 -- B-channel 0/25 successfully restarted on span 2 -- B-channel 0/26 successfully restarted on span 2 -- B-channel 0/27 successfully restarted on span 2 -- B-channel 0/28 successfully restarted on span 2 -- B-channel 0/29 successfully restarted on span 2 -- B-channel 0/30 successfully restarted on span 2 -- B-channel 0/31 successfully restarted on span 2 regards frank sautter ___ 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
Re: [Asterisk-Users] Asterisk chooses invalid outgoing interface (IAX2, virtual interfaces)
i have an asterisk box with two external ip addresses (say 1.2.3.4 on eth0 and 1.2.3.5 on eth0:1) and one internal vpn ip address (say 10.0.0.1 on tun0) The problem is: when a client conntects via iax to 1.2.3.4, the asterisk server responds on 1.2.3.5 and the packets are not accepted by the client. I can kinda fix this by using the bindaddr option, but then i can't accept calls on tun0. IMHO the asterisk responses should always be sent from the ip the request was received on. Any ideas on how to fix this? Its a misunderstanding of how packets are handled on your system. Packets are _not_ sent on an interface just because it arrived on that interface. The packets are sent based on your routing table (netstat -rn) entries. If your system does not specifically know how to reach the destination, the packet is sent on the default route. Try adding route statements to your system to define each network accessible via each interface, and ensure a single default route points to your catch-all (eg, Internet) interface. ___ 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] Re: Channel Restart - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
On Thu, 27 Jan 2005, Frank Sautter wrote: well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * from time to time (sometime within a few minutes sometime after hours) a complete PRI line or several PRI lines are kind of resetting (none of my colleagues reported a call interruption though). could this be a problem of the length (around 4kilometres) of the line between the telco switch and the NT providing the E1-PRI? The PRI line itself is only 3 metres long. is this the line build-out parameter in /etc/zaptel.conf? or is this something with timing of the span? Every hour (or possibly two hours) asterisk sends a reset command for all channels it considers idle. This is normally harmless and prevents the two ends from getting out of sync. On some remote ends (PAnasonic PRIs e.g.) due to bugs this can be a problem. If there has been no reports of dropped calls there should be no need to worry. Peter ___ 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] SoftClient for Pocket PC
Hi List, Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens)an register itwith asterisk? any suggestions? thx in advance.__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] IAX Softphone
Diax supports showing the number of new and old voicemail messages although I have not managed to get that working yet. There is a parameter in iax.conf which must be enabled in order to get this functionality. ; If mailboxdetail is set to yes, the user receives ; the actual new/old message counts, not just a yes/no ; as to whether they have messages. this can be set on ; a per-peer basis as well ; mailboxdetail=yes For the account you must have a line mailbox=xxx This is all you need to make it work. I have 'mailboxdetail=yes' in my iax.conf file. A typical extension configuration is :- [7000] type=friend regexten=7000 secret=password host=dynamic context=voipuk mailbox=7000 In voicemail.conf I have entries such as :- [local] 7000 = 1234,Users Name,[EMAIL PROTECTED] The voicemail system works but I dont get any new messages indicated with Diax. Can you see if I have done anything wrong? P.S I have a suggestion. It would be nice to be able to reconfigure the 'transfer' button as I believe currently it is just a shortcut for '#' and I intend to use a different code (** for example) for transfers to avoid problems with it interfering with some navigation systems. Regards Gareth ___ 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] IAX Softphone
Em Qui 27 Jan 2005 05:18, Dan escreveu: Hi Denis, - Original Message - From: Denis Galvão - iSolve [EMAIL PROTECTED] Hey I tried DIAX today and the speech quality was rather poor compared to X-lite. Dan, do you know wich iaxclient version firefly is build on!? I got better results(voice quality) using firefly, doesn't matter what CODEC I used. I don't know which library firefly uses. Can you describe in more detail the difference regarding voice quality? I mean... more distorted, drop-outs, tone, level, etc...? With Firefly I got better volume and the voice is more polished, I mean, with DIAX I got more noise. This is my expirience, I tried a lot of softphones in different computers, Firefly win the contest, but I think DIAX is the better of all in features! Like I told you before, I really want to use DIAX! P.S.: Someone forgot to say that DIAX supports USB Phones with /u flag too! For it is great Regards, Denis. ___ 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] Re: Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
On Thu, 27 Jan 2005, Frank Sautter wrote:
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten = _.,102,Busy() with
no effect. this is the part of the extensions.conf i'm using:
[pri-external]
exten = _5678.,1,SetCIDNum(0${CALLERIDNUM}) ; Add a leading zero
exten = _5678.,2,Goto(${EXTEN:4}|1) ; Strip trunk digits from the DDI
exten = h,HangUp()
include = durchwahl
include = pri-external-route
[pri-external-route]
exten = _.,1,Dial(Zap/g3/${EXTEN})
exten = _.,2,Hangup()
exten = _.,102,Busy()
The extension _. will match the 'h' context as well. You need to use
_X. or rearrange your hangup context into a separate context that is
included before the pri-external-route context in pri-external. The order
of evaluation is only well defined for included contexts.
Have you set priindication=outofband in zapata.conf? From the log it seems
that way. You have to decide if you want audio notification or isdn
notification of the busy condition. You may want to set the variable
PRI_CAUSE=17 prior to hangup to explicitly send the busy cause code over
isdn.
Peter
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[Asterisk-Users] analog lines via channel bank --
morn, I am looking into a situation where I need 50 or so analog extentions, all of them need to have caller ID. Anyone have any recommendations for channel banks and or tips or warnings on Caller ID to the analog stations. thanks mike -- |- - - - - - - - - - - - - - - - - - - -| |-Mike Deweyof -| |= All Technologies Unlimited, Inc =| |- phone: 303.667.0357 -| |- e-mail: [EMAIL PROTECTED] -| ___ 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] Re: analog fax on ericsson BP250 - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
On Thu, 27 Jan 2005, Frank Sautter wrote: well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * some faxes from our analog fax-machine on our ericsson BP250 do not get through or only after several tries. Is your timing correct? Do you have your pstn interface as the primary timing source on asterisk? How is the asterisk box connected to the pstn and the BP250? Through a single TE405P/TE410P? More data makes it easier to pinpoint the problems. Other problem sources include mismatched alaw/mulaw, interrupt problems on the machine etc. Can you describe the machine a bit? Peter ___ 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] Multi Asterisk Server Transfers
On Wed, 26 Jan 2005 22:42:52 -0800, Luki wrote: if it were my project, I'd look into Asterisk on a small form factor/embedded system like a Soekris Engineering box In that case I guess you could use a Linksys WRT54G instead and run * on it :-). Comes fully assembled (read: with a case) and a 4-port switch to connect the phone and a computer... and can probably even do QOS to prioritize voice traffic. Never mind the WiFi part, but might be handy as well. Someone asked WHY you would want to run * on a simple WRT54G, I guess this is a possible scenario. Perhaps. I personally don't like the idea of using the WRT54G. I've been burned by Linksys in the past. I run m0n0wall on Soekris 4501. The hardware with case cost me $200. I understand that WRAP boards are less than Soekris. You could, as an alternative, buy one of the new routers with SIP ATA capability and have it log into the ITSP and head office *. I'm not sure if they support multiple registries. But again, I'd prefer a real SIP phone with business class features. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] Asterisk QSIG
Hi everybody, Can somebody help me with real samples implemented with Asterisk as PBX system and PRI using Q.SIG protocol. Thx in advance, Radu Padure On Fri, 2005-01-21 at 09:36 -0600, [EMAIL PROTECTED] wrote: On Fri, Jan 21, 2005 at 12:24:32AM +0100, Marco Vescovi wrote: 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 ... Yeah, check my development tree from CVS to play with Q.SIG features. It is definitely not a full implementation, but I have a few basic features (MWI activate/deactivate, DivertingLegInformation2 receive, callingname, etc) implemented. You can (of course) do basic things like passing calls as well. Matthew Fredrickson ___ 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
Re: [Asterisk-Users] Asterisk chooses invalid outgoing interface (IAX2, virtual interfaces)
Peter Svensson wrote: I think the complaint is that asterisk does not use the destination address for the incoming request packet as the source address for the outgoing packet holding the reply. This will prevent the requestor from matching the quadtuple (src addr, dst addr, src port, dst port) which is usually used to identify a conversation. Tcp handles this transparently but for udp it is the responsibility of the application. yes thats exactly the problem. it uses a different source address in the reply than the destination address of the original packet and that causes some clients to discard the packet. stefan ___ 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] Multi Asterisk Server Transfers
Mike Sander wrote: Is this possible? Companies with multiple * servers in many remote office, surely have this system, to conserve bandwidth? How is the transfer made? Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] basic release. Simple way is to use the # transfers in asterisk on the main box then don't allow transfers on the remote boxes and don't use transfer buttons... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than myself. ___ 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] HEELP!! with Eyebeam
Hello Everyone I just bough my Xten Eyebeam but i don`t figure out how to make the video works i only see a black screen where the remote video suposse to appear, Any help regarding this matter will be very preciated Thank You ___ 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] Voice mail
HI I would like know if it's possible to use the VoiceMail only of the Asterisk Sytem without use the PBX part ? Thank. ___ 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] SoftClient for Pocket PC
Hi Richard There is a version of SJPhone (http://www.sjlabs.com/sjp.html) for PocketPC. Works ok on the Ipaq 2210, haven't tried it on anything else. Cheers, Edge. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of richard Coco Sent: Thursday, January 27, 2005 2:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SoftClient for Pocket PC Hi List, Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens) an register it with asterisk? any suggestions? thx in advance. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] Re: UPS for Asterisk
-Original Message- From: Michael Loftis [mailto:[EMAIL PROTECTED] Not old, just small it seems. The little Norstar (merlin?) Nortel's do NVRAM/Flash, as do Panasonic's. There's also the App/VM Module which is an OS/2 based system, or was. Toshiba Strata systems also use NVRAM to hold their configuration. They pitch this as a selling point -- no moving parts. I think their integrated voicemail system uses a hard disk, though. ___ 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] IAX Softphone
Hi, I have 'mailboxdetail=yes' in my iax.conf file. A typical extension configuration is :- [7000] type=friend regexten=7000 secret=password host=dynamic context=voipuk mailbox=7000 In voicemail.conf I have entries such as :- [local] 7000 = 1234,Users Name,[EMAIL PROTECTED] The voicemail system works but I dont get any new messages indicated with Diax. Can you see if I have done anything wrong? Strange... Which is your Asterisk version? It works with the CVS from september 2004, which is used by me now.. For older version there is a modification described in diax help file in order to get this functionality. P.S I have a suggestion. It would be nice to be able to reconfigure the 'transfer' button as I believe currently it is just a shortcut for '#' and I intend to use a different code (** for example) for transfers to avoid problems with it interfering with some navigation systems. It is not a shortcut. '#' uses the Standard asterisk transfer function and Transfer button an internal implemented unattended IAX transfer function... I'll take into consideration to make a configurable shortcut for the Transfer button. Best regards, Dan ___ 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] Re: Polycom and call waiting again...
Message: 10 Date: Wed, 26 Jan 2005 17:53:39 -0500 (EST) From: Sean A. Newton [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again.. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII On Wed, 26 Jan 2005, Noah Miller wrote: Have you tried adding SetGroup(), and CheckGroup() functions to the dialplan that rings the phone? It maybe something to try. I think the problem is that these functions only work from the dialplan. In this case, Sean is trying to get calls from a Queue (and not the dialplan) to the correct line on the phone. I was thinking about implementing a queue for our receptionists, but this problem prevents me from doing that, and I haven't figured out any way around it. Maybe the new 1.4.1 firmware provides a way to disable that horrid call-waiting feature? Has anybody gotten it to run successfully? I have a number of queues which ring to dedicated call appearances, if that's what you're trying to do. In my SIP config, I have: (sorry about capitalization... For some unknown reason, we had to standardize on M$ Outlook... *sigh*) [1234] Type=friend Context=whatever Host=dynamic Secret=password1234 Dtmfmode=inband Disallow=all Allow=ulaw [1234b] Type=friend Context=whatever Secret=password1234b Dtmfmode=inband Disallow=all Allow=ulaw Outgoinglimit=1 . . . Rinse, lather, and repeat for each queue you want on a phone, or as many call appearances as you have. Since we have IP600s, and nobody is in more than 5 queues currently, it works well for us. We avoid the call waiting issue using the outgoinglimit=1 directive, as the Asterisk server will only send one call to the phone at a time. I know that it is supposedly going away soon, but it's working right now. I just statically define the queues to have the appropriate call appearances like Member = SIP/1234b Then, in the phone1234.cfg file, I set each appearance to be 1234, 1234b, 1234c, etc. The problem with this is that each IP600 adds 80 lines to the sip.conf file, and each time we add queue members, I have to modify the queues.conf file. But it works for our needs. Exactly.. SetGroup was suggested by someone on the irc channel.. I looked at it briefly. I was then shot down by someone saying to save my effort, it didn't work. I suspected as much, due to the fact that the Queue function doesn't use the exten config for that phone. And it shouldn't.. The phone should be able to take care of this problem.. Yeah, I didn't think it would work, so I never went down that road either. I've unfortunately got myself into a bind because I've bought ~35 of these phones. :eek: Well, if you just can't use them, I could send you my address ;) If everyone thinks SetGroup and CheckGroup will work, I will spend the next days working with it, but I don't want to go barking up the tree of something that doesn't look like it will work. :| I'm also interested to try out the 1.4.1 firmware. Just need to procure a copy of it.. The 1.4.1 firmware is available now from a website that escapes me, but is linked from the WIKI. I've been testing it for about 12 hours, and so far so good :) --Sean Hope this helps, David Gomillion ___ 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] IAX Softphone
Hi, With Firefly I got better volume and the voice is more polished, I mean, with DIAX I got more noise. Pls play with the microphone level /mic boost and AGC. Microphone level will be adjustable from the interface in the next version. Like I told you before, I really want to use DIAX! Hope to solve that delay issue soon. P.S.: Someone forgot to say that DIAX supports USB Phones with /u flag too! In the 0.9.10a version will support the Yealink USB phone too, including the display, selectable ring tones, etc. Best regards, Dan ___ 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] Re: phone rings when I'm using it over VOIP - WHY?
On Thu, 2005-01-27 at 08:40 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Joseph [EMAIL PROTECTED] wrote:
When I use my phone to make VOIP call and another calls comes from POTS
my phone rings to POTS caller. Why?
Shouldn't it generate busy signal!
Yes, but there are all sorts of configuration errors that could result
in the behaviour described. Without knowing your particular setup, it
is impossible to know what the cause could be. Perhaps you could
describe in more detail.
My setup is really simple.
I have Sipura-3000 connected to * with phone1 and another SIP phone2.
Here is my context:
exten = 1,1,Dial(${phone1},20,tr)
exten = 1,102,Dial(${phone2},20,tr)
I have setup two phones and have VOIP, when I make call over VOIP I
think channel return status -1 (the call is bridged). So when a call
comes from POTS my phone1 keeps ringing and I want to ring phone2 not
mine.
If the channel return status 0 the call is transfered to priority n+1
and that is what I want.
Why priority is 0 when I pickup the phone and hear dial tone (without
calling out); and priority is -1 when call is connected bridged with
another party?
To my understanding in both cases the phone1 is busy so why return
different priority code???
--
#Joseph
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Re: [Asterisk-Users] Re: Polycom Phones
On Wed, Jan 26, 2005 at 10:20:24PM -0500, Cory Andrews said: Seshu - the 480i, although a great phones, is quite a bit more expensive than the Polycom IP300 or IP500, it is more comparable in price to the Polycom IP600. Hmm. Your own web site has it priced between the 500 and 600. If the difference is good support versus zero support, wouldn't the $50 difference between the 500 and the 480i be saved in the first 20 minutes you spend fighting with a problem? Another factor is that one company tests with * and the other shuns it. Just the availability of the firmware alone is almost worth the $50. I have no problem with polycom, and use their non-IP conference phones, but I'm not going to purchase a product from a manufacturer that refuses to provide even basic support (complete manuals and firmware.) It would be Very nice to have a phone platform that is fully documented that had firmware that was open and hackable. It seems that people on this list spend massive amounts of time trying to work around all the firmware bugs in various products (eg. call waiting on polycom.) If sayson provided developer documentation for their phones and allowed us to write our own firmware, they wouldn't be able to manufacturer them fast enough. They would corner the IP phone market. ___ 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] Re: TDM400 - channel out to lunch?
If a board design is that sensitive to touching, it certainly implies a design problem. Twenty-plus years of doing electronic repair/diagnostic work says that is no where near normal. Very very very sensitive to the injection of electrical noise. I'm going to do a bunch more testing this weekend with that approach and try some decoupling caps, etc, to see if the card stablizes. No wonder the digium support folks are having problems nailing down the stability issue. Not to mention the CPU spikes every n seconds. Rich, while you're testing, would you keep an eye on 'vmstst 1' and the 'system' (not user) CPU utilization? Thanks, Mike ___ 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] SoftClient for Pocket PC
Edge Bisset wrote: Hi Richard There is a version of SJPhone (http://www.sjlabs.com/sjp.html) for PocketPC. Works ok on the Ipaq 2210, haven't tried it on anything else. This client works fine with my Asus MyPal716 as well. -- Robert P. McKenzie | GammaRay Technical Services Ltd [EMAIL PROTECTED] | [EMAIL PROTECTED] http://www.uk-experience.com | http://www.gammaray-tech.com Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014 ___ 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] softphone headsets
You can use pretty much any headset you want. I use just a regular, inexpensive Labtec one from WalMart for now. Works fine. As for softphones... I just tried out SJphone yesterday and I like it more. I only tested it out briefly on my work Windows XP system, and haven't tried it at home on Linux yet. -- Dana - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 11:23 PM Subject: [Asterisk-Users] softphone headsets Anybody have a suggestion for a nice inexpensive headset for mobile users on a laptop with a softphone? What are people using for softphones on M$ platforms? I have been using the x-lite client. Is there something better out there? -- http://www.umich2.com ___ 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] ZAPHFC problem
Hi all, I have a cheap ISDN card with zaphfc module in TE mode (ptmp). Everything is going fine, I can make call from SIP phones to ISDN but sometimes while I'm speaking (SIP - ISDN or ISDN -SIP) the line just break. After that I can't dial or receive calls. Only after reloading zaphfc module I can make and receive calls. Anybody know what's happening Thank you ___ 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] R2 in Bolivia
These are log of incoming calls Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 MFC/R2 call control(1) Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 MFC/R2 make call Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Making a new call with CRN 32769 Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Tx bits 0x1 [1/ 1/ 0/ 0] -- Called g3/70513933 Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event Dialing - 0x9841460 Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Rx bits 0xD [1/ 40/201/ 0] Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Tx tone 7 on [2/ 40/202/ 0] Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Rx tone 5 on [2/ 40/202/201] Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Tx tone 7 off [2/ 40/202/201] Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Rx tone 5 off [2/ 40/202/201] Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Tx tone E on [2/ 40/202/201] Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Rx tone 5 on [2/ 40/202/207] Jan 26 18:17:50 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:50 mfcr2 Tx tone E off [2/ 40/202/207] Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Rx tone 5 off [2/ 40/202/207] Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Tx tone E on [2/ 40/202/207] Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Rx tone 4 on [2/ 40/202/207] Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Tx tone E off [2/ 40/202/207] Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Rx tone 4 off [2/ 40/202/207] Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Far end disconnected - state 0x40 Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event Far end disconnected - 0x9841460 Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:3181 handle_uc_event: CRN 32769 - far disconnect cause 42 -- Channel 0 got hangup -- UniCall/1-1 is circuit-busy Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 MFC/R2 call control(6) Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 MFC/R2 drop call(cause=16) Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Clearing fwd Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Tx bits 0x9 [2/ 800/209/207] -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Rx bits 0x9 [1/ 800/211/ 0] Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Call disconnected - state 0x800 Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event Drop call - 0x9841460 Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 MFC/R2 call control(7) Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 MFC/R2 release call Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:703 unicall_error: UniCall: 2005/01/26 22:17:51 mfcr2 Destroying call with CRN 32769 Jan 26 18:17:51 WARNING[8573]: chan_unicall.c:2848 handle_uc_event: UC event Release call - 0x9841460 -- UC channel 1 released -- H.323 call 'ip$200.87.125.195:30008/25331' cleared, reason 7 (Remote user stopped calling) On Wed, 2005-01-26 at 07:53 +0800, Steve Underwood wrote: Hi Jorge, You might be the first person to try the Bolivian variant. I need more information to make any sense of the problem. In /etc/asterisk/unicall.conf add the line: loglevel = 1023 and try again. You should get a much more detailed log of what happens. Send that to me. Regards, Steve [EMAIL PROTECTED] wrote: Hi I made some tests with new MFC/R2 an unicall support for asterisk and now have dialing out problem using UniCall / R2. This is the error report in cli UC channel 30 protocol
[Asterisk-Users] /usr/bin/ld: cannot find -lidn
Bueller? Is this a lib of some kind? Google and lists bring up nada, this is from ast cvs head latest on Fedora Core 3. /usr/bin/ld: cannot find -lidn collect2: ld returned 1 exit status make[1]: *** [app_curl.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# uname -a Linux zoot 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686 i686 i386 GNU/Linux ___ 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] how to manage Digium TDM04B outgoing calls correctly
On January 27, 2005 11:06 am, Begumisa Gerald M wrote: On Thu, 20 Jan 2005, Andrew Kohlsmith wrote: I'm curious -- what is the motherboard you're doing this on? CPU? Oops sorry hadn't seen this - the specs are basically 2.8GHz CPU, 512MB RAM. I'm not sure what motherboard specs you want (its PCI 2.2 compliant of course) but its no special box. Just some clone we picked up for the equivalent of about USD400. It's got 6 PCI slots in total actually. We had to unplug power supply to the CDROM drive to get enough power for the FXS Cards, though. The reason I'm asking is because many of us have a lot of trouble with even two TDM cards in one machine... You've got six without any issue whatsoever and I'm trying to figure out your secret. :-) If you've got the exact motherboard make/model somewhere that would really help. -A. ___ 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] Dialogic Boards
if it is on Linux hardware with * you'll need to get your hands on the Linux drivers for your dialogic board which are not publicy accesible (or are they?). You must have it recognized by the linux system before doing anything to it by modprobe. On Wed, 26 Jan 2005 10:11:17 -0800, James Ellis [EMAIL PROTECTED] wrote: Hi All, I have checked the supported hardware list of boards that will work with Asterisk. What I am curious about is whether or not I need to have the Dialogic software installed and loaded before I launch Asterisk. Thanks. Jim ___ 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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ 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] Dialogic Boards
im sorry insmod On Thu, 27 Jan 2005 09:22:11 -0500, Erick Perez [EMAIL PROTECTED] wrote: if it is on Linux hardware with * you'll need to get your hands on the Linux drivers for your dialogic board which are not publicy accesible (or are they?). You must have it recognized by the linux system before doing anything to it by modprobe. On Wed, 26 Jan 2005 10:11:17 -0800, James Ellis [EMAIL PROTECTED] wrote: Hi All, I have checked the supported hardware list of boards that will work with Asterisk. What I am curious about is whether or not I need to have the Dialogic software installed and loaded before I launch Asterisk. Thanks. Jim ___ 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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ 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] grandstream budgetone-100 updates
I can't tell you why it's failing, as I don't know. But to answer your other question, I have firmware 1.0.5.22 that I found from a link a short time ago on the mailing list. I'm having some issues with DHCP and the BudgeTone phone, as it doesn't seem to like the TFTP options we put in. (I do have the Aastra 480i working properly now though). -- Dana - Original Message - From: dean collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 6:28 PM Subject: [Asterisk-Users] grandstream budgetone-100 updates I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.2025 Jan 18:25 Error Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg.txt Octet, Send 192.168.16.2025 Jan 18:25 Error Can anyone tell me why these fail each time? Also what is the latest revision? Cheers, Dean ___ 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] IAX Softphone
On Thu, 2005-01-27 at 13:56, Dan wrote: Hi, I have 'mailboxdetail=yes' in my iax.conf file. A typical extension configuration is :- [7000] type=friend regexten=7000 secret=password host=dynamic context=voipuk mailbox=7000 In voicemail.conf I have entries such as :- [local] 7000 = 1234,Users Name,[EMAIL PROTECTED] The voicemail system works but I dont get any new messages indicated with Diax. Can you see if I have done anything wrong? Strange... Which is your Asterisk version? It works with the CVS from september 2004, which is used by me now.. For older version there is a modification described in diax help file in order to get this functionality. I have just upgraded to the latest CVS version and am still experiencing the same problem. I can't get call transfers working so don't know if there is a common problem. ___ 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] Using ChanIsAvail with SIP
Were you able to make it to work ChanIsAvail application? I have a similar problem. -- #Joseph On Wed, 2004-12-15 at 14:49 -0800, voipbuilder wrote: Hello Everyone, I am trying to use the ChanIsAvail application but I am not getting the results I expect when making calls... exten = 100,1,ChanIsAvail(SIP/100SIP/200SIP/300) exten = 100,2,Cut(theChannel=AVAILCHAN,,1) I tested this by placing a call to extension 100, 100 answers. while that call is up, I make another call to extension 100, and I would expect 200 to ring, but 100 Rings again. And I have set the incomingcalllimit=1 any ideas? suggestions? ___ 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] Directory service of voicemail extensions
Hi, Does Asterisk support Directory service of voicemail extensions using database? If yes, how to configure asterisk? I know that it supports this feature using conf files. Thanks, Jagan ___ 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] /usr/bin/ld: cannot find -lidn
On Jan 27, 2005, at 15:12, Matt Schulte wrote: Bueller? Is this a lib of some kind? Google and lists bring up nada, this is from ast cvs head latest on Fedora Core 3. /usr/bin/ld: cannot find -lidn collect2: ld returned 1 exit status make[1]: *** [app_curl.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 On my Apple Cube that I use for Asterisk, yum info libidn shows this: Name : libidn Arch : ppc Version: 0.5.4 Release: 1 Size : 569.34 kB Group : System/Libraries Repo : Yellow Dog Linux 4.0 Base Summary: Internationalized Domain Name support library Description: GNU Libidn is an implementation of the Stringprep, Punycode and IDNA specifications defined by the IETF Internationalized Domain Names (IDN) working group, used for internationalized domain names. So you're probably missing the libidn and libidn-devel packages. jens ___ 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] Re: TDM400 - channel out to lunch?
If a board design is that sensitive to touching, it certainly implies a design problem. Twenty-plus years of doing electronic repair/diagnostic work says that is no where near normal. Very very very sensitive to the injection of electrical noise. I'm going to do a bunch more testing this weekend with that approach and try some decoupling caps, etc, to see if the card stablizes. No wonder the digium support folks are having problems nailing down the stability issue. Not to mention the CPU spikes every n seconds. Rich, while you're testing, would you keep an eye on 'vmstst 1' and the 'system' (not user) CPU utilization? That cpu spiking is another issue separate from the stability issue (I think). Not sure where the discussion of the spiking ended up a few weeks ago, do you remember? I've not tried to dig through the code, but it wouldn't surprise me if some temp code exists that might be polling the tdm card (or something like that) as an aid towards identifying the stability issue. Gut feeling suggests that if stability is truly related to tdm design problems (or whatever), then resolving that issue probably should be a precursor to chasing the cpu spikes. Any thoughts? ___ 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] Re: New Firefly version
Also sound quality seems to be poor using the ULAW codec. I am using: - latest Firefly on Windows XP SP2 - Asterisk 1.0.5 patched coupled with Bristuff-0.2.0-RC5 with Florz patch for zaphfc - Linux kernel 2.6.9-1.681_FC3 Fedora Core 3 (obviously) - connecting to FWD dialing 411 info service Any other codec is better and useable. Clearly it seems to be optimized for iLBC. ULAW is unusable for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hhandresen Sent: Thursday, January 27, 2005 11:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: New Firefly version Hi Adam, Sory to say it, bu it still interupt the mouse if you have microsoft wireless mouse/keayboard. The mouse jumps around on the screen. Any news on this ? /HHA Adam Hart wrote: As always, I'm happy to announce a new version of Firefly. Firefly 1.9.8 has more of what you want and less of what you don't http://www.virbiage.com/firefly/download/firefly-thirdparty.exe There's a few bug fixes - notably fixed the Reject button and sending of audio before answering in some circumstances. -Adam ___ 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 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] Least Cost Routing
Matthew, I'm trying to do Least Cost Routing for some International Rates between VoipJet and LiveVoIP. I saw your post about the data in mysql and a later post about the crashing, so that means you did figure out how to get the data into mysql? I compiled it and asterisk loaded it w/o crashing so so far so good, but I don't know how to get the data into mysql. Any help/insight you can provide would be helpful. Then there's this: http://www.voip-info.org/wiki-Application+LCDial Looks similar. Have you got LCR working? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, October 13, 2004 3:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Least Cost Routing Anyone using the rate_engine from TrollPhone? There is absolutly no documentation on how to setup data in the tables. If someone could send sample data, or post it to the wiki, it would be helpful. If any others are successfully using another Least Cost Routing method, please pass it along. THanks, Matthew ___ 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
RE: [Asterisk-Users] Am I missing something really basic here?????helpwith Asterisk@home {Scanned}
Ok, I thought the point of [EMAIL PROTECTED]
was that it automatically detected the X100P board and configured it correctly.
Is this incorrect? You still need to
modify /etc/zaptel files? And not just using the AMP configurator.
There is no mention of this on the
[EMAIL PROTECTED] webpage.
Can anyone who has actually used
[EMAIL PROTECTED] confirm this one way or the other?
Thanks,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Shaw
Sent: Thursday, January 27, 2005
9:28 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Am I
missing something really basic here?helpwith [EMAIL PROTECTED] {Scanned}
Yes, You need to add channels to your zapata.conf file.
zapata.conf
[channels]
;
; X100P plugged into PSTN
; X100P # 1
;[line1]
context=line1
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived
channel = 1
You might need to edit /etc/zaptel.conf
Check fxsks=1-4 I have four X100P cards.
If you have one change it to fxsks=1
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp
; Console interface for demo
IAXINFO=guest
; IAXtel username/password
TRUNKL1=Zap/1
TRUNKL2=Zap/2
TRUNKL3=Zap/3
TRUNKL4=Zap/4
; Trunk interface
TRUNKMSD=1
; MSD digits to strip (usually 1 or 0)
[line1]
exten = s,1,Dial(SIP/101,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,101
exten = s,5,Hangup
Here I have TRUNKL1=Zap/? for each X100P cards.
[line1] tells asterisk how to answer that line.
Remember I'm very new at this, but I didn't see anyone
respond to your post.
Goog luck, David
- Original Message -
From: dean collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, January
26, 2005 5:36 AM
Subject: [Asterisk-Users]
Am I missing something really basic here? helpwith [EMAIL PROTECTED] {Scanned}
Im trying to install [EMAIL PROTECTED],
Ive just downloaded the latest cd from soundforge. I can get it to
install ok (network card didnt auto configure but I worked out
how to use netconfig).
I worked out how to add a few grandstream budgetone fine.
Worked out how to upload music etc. Worked out how to modify FOP.
Voicemail and meetmes work fine.
HOWEVER.
Im using a X100p. I cant get it to make a call out or
use the default extension for an incoming line.
What do I need to make the pstn connection work? Do I need
to modify Zapata.conf? there are zero instructions on the [EMAIL PROTECTED] page as
to what to do.
Can anyone help me out here.
TIA,
Dean
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RE: [Asterisk-Users] TFTP Server Facing the Internet
TFTP is inherently insecure :-) This insecurity is how I got my BroadVoice SIP UID and Pass a long time ago before they supported Asterisk, told them the MAC of my Cisco phone and just grabbed the config file off their tftp server, interesting stuff. FireWall is your only true solution but that stops the phone from being able to be mobile. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Wednesday, January 26, 2005 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TFTP Server Facing the Internet Since we're chatting about tftp servers... Let's say I have a new customer with Cisco 79xx phones, and he desires to SIP register on my Asterisk system. I would have to provide the SIPmac.cnf and SIPDefault.cnf files on my tftp server for his phones. These files would be world readable, which I don't want. Is the solution to put the tftp server behind the firewall and port redirect based on the customer's IP, or is there a better way of restricting access? Thanks, Mike ___ 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
Re: [Asterisk-Users] /usr/bin/ld: cannot find -lidn
On Thursday 27 January 2005 14:12, Matt Schulte wrote: Bueller? Is this a lib of some kind? Google and lists bring up nada, this is from ast cvs head latest on Fedora Core 3. Google brings up many pages for libidn. The very first hit being where you can download it. B ___ 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] Voice mail
How would you deliver calls to the voicemail system without the PBX functions? db On Thu, 2005-01-27 at 08:22, [EMAIL PROTECTED] wrote: HI I would like know if it's possible to use the VoiceMail only of the Asterisk Sytem without use the PBX part ? Thank. ___ 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
Re: [Asterisk-Users] /usr/bin/ld: cannot find -lidn
Matt Schulte wrote:
Bueller? Is this a lib of some kind? Google and lists bring up nada,
this is from ast cvs head latest on Fedora Core 3.
/usr/bin/ld: cannot find -lidn
collect2: ld returned 1 exit status
make[1]: *** [app_curl.so] Error 1
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk]# uname -a
Linux zoot 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686 i686
i386 GNU/Linux
Looks like asterisk is using cURL:
CURLLIBS=$(shell curl-config --libs)
ifneq (${CURLLIBS},)
APPS+=app_curl.so
And cURL uses libidn:
http://curl.netmirror.org/libs.html
So you likely need:
http://mirrors.kernel.org/fedora/core/3/i386/os/Fedora/RPMS/libidn-0.5.6-1.i386.rpm
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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Re: [Asterisk-Users] /usr/bin/ld: cannot find -lidn
On Thu, Jan 27, 2005 at 03:37:10PM +0100, Jens Vagelpohl wrote: On my Apple Cube that I use for Asterisk, yum info libidn shows this: This answers a question I had but did not think would be answered yes. Which cube are you using? Is a G3 300 (old world) minitower fast enough for a small network? Thanks, Geoff. -- Geoffrey S. Mendelson, Jerusalem, Israel [EMAIL PROTECTED] N3OWJ/4X1GM IL Voice: 972-544-608-069 IL Fax: 972-2-648-1443 U.S. Voice: 1-215-821-1838 I may be an old fart, but I'm a high-tech, up to date old fart. :-) ___ 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] Bristuff ZapHFC and Loosing D-Channel
Hi! Did you ever find the answer to your question? I am getting the same message on the console every second: == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up etc. etc. etc. I'm running Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 The error is only visible however if I run * with -v (but I guess I shouldn't see these messages nonetheless)? On Tue, 25 Jan 2005, Peer Oliver Schmidt wrote: Using the latest(?) bristuff (Asterisk 1.0.4-BRIstuffed-0.2.0-RC3a) I have problems with loosing the D-channel. Most of the time, after the message PRI D-channel down it only takes a second or so to come back up, noted by the message PRI D-channel up However, today most of the time the D-channel stays down. Calls come in, but can't be answered. Does anyone know of a fix for this, or might have some insights on how to circumvent this problem? Any and all help is greatly appreciated. ___ 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] Tall free number via FWD over IXA2
The superdial macro in the wiki will go further an allow you to specify multiple methods to dial out (ie try FWD, then another voip provider, then Zap, etc) Joseph ([EMAIL PROTECTED]) wrote: On Wed, 2005-01-26 at 18:17 +1100, Duane wrote: Joseph wrote: Thanks Kris, I found the solution: Here is how it suppose to look like: You can minimise all that with a simple macro and a little pattern matching, and it makes dial plans so much easier to track down problems with etc... I couldn't find anything on it, but I'm not sure if you can or can't shorten it any further something like exten = _18[00,66,77,88]. etc, but it won't parse correctly in that form because it will think they are arguments, not part of the regexp... [tollfree] exten = _1800.,1,Macro(tollfree, $) exten = _1866.,1,Macro(tollfree, $) exten = _1877.,1,Macro(tollfree, $) exten = _1888.,1,Macro(tollfree, $) exten = _3[13]800.,1,Macro(tollfree, $) exten = _44[58]00.,1,Macro(tollfree, $) exten = _44808.,1,Macro(tollfree, $) [macro-tollfree] exten = s,1,SetCallerID,$ exten = s,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = s,3,Playback(invalid) exten = s,4,Hangup exten = s,103,Busy Thank you Duane, that is a very good suggestion (one day I have to get into those macros). Though there is a small glitch. When I enter: exten = s,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r I get: -- Executing Dial(SIP/11-497d, IAX2/x:[EMAIL PROTECTED]/*) in new stack -- Called 491581:[EMAIL PROTECTED]/* and a recording not a valid extension When I change it to: exten = s,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r -- Executing Dial(SIP/11-713e, IAX2/491581:[EMAIL PROTECTED]/*s) in new stack -- Called 491581:[EMAIL PROTECTED]/*s and busy signal Do I need to enter ARG2 as some kind of global environment? ARG2=EXTEN -- #Joseph [tollfree] ; ; terminate toll-free no.'s via fwdnet ; ; ; US toll free access ; ; +1-800 exten = _1800.,1,SetCallerID,$ exten = _1800.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _1800.,3,Playback(invalid) exten = _1800.,4,Hangup exten = _1800.,103,Busy ; +1-866 exten = _1866.,1,SetCallerID,$ exten = _1866.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _1866.,3,Playback(invalid) exten = _1866.,4,Hangup exten = _1866.,103,Busy ; +1-877 exten = _1877.,1,SetCallerID,$ exten = _1877.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _1877.,3,Playback(invalid) exten = _1877.,4,Hangup exten = _1877.,103,Busy ; +1-888 exten = _1888.,1,SetCallerID,$ exten = _1888.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _1888.,3,Playback(invalid) exten = _1888.,4,Hangup exten = _1888.,103,Busy ; ; Netherlands toll free access ; exten = _31800.,1,SetCallerID,$ exten = _31800.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _31800.,3,Playback(invalid) exten = _31800.,4,Hangup exten = _31800.,103,Busy ; ; France toll free access ; exten = _33800.,1,SetCallerID,$ exten = _33800.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _33800.,3,Playback(invalid) exten = _33800.,4,Hangup exten = _33800.,103,Busy ; ; UK toll free access ; ; +44 500 exten = _44500.,1,SetCallerID,$ exten = _44500.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _44500.,3,Playback(invalid) exten = _44500.,4,Hangup exten = _44500.,103,Busy ; +44 800 exten = _44800.,1,SetCallerID,$ exten = _44800.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _44800.,3,Playback(invalid) exten = _44800.,4,Hangup exten = _44800.,103,Busy ; +44 808 exten = _44808.,1,SetCallerID,$ exten = _44808.,2,Dial,IAX2/$:[EMAIL PROTECTED]/*$,60,r) exten = _44808.,3,Playback(invalid) exten = _44808.,4,Hangup exten = _44808.,103,Busy -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ 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
Re: [Asterisk-Users] analog lines via channel bank --
I just bought an adit 600, and it works great. They can be picked up used pretty reasonably (approx. $10-$15/port) It also does callerid and callwaiting great, but seems to have problems doing callwaiting callerid, as do a lot of channel banks I believe, so if this is something important to you, make sure you check this first. Jon. On Thursday 27 January 2005 06:37 am, Mike Dewey wrote: morn, I am looking into a situation where I need 50 or so analog extentions, all of them need to have caller ID. Anyone have any recommendations for channel banks and or tips or warnings on Caller ID to the analog stations. thanks mike ___ 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] Re: Polycom Phones
Walt Reed wrote: If sayson provided developer documentation for their phones and allowed us to write our own firmware, they wouldn't be able to manufacturer them fast enough. They would corner the IP phone market. AMEN. I posted this same type of statement a few months back. I would absolutely love to get a 'hardphone' where I can write my own software for it! Give me source code level access to any of the IP 'screen' phones and IBIH. [* wakes up from dream *] Oh well :) -Chris ___ 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] Least Cost Routing
You could poblate with data using MySQL-Front from any windows/linux wks. www.mysqlfront.de LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan Sent: Thursday, January 27, 2005 11:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Least Cost Routing Matthew, I'm trying to do Least Cost Routing for some International Rates between VoipJet and LiveVoIP. I saw your post about the data in mysql and a later post about the crashing, so that means you did figure out how to get the data into mysql? I compiled it and asterisk loaded it w/o crashing so so far so good, but I don't know how to get the data into mysql. Any help/insight you can provide would be helpful. Then there's this: http://www.voip-info.org/wiki-Application+LCDial Looks similar. Have you got LCR working? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, October 13, 2004 3:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Least Cost Routing Anyone using the rate_engine from TrollPhone? There is absolutly no documentation on how to setup data in the tables. If someone could send sample data, or post it to the wiki, it would be helpful. If any others are successfully using another Least Cost Routing method, please pass it along. THanks, Matthew ___ 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 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] Voice mail
Perhaps another question is what does this person mean by without the pbx part? If they only want to define extensions, sip interface, and voicemail configuration then simply relay any unanswered call from SER to Asterisk then yes this works without the pbx part. _Steve David Boyd wrote: How would you deliver calls to the voicemail system without the PBX functions? db On Thu, 2005-01-27 at 08:22, [EMAIL PROTECTED] wrote: HI I would like know if it's possible to use the VoiceMail only of the Asterisk Sytem without use the PBX part ? Thank. ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ 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] Bristuff ZapHFC and Loosing D-Channel
Hi, that is the usual behaviour on a P2MP BRI line. When idle the telco will bring down layer 2 and layer 1. Bristuff will activate layer 1 and layer 2 again immediately. best regards Klaus Am Donnerstag, den 27.01.2005, 16:01 +0100 schrieb Remco Barende: Hi! Did you ever find the answer to your question? I am getting the same message on the console every second: == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up etc. etc. etc. I'm running Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 The error is only visible however if I run * with -v (but I guess I shouldn't see these messages nonetheless)? On Tue, 25 Jan 2005, Peer Oliver Schmidt wrote: Using the latest(?) bristuff (Asterisk 1.0.4-BRIstuffed-0.2.0-RC3a) I have problems with loosing the D-channel. Most of the time, after the message PRI D-channel down it only takes a second or so to come back up, noted by the message PRI D-channel up However, today most of the time the D-channel stays down. Calls come in, but can't be answered. Does anyone know of a fix for this, or might have some insights on how to circumvent this problem? Any and all help is greatly appreciated. ___ 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
Re: [Asterisk-Users] DTMF digit dropping
Just to clarify Paul, you're connecting to BroadVoice with SIP right? Does BroadVoice now support out-of-band DTMF? DTMF works for me occasionally (over IAX) on inbound calls over VoicePulse. Whenever I achieve success I get all excited and think maybe they fixed it. But then a few more tests and forget about it. Outbound DTMF always seems to work. Maybe it's time to look at the DTMF code in Asterisk. -mark On Jan 26, 2005, at 10:23 AM, Paul Rodan wrote: I have a small IVR on my Asterisk server connected to BroadVoice, I always used DTMF, but I tried to switch to rfc2833 the other day out of curiosity and interesting enough, when I called into my IVR w/ my cell phone, it recognized 1234 and whatever other digits I entered. So inbound DTMF worked using ULaw, however I never tried outbound. Could have been a fluke though. Give it a shot. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Re: Polycom and call waiting again..
Adam Goryachev wrote:
[local-stuff]
; This is where we pretend a channel is an extension
exten = 1234,1,SetGroup(SIP1234)
exten = 1234,2,CheckGroup(1)
exten = 1234,3,Dial(SIP/1234,15)
exten = 1234,104,Busy
[queue-stuff]
exten = 6939,1,AddQueueMember(Local/${CALLERIDNUM})
You are close... that should be:
AddQueueMember(Local/[EMAIL PROTECTED])
That way when the queue app tries to call the agent, it will have an
extension _and_ a context to deliver the call to.
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[Asterisk-Users] Adit 600
Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports Thanks, Isaac ___ 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] grandstream budgetone-100 updates
Try using the HTTP method. It seems to work well with x.18 firmware or newer. On Thu, 2005-01-27 at 09:24 -0500, Dana Olson wrote: I can't tell you why it's failing, as I don't know. But to answer your other question, I have firmware 1.0.5.22 that I found from a link a short time ago on the mailing list. I'm having some issues with DHCP and the BudgeTone phone, as it doesn't seem to like the TFTP options we put in. (I do have the Aastra 480i working properly now though). -- Dana - Original Message - From: dean collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 6:28 PM Subject: [Asterisk-Users] grandstream budgetone-100 updates I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.2025 Jan 18:25 Error Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg.txt Octet, Send 192.168.16.2025 Jan 18:25 Error Can anyone tell me why these fail each time? Also what is the latest revision? Cheers, Dean ___ 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 -- Kim Lux, Diesel Research Inc. ___ 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] grandstream budgetone-100 updates
Hi Dean. A nice site for the GrandStream firmware downloads is http://gs-firmware.gratissip.dk/ There you can download current as well as previous firmware versions. The site supports both http and tftp downloads, which is handy. Cheers, Edge. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson Sent: Thursday, January 27, 2005 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream budgetone-100 updates I can't tell you why it's failing, as I don't know. But to answer your other question, I have firmware 1.0.5.22 that I found from a link a short time ago on the mailing list. I'm having some issues with DHCP and the BudgeTone phone, as it doesn't seem to like the TFTP options we put in. (I do have the Aastra 480i working properly now though). -- Dana - Original Message - From: dean collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 6:28 PM Subject: [Asterisk-Users] grandstream budgetone-100 updates I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.2025 Jan 18:25 Error Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg.txt Octet, Send 192.168.16.2025 Jan 18:25 Error Can anyone tell me why these fail each time? Also what is the latest revision? Cheers, Dean ___ 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
Re: [Asterisk-Users] app_sms: problems sending a sms
Hello Seshu,
i think you solved your problem in the meantime, but here
are my points (for archive purposes), after it works here now.
Thanks Steffen. Please update me if this ever works.
The problem was (probably), that i put some lines in
extension.conf for sending a sms, and triggered
this by calling this extension. Probably the sms app
can't get the right channel then.
After changing this, i had in the extensions.conf:
[smsdial]
exten = _X.,1,SMS(default,,${EXTEN},${MSG})
exten = _X.,2,SMS(default)
exten = _X.,3,Hangup
exten = h,1,Hangup
(like a poster here posted it already) and put a call file
in /var/spool/asterisk/outgoing/
[EMAIL PROTECTED]:/tmp# cat testsms
Channel: Zap/g1/0090032669000
CallerID: SMS 35910
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: smsdial
Extension: 0179XXX
Priority: 1
SetVar: MSG=Text to send
That works here now.
A few hints:
- If it doesn't work, try to call the SMSC with your phone. You should
hear the sound of a bird chirp and the SMSC should hangup after a
few seconds.
If you get something like This number is not complete, try to add
a zero at the end of the number and repeat.
If nothing happens, check out if calling the number of the SMSC is
allowed from your line.
- Put the number of your desk phone in your call file instead of the
number of your SMSC. After putting the call file in the outgoing dir,
your desk phone should ring. Now you know that your asterisk-setup
is ok (or not).
- Get and read the ETSI ES 201 912 standard to understand what's going
on and to know the meaning of the message type codes the sms app
is printing. I think this is a good idea to put this in the wiki.
HTH,
Steffen
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Re: [Asterisk-Users] How to make channel busy signal?
Call waiting? -mark On Jan 26, 2005, at 6:09 PM, Joseph wrote: When I make a call over the Internet and call myself IN over POTS my phone rings to outside party but I can not hear it. Why isn't my channel extension indicating busy status when I'm making call over Internet? This way I could ring my next extension with n+101 priority. I'm using Sipura-3K unit. -- #Joseph ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Directory service of voicemail extensions
AFAIK it does not currently support that. IIRC it actually states somewhere, perhaps on voip-info, that once you enable voicemail db support you will break the directory listing feature. -mark On Jan 27, 2005, at 9:34 AM, Jagan Mohan wrote: Hi, Does Asterisk support Directory service of voicemail extensions using database? If yes, how to configure asterisk? I know that it supports this feature using conf files. Thanks, Jagan ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Anyone having problems with LiveVoIP?
If the problem is with asterisk userswhy is LiveVoip trying to change something at their end? -mark On Jan 26, 2005, at 10:33 AM, Tim Lewis wrote: LiveVoIP did not issue any end user patches last night. They had a problem connecting to Level 3's network. LiveVoIP claimed the problem was with asterisk users, I have not upgrade or install any patches and all is fine now. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Re: Polycom phones
From: Walt Reed [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Polycom Phones To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Jan 26, 2005 at 10:20:24PM -0500, Cory Andrews said: Seshu - the 480i, although a great phones, is quite a bit more expensive than the Polycom IP300 or IP500, it is more comparable in price to the Polycom IP600. Hmm. Your own web site has it priced between the 500 and 600. If the difference is good support versus zero support, wouldn't the $50 difference between the 500 and the 480i be saved in the first 20 minutes you spend fighting with a problem? Another factor is that one company tests with * and the other shuns it. Just the availability of the firmware alone is almost worth the $50. I have no problem with polycom, and use their non-IP conference phones, but I'm not going to purchase a product from a manufacturer that refuses to provide even basic support (complete manuals and firmware.) It would be Very nice to have a phone platform that is fully documented that had firmware that was open and hackable. It seems that people on this list spend massive amounts of time trying to work around all the firmware bugs in various products (eg. call waiting on polycom.) If sayson provided developer documentation for their phones and allowed us to write our own firmware, they wouldn't be able to manufacturer them fast enough. They would corner the IP phone market. I have the same dilemma with Polycom phones. Given their support (actually complete lack of), I am quite loathe to giving them business. On the other hand they are so darned cheap compared to other similar phones, I sure get tempted to use them if I can find a workaround. Compare Polycom IP-500 for $170 vs. Sayson-480i for $250 or SNOM-190 for $220. If it were a matter of 1 or 2 phones, I will gladly go with SNOM or Sayson, but if I have to buy 50, Polycoms become irresistible. As a matter of fact, my current client needs 120 phones to work with Asterisk. I have to make a decision soon about which one do I give to him. --jt ___ 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] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk
I'm testing a bunch of stuff before we implement our system. I've got a SIP account and Grandstream phones. We haven't started using asterisk yet. Generally we've got good voice quality from all the offices except: a) We get a lot of echo in the first 10 seconds or so of the call, only on the VOIP calling end. The callee says the speech sounds normal. To the caller, the first Hello is almost intelligible with echo. b) The first part of an abrupt statement from one party gets clipped. In conversation, when talking switches from one party to the other, a tiny bit of speach gets clipped. c) If both parties talk at once there is a bit of dropout. We'd like to improve the voice quality in these respects. Otherwise the voice quality is excellent. I've been told it is better than the traditional system several times. Questions: a) Are certain codecs better than others at quickly getting the echo cancellation setup ? Is there a way to get the echo out of the call immediately ? (Is there a document explaining the features and pitfalls of all the codecs somewhere ?) b) Is there a way to eliminate the speech clipping when speakers change or both talk at once ? I've read about asterisk injecting noise and/or sending packets in the absence of speech. Would that help ? Is this what the Grandstream Silence Suppression is about ? c) How does one know where to set the following: iLBC frame size: 20ms 30ms iLBC payload type: (between 96 and 127, default is 98) Silence Suppression: No Yes Voice Frames per TX: (up to 10/20/32/64 for G711/G726/G723/other codecs respectively) Layer 3 QoS: (Diff-Serv or Precedence value) Layer 2 QoS: 802.1Q/VLAN Tag 802.1p priority value (0-7) d) One place we've really got a problem is when we use a Grandstream in a big echoy (sp!) room. We seem to get echo from the room into the call which seems to fool the echo cancellation. Any ideas on how to get around this problem ? d) How is asterisk going to change our sound quality when it is added between the phones and the SIP provider ? Does it have features that will help with the echo and clipping and if so, how much improvement should we expect ? Thanks. -- Kim Lux, Diesel Research Inc. ___ 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] grandstream budgetone-100 updates
I don't use custom ringtones so maybe this is why the following files fail. Aborted 192.168.16.30 C:\Program Files\TFTP Desktop\1.0.5.22\ring1.bin Octet, Send 192.168.16.20 27 Jan 10:53 Error Aborted 192.168.16.30 C:\Program Files\TFTP Desktop\1.0.5.22\ring2.bin Octet, Send 192.168.16.20 27 Jan 10:53 Error Aborted 192.168.16.30 C:\Program Files\TFTP Desktop\1.0.5.22\cfg000b82003884Octet, Send 192.168.16.20 27 Jan 10:53Error Aborted 192.168.16.30 C:\Program Files\TFTP Desktop\1.0.5.22\cfg.txtOctet, Send 192.168.16.20 27 Jan 10:53 Error Is there anything wrong with these failing? Also how do you set up a http file transfer? TIA, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim Lux Sent: Thursday, January 27, 2005 10:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream budgetone-100 updates Try using the HTTP method. It seems to work well with x.18 firmware or newer. On Thu, 2005-01-27 at 09:24 -0500, Dana Olson wrote: I can't tell you why it's failing, as I don't know. But to answer your other question, I have firmware 1.0.5.22 that I found from a link a short time ago on the mailing list. I'm having some issues with DHCP and the BudgeTone phone, as it doesn't seem to like the TFTP options we put in. (I do have the Aastra 480i working properly now though). -- Dana - Original Message - From: dean collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 6:28 PM Subject: [Asterisk-Users] grandstream budgetone-100 updates I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.2025 Jan 18:25 Error Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg.txt Octet, Send 192.168.16.2025 Jan 18:25 Error Can anyone tell me why these fail each time? Also what is the latest revision? Cheers, Dean ___ 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 -- Kim Lux, Diesel Research Inc. ___ 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
Re: [Asterisk-Users] Adit 600
have you tried contacting carrier access to see if they have a config for asterisk and the adit 600? when we needed to hook an adit to our softswitch, they provided a config for both ends that worked flawlessly. i can give you the config off our adit and tell you what mgcp parameters we have enabled on our softswitch if that would help. On Thursday 27 January 2005 09:25, Isaac McDonald wrote: Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports Thanks, Isaac ___ 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 -- Chad Whitten Network Administrator neXband Communications [EMAIL PROTECTED] 601-944-4801 Phone 601-944-4803 Fax ___ 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] grandstream budgetone-100 updates
On Thu, 2005-01-27 at 10:59 -0500, dean collins wrote: Also how do you set up a http file transfer? My phones were running firmware version x.18. There was a field that allowed me to select automatic updates and how often. I selected yes and set it to 1 day. (I thought maybe 0 days would cause it to update immediately, but all it caused was an error.) There was a field for http updates. I set it to yes and set the http address to http://fm.grandstream.com/gs/ I then powered down the phone and powered it back up. This caused the firmware to upgrade. I then logged into it via the web interface and checked the firmware version on the basic tab. It was then at x.22. I saw was in the above statements because the user interface was different on the x.18 firmware and I am recalling from memory. -- Kim Lux, Diesel Research Inc. ___ 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] Bristuff ZapHFC and Loosing D-Channel
Remco Barende wrote: Did you ever find the answer to your question? I am getting the same message on the console every second: == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up etc. etc. etc. I'm running Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 The error is only visible however if I run * with -v (but I guess I shouldn't see these messages nonetheless)? No, I still have these messages, and was hoping RC5 would fix them. Nice to know that upgrading won't help :-( Do you get calls which stop in the middle of the conversation as well? -- Best regards Peer Oliver Schmidt ___ 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] Need some advises configuring asterisk to call over INTERNET
Hello guys I just looking for some advises in order to configure my asterisk server to receive and make calls over internet, i got a 384 kb adsl connection. i just need any information regarding this matter , codecs (installing g729,g723) bandwidth, configuring public IP with adsl and others things to keep in mind, i need anything, so anyone that allready done this please take a few seconds to give me some advise Thank You ___ 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] Re: TDM400 - channel out to lunch?
Not to mention the CPU spikes every n seconds. Rich, while you're testing, would you keep an eye on 'vmstst 1' and the 'system' (not user) CPU utilization? That cpu spiking is another issue separate from the stability issue (I think). Not sure where the discussion of the spiking ended up a few weeks ago, do you remember? I spoke with Matt O. at Digium tech support. He was in my machine running 'vmstat 1'. I demonstrated how the CPU spiking stopped when I unloaded the wcfxs module. I pointed-out that, during a spike, the total number of interrupts did not increase, suggesting that one interrupt was being held for 20+ms. Matt then did the same tests on his own machine and observed the same results. Matt said Mark was looking at the problem. I believe this spiking is at the root of my spandsp problems, causing a periodic frame slip and the failure of the fax transmission. That is the latest information I have. I've not tried to dig through the code, but it wouldn't surprise me if some temp code exists that might be polling the tdm card (or something like that) as an aid towards identifying the stability issue. Gut feeling suggests that if stability is truly related to tdm design problems (or whatever), then resolving that issue probably should be a precursor to chasing the cpu spikes. Agree. But if you can get the card to go crazy as a result of physical pressure then it would be interesting to see if there is a correlation to CPU usage. Something in wcfxs is causing it to not exit its interrupt routine in a normal manner. ___ 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] Re: Polycom and call waiting again..
Here is what I have done to get around the call waiting problem.
This is for a Polycom 500. This is kind of a pain, but it works.
Exten.conf
exten = 1051,1,Dial(SIP/1051,20,tTr)
exten = 1051,2,Voicemail(u${EXTEN})
exten = 1051,102,Dial(SIP/1051b,20,tTr)
exten = 1051,103,Dial(SIP/1051c,20,tTr)
exten = 1051,104,Voicemail(b${EXTEN})
Sip.conf
[1051]
type=friend
username=1051c
callerid=NMS0011051
host=dynamic
dtmfmode=rfc2833
mailbox=1051
context=sip
callgroup=1
pickupgroup=1
canreinvite=no
imcominglimit=1
[1051b]
type=friend
username=1051c
callerid=NMS0011051
host=dynamic
dtmfmode=rfc2833
mailbox=1051
context=sip
callgroup=1
pickupgroup=1
canreinvite=no
imcominglimit=1
[1051c]
type=friend
username=1051c
callerid=NMS0011051
host=dynamic
dtmfmode=rfc2833
mailbox=1051
context=sip
callgroup=1
pickupgroup=1
canreinvite=no
imcominglimit=1
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again..
Adam Goryachev wrote:
[local-stuff]
; This is where we pretend a channel is an extension
exten = 1234,1,SetGroup(SIP1234)
exten = 1234,2,CheckGroup(1)
exten = 1234,3,Dial(SIP/1234,15)
exten = 1234,104,Busy
[queue-stuff]
exten = 6939,1,AddQueueMember(Local/${CALLERIDNUM})
You are close... that should be:
AddQueueMember(Local/[EMAIL PROTECTED])
That way when the queue app tries to call the agent, it will have an
extension _and_ a context to deliver the call to.
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RE: [Asterisk-Users] Re: Polycom phones
/Snip/ I have the same dilemma with Polycom phones. Given their support (actually complete lack of), I am quite loathe to giving them business. On the other hand they are so darned cheap compared to other similar phones, I sure get tempted to use them if I can find a workaround. Compare Polycom IP-500 for $170 vs. Sayson-480i for $250 or SNOM-190 for $220. If it were a matter of 1 or 2 phones, I will gladly go with SNOM or Sayson, but if I have to buy 50, Polycoms become irresistible. As a matter of fact, my current client needs 120 phones to work with Asterisk. I have to make a decision soon about which one do I give to him. --jt /Snip/ I use Polycom phones and have no issues with the firmware or the quality of the calls. My opinion (guess) on Polycom's Asterisk policy is - It is not that Polycom does not want their phones to be used with Asterisk. At the price these phones are sold, they will not be able provide support for all the features (AKA bugs or quirks) of Asterisk and make them transparent to Asterisk SIP stack and more notably - be user friendly for the Asterisk newbie user community. :) Remember that is is just like any other good SIP Phone and there are thousands of satisfied customers who did not have any problem with the firmware. If you know how to install this phone, you have a good product there and it works as long as you don't need upgrades. Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] TDM-400P + CallerID
Hi, Im just starting out with Asterisk, in combination with a TDM400, filled with 2 FXS on channels 1 and 2, and 1 FXO on 4. Having just started, all I want right now is to be able to answer incoming calls on a phone connected to channel 1. The trouble is the caller id. I have caller id enabled on my line, my phone supports it, and when I connect the phone directly to the line, it works. However, it doesnt work with *. When I call myself (with a cellphone), and I type show channel zap/4-1 in the * console, it shows my cellphone# in the caller id field. Asterisk gets the correct callerid from my line, appearantly. When I type show channel zap/1-1, the caller id just shows s. I have a feeling that this s is the originating extension, seen from the FXS point of view. My phone just shows external call, instead of a number. How do I make * forward the callerid from the incoming call to my phone? --Pieter My zapata.conf: context=buitenlijn signalling=fxs_ks immediate=yes usecallerid=yes cidsignalling=dtmf cidstart=polarity hidecallerid=no callerid=asreceived callwaiting=no callwaitingcallerid=no adsi=no channel = 4 signalling=fxo_ks language=nl usedistinctiveringdetection=no busydetect=yes echocancel=yes echotraining=no channel = 1 channel = 2 My extensions.conf: [buitenlijn] exten = s,1,Wait(2) exten = s,2,Dial(Zap/1,30,t) ___ 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
