Re: [asterisk-users] Best wifi IP phone for asterisk
Michelle Dupuis wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the only telco's get documentation crap) NOT SIP, but AFAIK supported by Asterisk: Cisco IP Phone 7920. SCCP, basic functions (answer, place a call, transfer) does work according to voip-info page. Price: starting at US$ 150 @ eBay. US$ 345 NIB. If it is a large purchase and you are in the US, Cisco does offer a very competitive lease financing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk hardware E1 pri card
Dear all I have setup with mediant 2000 with avaya now i want to install E1/PRI card with asterisk and trunk with E1 with Avaya E1 port so i want to buy E1 card for asterisk so which card is best and cast effective for my setup i want 1 port E1 card so can you suggest me which card is best for my setup and i want QSIG singaling with avaya Regards satish patel - Got a little couch potato? Check out fun summer activities for kids.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got reject for frame
Many thanks, The cable is working, i only changed the server and the telephony card from sangoma to digium. After this, the error occurs, the timing comes from one port (pri_cpe mode) and 2 other port get it (pri_net). with asterisk 1.2 (newest verion) i got this error every minute, with asterisk 1.4 i got it every hour. any idea? do you need the configuration files? Thanks Nico On Tue, 3 Jul 2007, Matthew Fredrickson wrote: These are the things you should check first: 1.) Make sure that your cable/line is not faulty. 2.) Make sure you are running the latest version of zaptel for your particular branch (1.2 or 1.4) 3.) Make sure that your timing is correct for the span in zaptel.conf Example: (If it's a span from the telco, second digit should be a one) span=1,1,0,esf,b8zs (if it's a span to another PBX or channel bank which is pulling timing from you, the second digit should be a 0) span=1,0,0,esf,b8zs --- Matthew Fredrickson Software Engineer Digium, Inc. On Jul 2, 2007, at 2:52 AM, [EMAIL PROTECTED] wrote: Hello, I have many machines working with asterisk and a Digium 4E1 card. I build a new machine and a failure started like: Got reject for frame 46, retransmitting frame 46 now, updating n_r! This error i got every 2 minutes. Do anybody has an idea? Thanks Nico ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2 TDM24xx and B410P
Matthew Fredrickson wrote: On Jul 2, 2007, at 6:02 PM, Tzafrir Cohen wrote: [...] That's what I would say as well. Also, what's the output of dmesg when you releoad the card. Hi Matthew, this is what we get after rmmod modprobe wctdm24xxp [Jul 4 00:25:24] ERROR[20034]: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Jul 4 00:25:24] ERROR[20034]: Unable to register channel '1-2' [Jul 4 00:25:24] WARNING[20034]: chan_zap.so: load_module failed, returning -1 [Jul 4 00:25:24] WARNING[20034]: Loading module chan_zap.so failed! [Jul 4 00:25:24] NOTICE[20053]: CDR simple logging enabled. [Jul 4 00:25:24] WARNING[20053]: Unable to specify channel 1: No such device or address [Jul 4 00:25:24] ERROR[20053]: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Jul 4 00:25:24] ERROR[20053]: Unable to register channel '1-2' [Jul 4 00:25:24] WARNING[20053]: chan_zap.so: load_module failed, returning -1 [Jul 4 00:25:24] WARNING[20053]: Loading module chan_zap.so failed! [Jul 4 00:25:25] NOTICE[20070]: CDR simple logging enabled. [Jul 4 00:25:25] WARNING[20070]: Unable to specify channel 1: No such device or address [Jul 4 00:25:25] ERROR[20070]: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gigaset 450IP loses registration
Hi Token PBX, we are replacing all Gigaset with ATA + analogic cordless. We think Siemens is the only one who can solve the problem fixing bugs and letting us to downgrade to previous firmware. BTW if asked I'd tell everyone not to buy Gigaset C450IP to be used with Asterisk. In our opinion, a company who does not let to manage firmware version is not serious. Giorgio Incantalupo Token PBX wrote: Hi! I have the same phone with the same problems: 1. Asterisk box does not have fixed IP address, but dyndns name. 2. Phone is at a different location, connected to a router/ADSL modem Siemens Gigaset (with option not to disconnect from internet ever - set on). 3. Inside asterisk LAN, phone didn't loose connection ever. 4. In sip.conf NAT is set 5. In phone settings NAT is set also, and sip proxy is set to asterisk's box dyndns name. 6. When phone is seen as unreachable by asterisk box, and router is reset on remote location, phone reregisters. Any help is appreciated. Thnx Mihaela MJ. On 7/3/07, *gincantalupo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Olivier, I forgot to mention it is a C450IP. But if you have some hint on S maybe it can help me. Perhaps it is some configuration...I tried with qulify=no as I read on a web page without success. Thank you. Giorgio Incantalupo Olivier wrote: Is it a S 450IP ou C 450IP ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] registering Asterisk on SIP/Nortel MCS server
Hi Kate, The Nortel MCS' SIP stack is a little special. You need to add special headers to the REGISTER messages in order to register with it. That means you'll need to touch Asterisk's source code. You can use wireshark to capture a successful REGISTER and see what headers you need. Regards, Brian Neotiq Consulting www.neotiq.com On 6/28/07, Kate Kretz [EMAIL PROTECTED] wrote: hello there... our telecom sold us VoIP-numbering, managed by Nortel MCS I successfully registered Ekiga to it ( http://sol.chel.skbkontur.ru/ekiga.png) What exactly do I have to write in sip.conf to make Asterisk register on this SIP ? Cheers, Kate ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk TV will go live this Friday
In conjunction with Mark Spencer's visit to our Paris office, we'll be
kicking off Asterisk TV (http://asterisktv.com) live during the weekls
Asterisk Users Conference. I believe someone from Lumenvox will be
back with us on the conference, now that I've had a chance to play
with their speech recognition product.
I think we're starting to get some great info from the user community
and I would like to see more service providers show themselves. I
think it's great to hear what people like JerJer of Nufone to say.
We've had some bleeding edge stuff on past conferences as well, with
Jay from Adhearsion and I'm waiting for some video content from him as
well.
Digium has been great about following the evolution of these community
projects and providing time for people like Russell to join the
conference and take users' questions.
If you have any video to contribute to this 24/7 Internet TV channel,
please contact me off list. And whatever you do, if humanly possible,
join the conference with Mark this Friday
at 12:30PM EDT (9:30AM PDT, 16:30 GMT, ${UNGODLYHOUR}+${NEXTDAY} in Tokyo).
Instructions: http://x2z.eu
Video stream: http://asterisktv.com (live Friday)
randy
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Re: [asterisk-users] Music on hold - 1.4.5
Stephen Bosch wrote: Ade Vickers wrote: Hi Richard, Thanks for those replies - I'll give them a shot shortly. That's not really what I meant by configuration -- you can choose the MOH source for Asterisk. It's only the native player that restarts the music file every time someone is put on hold. We're still using 1.2, which doesn't have this problem, so I don't know how it's done in 1.4. Ah well, whatever was meant, using rawplayer works more or less as I'd hoped (music doesn't re-cue from the start). It's not a constant stream (if no MoH is playing at all, the stream freezes so the next person to get music gets it from the point where the previous person left off... The most amusing bit was when the original raw files I'd converted from MP3 using a Windows tool played at approx 2x their normal speed! Re-converting with mpg123 fixed that... Cheers! Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.14/885 - Release Date: 03/07/2007 10:02 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialout Macro and transfer call in progress
Dear All,
I can not transfer call in progress. What's wrong with my macro? I think tT
flags is enough right?
[macro-stdexten]
exten = s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key
exten = s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key
exten = s,3,GotoIf($[${temp} = ]?5); If not existing, goto
priority 5
exten = s,4,Dial(Local/[EMAIL PROTECTED]/n); Unconditional Forward
exten = s,5,GoToIf($[${DNDStatus} = ]?7) ; If not existing ring the
interface
exten = s,6,Voicemail(u${ARG1}); If CFU failed, send to
voicemail w/ unavail announce
exten = s,7,Dial(${ARG2},20,tTrR) ; Ring the interface
exten = s,8,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Set(temp=${DB(CFNA/${ARG1})}) ; Get CFNA key
exten = s-NOANSWER,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-NOANSWER,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on No Answer
exten = s-NOANSWER,4,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,5,Goto(s,5) ; If they press #, return to
ring the interface
exten = s-BUSY,1,Set(temp=${DB(CFB/${ARG1})}) ; Get CFB key
exten = s-BUSY,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-BUSY,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on Busy
exten = s-BUSY,4,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
exten = s-BUSY,5,Goto(s,5) ; If they press #, return to
ring the interface
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten = a,1,VoicemailMain([EMAIL PROTECTED]) ; If they press *, send the
user into VoicemailMai
[from-internal]
exten = 200,1,Macro(stdexten,200,SIP/user200)
...
Thanks in advance,
Dominik___
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Re: [asterisk-users] Asterisk Support Question
On Tue, 3 Jul 2007, Jonathan Creasy wrote: If it is one of the ones I am familiar with it's only one ethernet interface and it's literally a switch on a PCI card. The system sees one interface and there are 4 ports out the back. If this is the case it's not really instead of a switch so it will work fine. Anyone care to tell give me a part number/product code/manufacturer? As I'd be intersted in this too... I have used a 4-port Ethernet card in the past - D-Link using the sundance drivers, but while I used them successfully under the 2.4 kernel, I've never been able to use them under 2.6 )-: (And using the kernels 802.1d features, you can turn individual ethernet interfaces into a switch, not as efficient as a switch, but there you go, and it's worked well for me in the past) Gordon -Jonathan Goran Donev wrote: I am thinking of building an Asterisk PBX, and had a question on a piece of hardware support. I want to include a 4 port PCI 10/100 Switch router card. For those not familiar it's a PCI card that acts as a switch. My question is would I be able to configure those 4 ports to support sip phones plugged in directly to the asterisk box instead of a switch. Thanks in advance ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google acquires Grand Central
don't scare people for GOD sake :) LOL On 7/4/07, Jaswinder Singh [EMAIL PROTECTED] wrote: Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . On 03/07/07, Dean Collins [EMAIL PROTECTED] wrote: Ooops did Google just become a carrier :) http://googleblog.blogspot.com/2007/07/all-aboard.html I hear stocks crumbling worldwide as I type. Cheers, Dean ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialout Macro and transfer call in progress
whats the key sequence you use to transfer. check the key sequence
definition in features.conf, you maybe using a wrong keysequence. also check
whether your asterisk is recieving the dtmf digits passed from your user
agent or not. if the digits are not recieved by asterisk then there is no
way you can tell asterisk to transfer the call. check for the following
settings in sip.conf file general settings:
dtmf=rfc2833
canreinvite=no
On 7/4/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Dear All,
I can not transfer call in progress. What's wrong with my macro? I think
tT flags is enough right?
[macro-stdexten]
exten = s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key
exten = s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key
exten = s,3,GotoIf($[${temp} = ]?5) ; If not existing, goto priority
5
exten = s,4,Dial(Local/[EMAIL PROTECTED]/n) ; Unconditional Forward
exten = s,5,GoToIf($[${DNDStatus} = ]?7) ; If not existing ring the
interface
exten = s,6,Voicemail(u${ARG1}) ; If CFU failed, send to voicemail w/
unavail announce
exten = s,7,Dial(${ARG2},20,tTrR) ; Ring the interface
exten = s,8,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Set(temp=${DB(CFNA/${ARG1})}) ; Get CFNA key
exten = s-NOANSWER,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-NOANSWER,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on No Answer
exten = s-NOANSWER,4,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,5,Goto(s,5) ; If they press #, return to ring the
interface
exten = s-BUSY,1,Set(temp=${DB(CFB/${ARG1})}) ; Get CFB key
exten = s-BUSY,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-BUSY,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on Busy
exten = s-BUSY,4,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy
announce
exten = s-BUSY,5,Goto(s,5) ; If they press #, return to ring the
interface
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain([EMAIL PROTECTED]) ; If they press *, send the
user into VoicemailMai
[from-internal]
exten = 200,1,Macro(stdexten,200,SIP/user200)
...
Thanks in advance,
Dominik
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Best Regards
Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com
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Re: [asterisk-users] call transfer not working
check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general settings. On 7/4/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]---[Mediant2k][Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task regards -- Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with internal extensions
plx show us your sip and extension configuration. On 7/3/07, Steve Dickey [EMAIL PROTECTED] wrote: I have a cisco 7905 running sip code that is successfully connecting to my asterisk system. I also have a softphone that is connecting to the system. I can make a call from the cisco extension and the softphone rings. However; I can not make a call from the softphone to the cisco extension. it says the user is unavailable. I am not sure what to check. thanks in advance. scd -- Steve Dickey Who is John Galt? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Mail not Receive
i think your goto statement after dial is incorrect. you are missing the
first comma in it:
your statement is:
exten = 50,4,Goto(ss-${DIALSTATUS},1) ; here you r telling the atserisk
server that you want to jump to context ss-${DIALSTATUS}, extension 1.
it should be
exten = 50,4,Goto(,ss-${DIALSTATUS},1) ; here its telling asterisk to jump
to extension ss-${DIALSTATUS}, priority 1, in the same context.
On 6/29/07, Asif Raza [EMAIL PROTECTED] wrote:
hi,
i am using Asterisk 1.4. and unable to get Voice Mail below is my config
extensions.conf
exten = 50,1,NoOp(Failover)
exten = 50,2,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = 50,3,Dial(SIP/101,18)
exten = 50,4,Goto(ss-${DIALSTATUS},1)
exten = ss-NOANSWER,1,StopMixMonitor()
exten = ss-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
voicemail.conf
[salesvoice]
777 = 1212, sales, [EMAIL PROTECTED]
with same setting i m getting voice mail when i use Asterisk-1.2 but
when i use Asterisk-1.4 i m not able to get a voice mail with these
setting.
Please help me regarding this issue.
thanks
Muhammad Asif
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Rizwan Hisham
Software Engineer
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www.axvoice.com
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[asterisk-users] List delays
Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7920
Has anyone made the Cisco 7920 work with Asterisk? Any feedback on functionality, quality, compatibility, etc? -MD- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Wednesday, July 04, 2007 1:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best wifi IP phone for asterisk Michelle Dupuis wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the only telco's get documentation crap) NOT SIP, but AFAIK supported by Asterisk: Cisco IP Phone 7920. SCCP, basic functions (answer, place a call, transfer) does work according to voip-info page. Price: starting at US$ 150 @ eBay. US$ 345 NIB. If it is a large purchase and you are in the US, Cisco does offer a very competitive lease financing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
On 7/3/07, Joe acquisto [EMAIL PROTECTED] wrote: Contrary to the opinions of Anglo-Philes, we, here in the Colonies, speak American, not English. In some places, 'Murican. We get to do that, because, back in the late 1700's . . . we won. It is only referred to as English out of a sense of compassion. Oh, so anyway, who was guy Eng you named the country after? joe a. ANd here I thought we spoke Spanish, because where I'm from, you better or else you won't be able to talk to any of the contractors that show up at your house or business (plumbers, electricians, cable installers, etc.) nor the people at the grocery stores, restaurants, just about anyone. The official language around here is Spanish. They just, for some reason, haven't passed the law yet. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] You speaka Ingrish!!! WAS RE: Suing Dell||Dull Computers for CID abuse
You must be in Miami! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, July 04, 2007 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse On 7/3/07, Joe acquisto [EMAIL PROTECTED] wrote: Contrary to the opinions of Anglo-Philes, we, here in the Colonies, speak American, not English. In some places, 'Murican. We get to do that, because, back in the late 1700's . . . we won. It is only referred to as English out of a sense of compassion. Oh, so anyway, who was guy Eng you named the country after? joe a. ANd here I thought we spoke Spanish, because where I'm from, you better or else you won't be able to talk to any of the contractors that show up at your house or business (plumbers, electricians, cable installers, etc.) nor the people at the grocery stores, restaurants, just about anyone. The official language around here is Spanish. They just, for some reason, haven't passed the law yet. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
On Wed, Jul 04, 2007 at 08:06:49AM -0500, Lacy Moore - Aspendora wrote: On 7/3/07, Joe acquisto [EMAIL PROTECTED] wrote: Contrary to the opinions of Anglo-Philes, we, here in the Colonies, speak American, not English. In some places, 'Murican. Merkins speaking Murican ... We get to do that, because, back in the late 1700's . . . we won. We let you win, you were terrorists and England's never been good at fighting terrorists. Now you're having the same problem !!! It is only referred to as English out of a sense of compassion. American English ... Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
On Jul 4, 2007, at 9:59 AM, Steve Kennedy wrote: Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Amerigo Vespucci -chris www.mythtech.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Being Independence Day and all Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Steve The name was derived from the Latinized http://en.wikipedia.org/wiki/Latinversion of the explorer Amerigo Vespucci http://en.wikipedia.org/wiki/Amerigo_Vespucci's name, *Americus Vespucius*, in its feminine form, *America*, as the other continents all have Latin feminine names. Wikipedia and a memory for things historical. Have a good 4th. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with misdn and ChanIsAvail
Hello guys,
i have some problems with chanisavail and misdn.
Used the following syntax
Chanisavail(misdn/g:TEPortsIAX2/trunktosecondserver)
Checked with
Chanisavail(misdn/1IAX2/trunktosecondserver)
Chanisavail(misdn/1/${EXTEN}IAX2/trunktosecondserver)
too.
I always get the reply mISDN/0-u11 (only the id after the - is
different) for the variable ${AVAILCHAN}.
But mISDN has no port 0, it starts at port 1. I am not able to
understand the problem.
With IAX and chanisavail i have no problems.
Is anybody able to help me with this problem?
Thank you very much.
Dominic
HDPnet GmbH
Erwin-Rohde-Str. 18
69120 Heidelberg
Geschaeftsfuehrer: Marc Hermann
Registergericht: Mannheim HRB 337012
Sitz: Heidelberg
Umsatzsteuer ID Nr.: DE 211 257 470
www.hdpnet.de
Diese E-Mail enthaelt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail
irrtuemlich erhalten haben, informieren Sie bitte sofort den Absender und
vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte
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Re: [asterisk-users] List delays
Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! I'm getting new messages within a matter of minutes. I dunno. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. On 6/27/07, Joe Greco [EMAIL PROTECTED] wrote: Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. . Has anyone else seen a working hardware solution that didn't cost an arm and a leg? It seems to me that a BRI card should cost less than $100. I think I remember a German friend telling me that they go for around $40 dollars. I know that I can get a BRI with voice service out of Bell. I think they have to provide it because of the CRTC tariffs. The thing that has stopped me from trying it in the past is the uncertainty around hardware. Do I understand correctly that NA (North American?) BRI is different from the European version and that European hardware won't work? If I could get a card for a few hundred bucks then I'd be willing to give this a shot. Unfortunately, I can't afford a few grand for the Adtran setup described, although it does sound cool and the BRI-PRI conversion approach is a clever way of overcoming the hardware scarcity. For the number of times that I see people trying to get digital style features out of analog lines, and banging their head against the wall, I'd love to get a BRI working and be able to tell you all how it worked out. Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
In fact it is nearly the same thing with the English who come from a group of germanic speaking group the Angles... So it is not Eng but Angles (like in geometry) ... you can see http://en.wikipedia.org/wiki/Angles 2007/7/4, Bruce Reeves [EMAIL PROTECTED]: Being Independence Day and all Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Steve The name was derived from the Latinizedhttp://en.wikipedia.org/wiki/Latinversion of the explorer Amerigo Vespucci http://en.wikipedia.org/wiki/Amerigo_Vespucci's name, *Americus Vespucius*, in its feminine form, *America*, as the other continents all have Latin feminine names. Wikipedia and a memory for things historical. Have a good 4th. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H263-2000 video format
I'm trying to connect my asterisk 1.4.6 to a system that provides video content (through SIP). Problem is my video system only speaks H263-2000 version (aka H263++). As far as I can see, * only understands H263 and H263+ in and sdp. Can anybody tell me how to extend asterisk so it'll support H263++? From what I've heard, H263+ and H263++ should be compatible. So I was thinking there's no need for a new codec. Am I right? Cheers, K ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
On Wed, 4 Jul 2007, Dave Donovan wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... http://www.tekheads.co.uk/s/product?product=604415gclid=CN6Qp6abjo0CFRDXEAodGHjJjQ Whether it will work with asterisk is another matter though. And what it has (or rather doesn't have!) when compared to those that cost £100s is an intersting question, but I'm suspecting it's something along the lines of the soft modem equivalent - ie. it's doing a lot more in software to simplfy the hardware which may mean more CPU overheard. But saying that, I once used a cheap ISDN card for over a year on a data connection in a linux box (used the ISDN4Linux drivers in the kernel) and it just worked and didn't notice any additional CPU overheard. Gordon On 6/27/07, Joe Greco [EMAIL PROTECTED] wrote: Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. . Has anyone else seen a working hardware solution that didn't cost an arm and a leg? It seems to me that a BRI card should cost less than $100. I think I remember a German friend telling me that they go for around $40 dollars. I know that I can get a BRI with voice service out of Bell. I think they have to provide it because of the CRTC tariffs. The thing that has stopped me from trying it in the past is the uncertainty around hardware. Do I understand correctly that NA (North American?) BRI is different from the European version and that European hardware won't work? If I could get a card for a few hundred bucks then I'd be willing to give this a shot. Unfortunately, I can't afford a few grand for the Adtran setup described, although it does sound cool and the BRI-PRI conversion approach is a clever way of overcoming the hardware scarcity. For the number of times that I see people trying to get digital style features out of analog lines, and banging their head against the wall, I'd love to get a BRI working and be able to tell you all how it worked out. Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Steve Kennedy wrote: On Wed, Jul 04, 2007 at 08:06:49AM -0500, Lacy Moore - Aspendora wrote: On 7/3/07, Joe acquisto [EMAIL PROTECTED] wrote: Contrary to the opinions of Anglo-Philes, we, here in the Colonies, speak American, not English. In some places, 'Murican. Merkins speaking Murican ... merkin - n. A pubic wig for women (http://www.yourdictionary.com/ahd/m/m0230750.html) -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
I have the same problem. My mail sent yesterday around 20:00h and it still not arrived at the list. Sent from germany by the way. Christian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Has anyone else seen a working hardware solution that didn't cost an arm and a leg? It seems to me that a BRI card should cost less than $100. I think I remember a German friend telling me that they go for around $40 dollars. I believe that there's no electrical difference. I know that I can get a BRI with voice service out of Bell. I think they have to provide it because of the CRTC tariffs. The thing that has stopped me from trying it in the past is the uncertainty around hardware. Do I understand correctly that NA (North American?) BRI is different from the European version and that European hardware won't work? The European hardware is supposed to be just fine. Unfortunately, it is the European software that is a stumbling point. US signalling formats are different, and so your software (drivers) need to be aware of how to deal with US BRI. There's supposedly some extremely expensive US BRI ISDN card that works with Asterisk, but it is pretty damn expensive, and the few situations where I've heard it has been used were reported to be fairly unstable. Eicon Diva? I'm running late or I'd look it up. Haven't heard of anyone trying it recently. YMMV. If I could get a card for a few hundred bucks then I'd be willing to give this a shot. Unfortunately, I can't afford a few grand for the Adtran setup described, although it does sound cool and the BRI-PRI conversion approach is a clever way of overcoming the hardware scarcity. For the number of times that I see people trying to get digital style features out of analog lines, and banging their head against the wall, I'd love to get a BRI working and be able to tell you all how it worked out. That was why we ended up spending an arm and a leg, and still didn't end up perfectly happy. Really, if you find a solution, post it. And feel free to mail me about it too. I'd love a highly flexible solution that was well-integrated with Asterisk. What we have now works, mostly, but is a botch. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Dave Donovan [EMAIL PROTECTED]: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. I hear you loud and clear - I am in much the same situation. from what I know about the cards (which isn't much) there are 2 types just like there are modems and winmodems, the hard bri cards understand all the signalling in hardware (its different between na/euro much like a modem has different standards it can adhere to when connecting to different remotes even though they are all carried on the same physical type of line) so if you get a hardware card you need one that understands our signalling here. The soft cards are pretty much universal since they are implemented in software only, but you need a driver that understands the NA signalling - I am not sure about the state of this situation right now. just like their analog cousins the soft cards are cheap, the hard cards are costly, and the NA ones rare. On 6/27/07, Joe Greco [EMAIL PROTECTED] wrote: Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. . Has anyone else seen a working hardware solution that didn't cost an arm and a leg? It seems to me that a BRI card should cost less than $100. I think I remember a German friend telling me that they go for around $40 dollars. I know that I can get a BRI with voice service out of Bell. I think they have to provide it because of the CRTC tariffs. The thing that has stopped me from trying it in the past is the uncertainty around hardware. Do I understand correctly that NA (North American?) BRI is different from the European version and that European hardware won't work? If I could get a card for a few hundred bucks then I'd be willing to give this a shot. Unfortunately, I can't afford a few grand for the Adtran setup described, although it does sound cool and the BRI-PRI conversion approach is a clever way of overcoming the hardware scarcity. For the number of times that I see people trying to get digital style features out of analog lines, and banging their head against the wall, I'd love to get a BRI working and be able to tell you all how it worked out. Dave Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Panasonic TDA200
On Tue, 2007-07-03 at 23:50 -0400, C F wrote: Change it to ISDN. There is no point in not to, what card do you have in the TDA200? A PRI or or just T/E1? Since it's too differenct cards on the TDA200. In fact accroding to Panasonic CallerID isn't supported on none PRI, although some have gotten it to work. The card on the TDA200 is a MFC/R2 E1 card, you cannot use ISDN. Obviously the customer is not willing to pay for an ISDN card. The TDA received CID from the phone company without any problem, but it will not receive it from Asterisk. If you enable CID on the TDA it will give a protocol error on the call. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QueueMemberStatus
Lee Jenkins wrote: I've been poking for the definition of QueueMemberStatus and all the source file indicates is that it is a integer member of the member structure. Anyone know where I can find the CONSTANTS definitions? OK, I didn't know this, but QueueMemberStatus returns the same codes for channel status as defined in devicestate.h. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? less and less :) it was popular for data in the mid 90's, adsl killed it, its practically unheard of for voice, its tariffed though so its still offered but you really have to know what rock to turn over to order it. In Canada at least there is no such thing as a residential service or more people would be hacking around with it, but paying the commercial price is a bit steep to play with to try hardware or software. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Mark Phillips wrote: Damn!!! Beat me to it ;-} As an Englishman now living in New Jersey (strangely nowhere near an exit) I have to say that the local idiom and accent leaves a significant amount to be desired. Terms like New Joisey, Shuwa ,wadder, badderies, congradulations etc make me wonder if I'm in an English speaking country at all. I've heard better English spoken in Nigeria. And I've heard unintelligible English in parts of England. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
Andrew Kohlsmith wrote: On Tuesday 03 July 2007 9:47 pm, Joe acquisto wrote: We get to do that, because, back in the late 1700's . . . we won. Hey man, I'm Canadian... We've got our own set of funny accents, and don't get us started on the Quebecois. Not even the Parisians can understand THEM! :-) Parisians only PRETEND not to understand them. French is French, even in Québec. You might be referring to joual, which is a street dialect. And France has its share of regional dialects unintelligible to the Parisians. I suspect the Parisian ignorance is feigned; a bit of snobbery vis-à-vis the colonies. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best wifi IP phone for asterisk
ODM is the same as WIP300. Probably the same phone as the D-Link Dual mode. On 7/3/07, Ron Arts [EMAIL PROTECTED] wrote: You might want to look at the Pirelli Dual Mode DP-L10. I tested one, and sound quality and stability are much better than the Nokia E61 or of any other wiFi phone I tested. Ron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call still in queue after Reject Signal
Hi, I have a queue with maxlen=1, and when i make a call, the call enters into the queue, but he doesn't exit from it after a reject signal received from the agent?? please, have you any idea how to remove calls after a reject signal??? Thanks. Rachid ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google acquires Grand Central
Jaswinder Singh wrote: Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . If this happens I am going back to tin cans and string. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrade Asterisk
Hi! Just ashort question - obviously I am too stupid too find the answer on the net. :-) I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have to do? Just install it over the existing version? Do I need to backup the configuration? Will I need to reconfigure the source or will the new version import my old settings? Will I need to update Zaptel and Libpri too? Argh - I installed like 50 asterisk systems but this one is the first production machine with issues so heavy that I have to upgrade it. Please point me to a update/upgrade howto etc. if available on the net. Thanks a ton Christian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
Doug Lytle wrote: Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! Mine arrive instantly. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Dave Donovan wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. On 6/27/07, *Joe Greco* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. . Has anyone else seen a working hardware solution that didn't cost an arm and a leg? It seems to me that a BRI card should cost less than $100. I think I remember a German friend telling me that they go for around $40 dollars. I know that I can get a BRI with voice service out of Bell. I think they have to provide it because of the CRTC tariffs. The pricing from Telus wasn't that bad; it used to be much more expensive. The thing that has stopped me from trying it in the past is the uncertainty around hardware. Do I understand correctly that NA (North American?) BRI is different from the European version and that European hardware won't work? If I could get a card for a few hundred bucks then I'd be willing to give this a shot. Unfortunately, I can't afford a few grand for the Adtran setup described, although it does sound cool and the BRI-PRI conversion approach is a clever way of overcoming the hardware scarcity. Sangoma just came out with a BRI card. What's missing is a 2B1Q driver. I am trying to persuade them to write a 2B1Q driver for it. Dave -- if you would get in touch with them and add your voice to the chorus, it wouldn't hurt. I don't know what the price on the Sangoma card is, but it would be a couple hundred dollars, I suspect. Nowhere near the Adtran contraption previously described. The hardware for most cards is the same; the signalling is managed by the driver. The difference is just in the signalling; the technology is basically the same. For the number of times that I see people trying to get digital style features out of analog lines, and banging their head against the wall, I'd love to get a BRI working and be able to tell you all how it worked out. There's an untapped market here, for sure. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
On Wed, 04 Jul 2007 12:46:35 -0400, Jon Pounder wrote: Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? less and less :) it was popular for data in the mid 90's, adsl killed it, its practically unheard of for voice, its tariffed though so its still offered but you really have to know what rock to turn over to order it. In Canada at least there is no such thing as a residential service or more people would be hacking around with it, but paying the commercial price is a bit steep to play with to try hardware or software. Actually, there is one application where BRI is still fairly common.real-time audio recording sessions for TV. The voice talent that TV stations hire to do their announcements, which happen almost daily, usually use BRI connected hardware. This became the norm as it was the most reliable connection possible in the mid-90's when this workflow model was established. Here in Texas the state regulatory body has decent prices listed for BRIs, certainly cheaper than POTS lines with a couple of features. I looked for hardware that was capable of handling US signaling and was never able to find anything affordable. If Xorcom could pull it off then that'd be a nice experiment to try. It could significantly reduce the need for small FXO interfaces, which were always troublesome as well. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Jon Pounder wrote: Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? less and less :) For data, yes. it was popular for data in the mid 90's, adsl killed it, its practically unheard of for voice, its tariffed though so its still offered but you really have to know what rock to turn over to order it. I just called the Business Services line at Telus and said I want pricing on a BRI. I got You mean a PRI? I said, No, I mean a BRI -- Basic Rate Interface. I don't need more than a few channels. I got Okay, I'll look into it for you. I got a call a couple of hours later with pricing. Amazing. In Canada at least there is no such thing as a residential service or more people would be hacking around with it, but paying the commercial price is a bit steep to play with to try hardware or software. If you can afford two business lines with call waiting and caller ID, you can afford a BRI. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Jon Pounder wrote: Quoting Dave Donovan [EMAIL PROTECTED]: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. I hear you loud and clear - I am in much the same situation. from what I know about the cards (which isn't much) there are 2 types just like there are modems and winmodems, the hard bri cards understand all the signalling in hardware (its different between na/euro much like a modem has different standards it can adhere to when connecting to different remotes even though they are all carried on the same physical type of line) so if you get a hardware card you need one that understands our signalling here. The soft cards are pretty much universal since they are implemented in software only, but you need a driver that understands the NA signalling - I am not sure about the state of this situation right now. just like their analog cousins the soft cards are cheap, the hard cards are costly, and the NA ones rare. Okay, guys -- since this keeps coming up: http://sangoma.com/datasheets/A500BRI This card is very new. I don't have a price but I don't think it will be much more expensive than the A200 with some modules in it. I have been told -- tentatively -- that no 2B1Q driver was planned for this card, but it's early days and if you guys make some noise, I think Sangoma could be persuaded to write one. My hope is stoked by the fact that no mention is made of signalling type for this card. Anyway -- it uses the wanpipe platform and would work just fine with Asterisk. (I will be visiting Sangoma in late July and would like to have something to bring them to encourage a 2B1Q driver -- please contact me off-list to discuss.) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
On Wed, Jul 04, 2007 at 12:46:35PM -0400, Jon Pounder wrote: Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? less and less :) What signalling? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Stephen Bosch [EMAIL PROTECTED]: http://www.sangoma.com/datasheets/A500BRI is that the card you mean ? it says it supports asterisk Dave Donovan wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. On 6/27/07, *Joe Greco* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. . Has anyone else seen a working hardware solution that didn't cost an arm and a leg? It seems to me that a BRI card should cost less than $100. I think I remember a German friend telling me that they go for around $40 dollars. I know that I can get a BRI with voice service out of Bell. I think they have to provide it because of the CRTC tariffs. The pricing from Telus wasn't that bad; it used to be much more expensive. The thing that has stopped me from trying it in the past is the uncertainty around hardware. Do I understand correctly that NA (North American?) BRI is different from the European version and that European hardware won't work? If I could get a card for a few hundred bucks then I'd be willing to give this a shot. Unfortunately, I can't afford a few grand for the Adtran setup described, although it does sound cool and the BRI-PRI conversion approach is a clever way of overcoming the hardware scarcity. Sangoma just came out with a BRI card. What's missing is a 2B1Q driver. I am trying to persuade them to write a 2B1Q driver for it. Dave -- if you would get in touch with them and add your voice to the chorus, it wouldn't hurt. I don't know what the price on the Sangoma card is, but it would be a couple hundred dollars, I suspect. Nowhere near the Adtran contraption previously described. The hardware for most cards is the same; the signalling is managed by the driver. The difference is just in the signalling; the technology is basically the same. For the number of times that I see people trying to get digital style features out of analog lines, and banging their head against the wall, I'd love to get a BRI working and be able to tell you all how it worked out. There's an untapped market here, for sure. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) And received at 10:58 CDT (16:58 UTC)... May have more to do with where it is sent from than the list itself. John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 12:46:35PM -0400, Jon Pounder wrote: Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? less and less :) What signalling? I forget the actual name but its the nortel/dms100 signalling -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Tzafrir Cohen wrote: On Wed, Jul 04, 2007 at 12:46:35PM -0400, Jon Pounder wrote: Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? less and less :) What signalling? 2B1Q. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: http://www.sangoma.com/datasheets/A500BRI is that the card you mean ? it says it supports asterisk Yes, that's the card I mean and yes, it supports Asterisk. The problem: I have been told -- again, this is tentative -- that there were no plans for a 2B1Q driver for it. I think if Sangoma heard from enough interested North American users, they would write the driver. I, for one, would be a customer. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
John Faubion wrote: Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) And received at 10:58 CDT (16:58 UTC)... May have more to do with where it is sent from than the list itself. If you view the full headers, it shows the time received at each hop from original sender, through the mailing list to the recipient (you). This should help figure out where the delay is. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
On 7/4/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jul 04, 2007 at 12:46:35PM -0400, Jon Pounder wrote: Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Jul 04, 2007 at 11:05:52AM -0500, Joe Greco wrote: Sorry I'm a little late to the thread but this question has puzzled me as well. My key thing for me is hardware. In the UK but ... Cheap BRI card... To use it in the US, you need it to support US signalling. What do they use for BRI in the US? less and less :) What signalling? I'm not an engineer so I don't have the specific answer. I suppose you're thinking about your BRI product? http://xorcom.com/products/astribank_bri It would be great if we could get something like this working. Typically, telcos in Canada and the US are using either Lucent 5ESS or Nortel DMS (100, 250, 500) type switches. I imagine the specs of that equipment dicates how the BRI will work. I could do a bit of research and see if I can dig up something. Is there a specific set of questions I should be asking other than what signalling? Dave -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got 404 when route calls through Asterisk to another proxy
Check the entry in your extension.conf , that if you have made the entry for phone B , on ur asterisk from proxy A. Jason Ma [EMAIL PROTECTED] wrote: Buddies, Here is my test case softphoneA--proxyA---Asterisk--proxyB--softphoneB | softphone C softphoneC is registered with Asterisk. I can place call from softphoneA to softphoneC,and also can make calls from softphoneC to softphoneB. But when I placed call from softphoneA to softphoneB,I got 404 not found that sent from Asterisk.What wrong with my configuration?Need I do someting to let Asterisk can receive inbound calls and chosse proper routes and send it our?Please advise.Thanks a lot. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Stephen Bosch [EMAIL PROTECTED]: Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: http://www.sangoma.com/datasheets/A500BRI is that the card you mean ? it says it supports asterisk Yes, that's the card I mean and yes, it supports Asterisk. The problem: I have been told -- again, this is tentative -- that there were no plans for a 2B1Q driver for it. that sounds a little strange to me given they are located in north america themselves. I think if Sangoma heard from enough interested North American users, they would write the driver. I, for one, would be a customer. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] List delays
Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! Interesting, since I am only able to see the replies from the archives, I took note of the headers as Drew Gibson suggested. Looks like mail is getting held up between INXS.digium.internal and lists.digium.com INXS.digium.internal received it the first of July, lists.digium.com received it on the 4th. drdos.info (ME) received it from lists.digium.com on that same day (Today). Attached: From - Wed Jul 04 14:19:03 2007 X-Account-Key: account2 X-UIDL: 86007 X-Mozilla-Status: 0001 X-Mozilla-Status2: X-Mozilla-Keys: Return-Path: [EMAIL PROTECTED] Received: from lists.digium.com ([192.168.145.1]) by drdos.info with hMailServer ; Wed, 4 Jul 2007 14:11:14 -0400 Received: from lists.digium.com ([216.207.245.17] helo=lists.digium.com) by ASSP-nospam; 4 Jul 2007 14:12:33 -0301 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I53rV-0005GQ-A7; Sun, 01 Jul 2007 13:09:29 -0500 Received: from exprod8mx6.postini.com ([64.18.3.106] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I53rM-0005Fx-Sc for asterisk-users@lists.digium.com; Sun, 01 Jul 2007 13:09:21 -0500 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Stephen Bosch [EMAIL PROTECTED]: Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: http://www.sangoma.com/datasheets/A500BRI is that the card you mean ? it says it supports asterisk Yes, that's the card I mean and yes, it supports Asterisk. The problem: I have been told -- again, this is tentative -- that there were no plans for a 2B1Q driver for it. I think if Sangoma heard from enough interested North American users, they would write the driver. doesn't it seem strange they went to the trouble to get the fcc certification (as per their site) if it doesn't understand the north american signalling ? If someone already has a customer relationship with them, ask straight out does it work in US/Canada with the BRI available here with asterisk. if they can't answer, ask for a demo card to try (as long as you have a line to try with.) PS - to whoever it was that said they got the telus quote ... what province was that in ? In Ontario Telus is just not interested in selling BRI (could just be the sales rep too, he is kind of lazy. here though its a resale of a bell canada service though, and btw whew that telus/bell merger plan last week died quickly thank goodness, combining bad and worse never makes anything good) I, for one, would be a customer. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
John Faubion wrote: Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) Read the message headers. They tell you exactly what time and by what host the message was received at. There is no mystery here, it is all in the message headers. If your e-mail program does not allow you to view the headers then get an e-mail client that does not suck. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: http://www.sangoma.com/datasheets/A500BRI is that the card you mean ? it says it supports asterisk Yes, that's the card I mean and yes, it supports Asterisk. The problem: I have been told -- again, this is tentative -- that there were no plans for a 2B1Q driver for it. I think if Sangoma heard from enough interested North American users, they would write the driver. doesn't it seem strange they went to the trouble to get the fcc certification (as per their site) if it doesn't understand the north american signalling ? If someone already has a customer relationship with them, ask straight out does it work in US/Canada with the BRI available here with asterisk. I just got off the phone with my sales rep. It appears I'm the third person today to ask about this. (I wonder why?) The answer is no it will not work in NA. Their reasoning being that with limited resources they went after the biggest market. I get the impression that there are no plans to write a North American driver as the demand seems to be very low. -- Jeff Davis Netsource Consulting Richmond, VA ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
On 7/4/07, Jon Pounder [EMAIL PROTECTED] wrote: I think if Sangoma heard from enough interested North American users, they would write the driver. doesn't it seem strange they went to the trouble to get the fcc certification (as per their site) if it doesn't understand the north american signalling ? I wonder if this is issue is largely limited to to Canada. (thus limiting the market) In the states I think you can get PRI for around $250. Am I right? In Canada, you have to have about 9 or 10 lines to justify a PRI. At $250, the cost and added features could justify PRI at around 4 lines. Mind you, that still leaves a whole tonne of systems at the 4 lines and under mark. The next time a Sangoma staffer comes out to a Toronto Asterisk User Group meeting, I'll get 'em all tipsy and make them sign something about committing to supporting the North American BRI. :-) PS - to whoever it was that said they got the telus quote ... what province was that in ? In Ontario Telus is just not interested in selling BRI (could just be the sales rep too, he is kind of lazy. here though its a resale of a bell canada service though, and btw whew that telus/bell merger plan last week died quickly thank goodness, combining bad and worse never makes anything good) I think that's a regulatory thing. In Ontario, Telus is considered a competitor not the incumbent. I think that only the incumbent in any area is bound by the tariffs. Out west it would be the opposite, Telus would be the incumbent there. For anyone who' interested, I decided to read up a bit and found an interesting primer. http://www.ralphb.net/ISDN/ Tzafrir: It discusses North America specifically and identifies the specific ITU Q-Series documents that define the various protocol layers. Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to find the file zaptel.conf aftercompiling asterisk and zaptel
Your post was delayed three days before I saw it, so you probably have found zaptel.conf by now. It's under /etc/, not /etc/asterisk/, because its use is not limited to asterisk. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Sunday, July 01, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Not able to find the file zaptel.conf aftercompiling asterisk and zaptel Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4 and zaptel 1.4 ? Regards - ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: + (965) 9849460 Yahoo ID: [EMAIL PROTECTED] MSN ID: [EMAIL PROTECTED] Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Panasonic TDA200
The obviously th settings in asterisk are wrong On 7/4/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Tue, 2007-07-03 at 23:50 -0400, C F wrote: Change it to ISDN. There is no point in not to, what card do you have in the TDA200? A PRI or or just T/E1? Since it's too differenct cards on the TDA200. In fact accroding to Panasonic CallerID isn't supported on none PRI, although some have gotten it to work. The card on the TDA200 is a MFC/R2 E1 card, you cannot use ISDN. Obviously the customer is not willing to pay for an ISDN card. The TDA received CID from the phone company without any problem, but it will not receive it from Asterisk. If you enable CID on the TDA it will give a protocol error on the call. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
I wonder if this is issue is largely limited to to Canada. (thus limiting the market) In the states I think you can get PRI for around $250. Am I right? In Canada, you have to have about 9 or 10 lines to justify a PRI. At $250, the cost and added features could justify PRI at around 4 lines. Mind you, that still leaves a whole tonne of systems at the 4 lines and under mark. No way.message rates lines hover at $350, and flat rate's run $450-$500 or so. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reliaclear.com
Anyone hear of them before ? they setup shop quite literally just down the road from me, yet it took an article in the paper today to realize they exist. Web Site has absolutely no technical information on it, not even so much as to say you need an internet connection to use the service. Is this whole thing real or just vapourware ? Will it work with asterisk and if anyone has tried it what is the quality like ? Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Darren Wright [EMAIL PROTECTED]: I wonder if this is issue is largely limited to to Canada. (thus limiting the market) In the states I think you can get PRI for around $250. Am I right? In Canada, you have to have about 9 or 10 lines to justify a PRI. At $250, the cost and added features could justify PRI at around 4 lines. Mind you, that still leaves a whole tonne of systems at the 4 lines and under mark. how do you come up with that ? (what are you assuming for line and pri costs ?) when we had a bunch of lines (10+) through an att reseller in toronto we were paying $35 each with all the features and I have never seen any sort of t1 less than about $700 in Ontario, so that works out to about 20 lines for breakeven - at that point what are you really gaining except making it easier on the telco to deliver, yet you have all your eggs in one basket and if there is a hardware or physical plant issue you are completely down. No way.message rates lines hover at $350, and flat rate's run $450-$500 or so. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ANNOUNCEMENT : A2Billing (Asterisk2Billing) - V1.3.0 STABLE (Yellowjacket)
I am pleased to announce the new version of Asterisk2Billing, V1.3.0 STABLE (Yellowjacket) PROJECT URL : http://trac.asterisk2billing.org A2Billing has completely re-written some its modules such as : Invoicing, template management with Smarty, the call-back, added new methods of online payment integration with Moneybookers and Authorize.net in addition to Paypal. A2Billing have also improved the rating engine, giving the operator the ability to create Free Minutes packages to certain destinations. Additional reporting functions and alarms have also been added in the interests of revenue protection including automatic emails for High or Low ASR (Answer Seize Ratio), ALOC (Average Length of Call) and CIC (Consecutive Incomplete Calls) Alarms and many more good stuff... As you can see, we decided to take a better direction for this project and make it easier for the community to contribute and participate to our development. Trac is providing Wiki, Ticket system, Timeline, etc... A public SVN is available : http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Development SVN server : http://svn.a2billing.net/svn/asterisk2billing/ We hope that this will make A2Billing more transparent and easier to contribute. As usual, the forum the online demo are still available : FORUM - http://forum.asterisk2billing.org/ DEMO - http://demo.asterisk2billing.org/ CALL-LABS : http://www.call-labs.com Register and try Call-Labs, our A-Z provider! If you are looking for A-Z termination at good rates, this could be your solution! Please don't forget to make a donation if you find our software useful and want to support the development of Asterisk2Billing : http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Donate%20to%20A2Billing Kind regards, /Areski -- -- ~ - Belaid Arezqui ( [EMAIL PROTECTED] ) 'v'- CEO/CTO A2Billing ; Yellowjacket Whisperer // \\ - Cell Phone. : (+34) 650 78 43 55 (Spain, GMT+1hr) /( )\ - http://asterisk2billing.org/ - http://www.areski.net ^`~'^ - LinkedIN : http://www.linkedin.com/in/areski ___ -- To support A2Billing - paypal moneybookers : [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reliaclear.com
***No-Borders Numbers http://reliaclear.com/en-usa/noborders.html *View Current Service Areas in: Canada http://reliaclear.com/en-usa/noborders.html#Canada | United States http://reliaclear.com/en-usa/noborders.html#United_States I guess it is no borders as long as you only mean the US and Canadian border. -- Chris Mason Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670 International: (305) 704-7249 Fax: (815)301-9759 Yahoo IM only: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reliaclear.com
Quoting Chris Mason (Lists) [EMAIL PROTECTED]: ***No-Borders Numbers http://reliaclear.com/en-usa/noborders.html *View Current Service Areas in: Canada http://reliaclear.com/en-usa/noborders.html#Canada | United States http://reliaclear.com/en-usa/noborders.html#United_States I guess it is no borders as long as you only mean the US and Canadian border. well I tried the signup process just to see what other information I could get out of the site - before I even get a monthly total or mention the hardware I need, they want my bank account info for ACH transactions - NOT quite! 1) any 2 bit operation can get a visa/mc merchant account so where is theirs ? 2) you are not even allowed to collect that sort of information under credit card processing rules until the customer is presented with the total amount to be billed. yikes - let someone else try this one out first. The article in the paper sounded good about hiring 2000+ people in the next couple years, but we'll wait and see I guess. -- Chris Mason Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670 International: (305) 704-7249 Fax: (815)301-9759 Yahoo IM only: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reliaclear.com
I love the smell of lemonade in the morning -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with SIP Registration on VPN Link
Hi, We are having major problems with a remote site that links to the head office via a VPN tunnel. The phones will register fine and work for a few minutes to hours but then will drop their connection and will no register to asterisk even with a restart of the phone. We have 2 other remote sites that work exactly same and they are not having any issues so i believe it has to be be something to do with the network rather then asterisk but this is the sip debug for a phone trying to register. Any idea where i should start to look as this has me totally confused as obviously the phones can communicate with asterisk at all times just something is causing the registration to get screwed up. Jul 4 09:43:46 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Transmitting (NAT) to 192.168.12.63:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63 From: Edmonton Boardroom 1 sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805 To: sip:[EMAIL PROTECTED];user=phone;tag=as4d6893cc Call-ID: [EMAIL PROTECTED] CSeq: 10005 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=48f69f92, stale=true Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms cnsmavs1*CLI -- SIP read from 192.168.12.63:5060: REGISTER sip:192.168.10.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15 From: Edmonton Boardroom 1 sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805 To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Supported: path Authorization: Digest username=763, realm=asterisk, algorithm=MD5, uri=sip:192.168.10.12, nonce=587da437, response=4bd29b9213057e3e2f3a5270748fbe85 all-ID: [EMAIL PROTECTED] CSeq: 10005 REGISTER Expires: 3600 User-Agent: Grandstream GXP2000 1.1.2.23 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,M ESSAGE Content-Length: 0 --- (14 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.12.63 : 5060 (NAT) Transmitting (NAT) to 192.168.12.63:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63 From: Edmonton Boardroom 1 sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 10005 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Jul 4 09:43:48 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Transmitting (NAT) to 192.168.12.63:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63 From: Edmonton Boardroom 1 sip:[EMAIL PROTECTED];user=phone;tag=65cbed22c3593805 To: sip:[EMAIL PROTECTED];user=phone;tag=as4d6893cc Call-ID: [EMAIL PROTECTED] CSeq: 10005 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=750fc224, stale=true Content-Length: 0 Nathan Dennis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Xorcom Bri and asterisk crashes
We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time will pass before a crash out. The first time I didn't have much logging so I didn't get anything to work with. I have since turned on debugging and following is the logs from the time of the last crash. Can anyone point out where the problem may lay, suggested updates or changes? Jul 4 11:56:51 DEBUG[20042] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 4 11:56:54 DEBUG[20042] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jul 4 11:56:55 VERBOSE[20050] logger.c: -- Accepting voice call from '' to '40312688' on channel 0/2, span 5 Jul 4 11:56:55 DEBUG[20050] chan_zap.c: Enabled echo cancellation on channel 14 Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'CID withheld' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, CALLERID(name)=Old Main Line:CID withheld) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Goto(Zap/14-1, mainq|q|1) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,1) Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'false' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, NightMode=false) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Expression result is '0' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIf(Zap/14-1, 0?afterhoursq|q|1) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Not taking any branch Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIfTime(Zap/14-1, 8:00-17:30|mon-fri|*|*|?businesshours) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,5) Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, __ALERT_INFO=http://www.example.com;info=MainQ) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Queue(Zap/14-1, mainq1|twr|||10) in new stack Jul 4 11:56:55 DEBUG[20295] chan_zap.c: Requested indication 3 on channel Zap/14-1 Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Called Local/[EMAIL PROTECTED] Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, Extension=700) in new stack Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, __ALERT_INFO=http://www.example.com;info=MainQ) in new stack Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/700||tw) in new stack Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on RTP to 524288 Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on VRTP to 524288 Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Outgoing Call for 700 Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Called 700 Jul 4 11:56:55 DEBUG[20042] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jul 4 11:56:55 DEBUG[20042] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jul 4 11:56:55 VERBOSE[20298] logger.c: -- SIP/700-09530a90 is ringing Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Local/[EMAIL PROTECTED],1 is ringing Jul 4 11:56:56 DEBUG[20042] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 4 11:56:59 DEBUG[20042] chan_sip.c: Acked pending invite 102 Jul 4 11:56:59 DEBUG[20042] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jul 4 11:56:59 DEBUG[20042] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Jul 4 11:56:59 DEBUG[20027] channel.c: Avoiding initial deadlock for 'SIP/700-09530a90' Jul 4 11:56:59 VERBOSE[20298] logger.c: -- SIP/700-09530a90 answered Local/[EMAIL PROTECTED],2 Jul 4 11:56:59 DEBUG[20295] app_queue.c: Dunno what to do with control type -1 Jul 4 11:56:59 VERBOSE[20295] logger.c: -- Local/[EMAIL PROTECTED],1 answered Zap/14-1 Jul 4 11:56:59 DEBUG[20295] chan_zap.c: Set option TONE VERIFY, mode: MUTECONF(1) on Zap/14-1 Jul 4 11:56:59 DEBUG[20295] chan_zap.c: No echo training requested Jul 4 11:56:59 DEBUG[20298] channel.c: Planning to masquerade channel SIP/700-09530a90 into the structure of Local/[EMAIL PROTECTED],1 Jul 4 11:56:59 DEBUG[20298] channel.c: Done planning to masquerade channel SIP/700-09530a90 into the structure of Local/[EMAIL PROTECTED],1 Jul 4 11:56:59 DEBUG[20295] channel.c: Got clone lock for masquerade on 'SIP/700-09530a90' at 0x952ab64 Jul 4 11:56:59 DEBUG[20298] chan_local.c: Not posting to queue since already masked on 'Local/[EMAIL PROTECTED],2' Jul 4 11:56:59 DEBUG[20295] channel.c: Putting channel SIP/700-09530a90
[asterisk-users] FW: Openmoko ads now on youtube
Hi Guys, Great example of how some of the OpenMoko guys are getting the word out there. If you don't know about OpenMoko you can review it here. http://deancollinsblog.blogspot.com/2006/11/fic-gta001.html http://deancollinsblog.blogspot.com/2006/11/open-phones.html http://deancollinsblog.blogspot.com/2007/06/home-brew-startrek-communica tor.html Otherwise check out some of the links below. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:community- [EMAIL PROTECTED] On Behalf Of Adam Krikstone Sent: Wednesday, 4 July 2007 4:59 PM To: [EMAIL PROTECTED] Subject: Openmoko ads now on youtube Good and bad, here are some ads for openmoko and the neo1973 I did. Sorry for the bad quality on some but there aren't many videos or pictures of the neo1973 besides the wiki. I stayed with the free your phone, aspect since advertising linux to the public is not going to work. I can make better ones if someone can get me high res photos and video (720x480 and above). Playlist: http://www.youtube.com/view_play_list?p=472DE700A3CC70A4 Individual: http://www.youtube.com/watch?v=DCQ7dmGuAU8 http://www.youtube.com/watch?v=tQPjfUqp-dk http://www.youtube.com/watch?v=4qP-K1HOMHk http://www.youtube.com/watch?v=S--2HeQqjq4 http://www.youtube.com/watch?v=dpwxzEopg60 http://www.youtube.com/watch?v=EuG2hYiO9AU http://www.youtube.com/watch?v=lGjY7tigdkA http://www.youtube.com/watch?v=YR4ezMgRlWo http://www.youtube.com/watch?v=OZC3mjRW5Tg http://www.youtube.com/watch?v=GxsVFG7jHI8 http://www.youtube.com/watch?v=62kLhNngE20 ___ OpenMoko community mailing list [EMAIL PROTECTED] http://lists.openmoko.org/mailman/listinfo/community ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Group Function
I cant figure this out. I have seen this same example many places but the
group never gets incremented. Am I missing something?
exten = 99,1,Set(GROUP(99) = G99)
exten = 99,2,GotoIf($[${GROUP_COUNT(99)}0]?103)
exten = 99,3,dial(SIP/qoqieoeiwq)
exten = 99,103,Hangup
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[asterisk-users] Need Help in Asterisk BLF/Presence/Hints
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if
somebody help me.
Regards
FArooq
**
1
**
in my extension_additional.conf
[ext-local]
include = ext-local-custom
exten = 501,1,Macro(exten-vm,501,501)
exten = 501,n,Hangup
exten = 501,hint,SIP/501
exten = ${VM_PREFIX}501,1,Macro(vm,501,DIRECTDIAL)
exten = ${VM_PREFIX}501,n,Hangup
exten = 502,1,Macro(exten-vm,502,502)
exten = 502,n,Hangup
exten = 502,hint,SIP/502
exten = ${VM_PREFIX}502,1,Macro(vm,502,DIRECTDIAL)
exten = ${VM_PREFIX}502,n,Hangup
exten = 503,1,Macro(exten-vm,503,503)
exten = 503,n,Hangup
exten = 503,hint,SIP/503
exten = ${VM_PREFIX}503,1,Macro(vm,503,DIRECTDIAL)
exten = ${VM_PREFIX}503,n,Hangup
; end of [ext-local]
***
2
**
SIP_additional.conf
one of my extension is configured as
--
[507]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=507 at device
host=dynamic
dtmfmode=rfc2833
dial=SIP/507
context=from-internal
canreinvite=no
subscribecontext = ext-local
notifyringing = yes
callerid=device 507
3
ext 501 phone is configured with complete contact directory.
Buddywatch was enabled in the polycom contact directory
using config like below
item
lnDoe/ln
fnJohn/fn
ct507/ct
sd1/sd
rt1/rt
dc /
ad0/ad
ar0/ar
bw1/bw
bb0/bb
/item
**
Results
***
localhost*CLI show hints
localhost*CLI
-= Registered Asterisk Dial Plan Hints =-
507 : SIP/507 State:Unavailable Watchers 0
506 : SIP/506 State:Unavailable Watchers 0
505 : SIP/505 State:Unavailable Watchers 0
504 : SIP/504 State:IdleWatchers 0
503 : SIP/503 State:Unavailable Watchers 0
502 : SIP/502 State:IdleWatchers 0
501 : SIP/501 State:IdleWatchers 0
- 7 hints registered
localhost*CLI
localhost*CLI sip show subscriptions
Peer UserCall ID ExtensionLast state Type
0 active SIP subscriptions
localhost*CLI
--
--
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Re: [asterisk-users] North American voice BRI - Informal survey
Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: http://www.sangoma.com/datasheets/A500BRI is that the card you mean ? it says it supports asterisk Yes, that's the card I mean and yes, it supports Asterisk. The problem: I have been told -- again, this is tentative -- that there were no plans for a 2B1Q driver for it. I think if Sangoma heard from enough interested North American users, they would write the driver. doesn't it seem strange they went to the trouble to get the fcc certification (as per their site) if it doesn't understand the north american signalling ? Certifications are often handled by agencies that submit a device for certification with multiple certifying bodies. It's as much trouble to submit for EC certification as it is for FCC, so if you streamline the process to avoid replicating effort, you can get a lot more done for less. Having FCC certification does not automatically lead to a product being useful in the United States. If someone already has a customer relationship with them, ask straight out does it work in US/Canada with the BRI available here with asterisk. I can tell you right now -- the answer is no. The driver does not support it. I'll change my story if I'm told otherwise. if they can't answer, ask for a demo card to try (as long as you have a line to try with.) I'm not going to waste my time if the guy who engineered the card is telling me it won't work. It needs 2B1Q signalling support in the driver. If the will is there it's not a big deal to add it, but as it stands, it's not going to work. PS - to whoever it was that said they got the telus quote ... what province was that in ? In Ontario Telus is just not interested in selling BRI (could just be the sales rep too, he is kind of lazy. here though its a resale of a bell canada service though, and btw whew that telus/bell merger plan last week died quickly thank goodness, combining bad and worse never makes anything good) You'd be disinterested too, if you were reselling your number 1 rival's service. I guarantee you -- Telus' service requests go to the bottom of Bell's inbox. The only reason they handle those at all is because the law says they have to. The quote was for a BRI in Alberta. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Jeff Davis wrote: Jon Pounder wrote: If someone already has a customer relationship with them, ask straight out does it work in US/Canada with the BRI available here with asterisk. I just got off the phone with my sales rep. It appears I'm the third person today to ask about this. (I wonder why?) Your rep at Sangoma? Or your reseller? The answer is no it will not work in NA. Their reasoning being that with limited resources they went after the biggest market. I get the impression that there are no plans to write a North American driver as the demand seems to be very low. This is a real chicken-and-egg problem. More people would get BRI if there were affordable hardware for it. I would like to see them write a NAm driver for it. To get them to take the chance, there have to be enough people willing to purchase the card to make them consider it seriously. The other option is a bounty or community support to get it done. The hardware already exists. The more people make noise about this, the better the chances of that happening. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reliaclear.com
Chris Mason (Lists) wrote: I love the smell of lemonade in the morning Do lemons grow in backyards in Anguilla? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
Eric ManxPower Wieling wrote: John Faubion wrote: Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) Read the message headers. They tell you exactly what time and by what host the message was received at. There is no mystery here, it is all in the message headers. If your e-mail program does not allow you to view the headers then get an e-mail client that does not suck. APPLAUSE -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Xorcom Bri and asterisk crashes
On Thu, Jul 05, 2007 at 09:14:12AM +1000, Nathan Dennis wrote: We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time will pass before a crash out. The first time I didn't have much logging so I didn't get anything to work with. I have since turned on debugging and following is the logs from the time of the last crash. Can anyone point out where the problem may lay, suggested updates or changes? Jul 4 11:56:51 DEBUG[20042] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 4 11:56:54 DEBUG[20042] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jul 4 11:56:55 VERBOSE[20050] logger.c: -- Accepting voice call from '' to '40312688' on channel 0/2, span 5 Jul 4 11:56:55 DEBUG[20050] chan_zap.c: Enabled echo cancellation on channel 14 Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'CID withheld' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, CALLERID(name)=Old Main Line:CID withheld) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Goto(Zap/14-1, mainq|q|1) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,1) Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'false' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, NightMode=false) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Expression result is '0' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIf(Zap/14-1, 0?afterhoursq|q|1) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Not taking any branch Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIfTime(Zap/14-1, 8:00-17:30|mon-fri|*|*|?businesshours) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,5) Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, __ALERT_INFO=http://www.example.com;info=MainQ) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Queue(Zap/14-1, mainq1|twr|||10) in new stack Jul 4 11:56:55 DEBUG[20295] chan_zap.c: Requested indication 3 on channel Zap/14-1 Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Called Local/[EMAIL PROTECTED] Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, Extension=700) in new stack Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, __ALERT_INFO=http://www.example.com;info=MainQ) in new stack Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/700||tw) in new stack Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on RTP to 524288 Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on VRTP to 524288 Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Outgoing Call for 700 Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Called 700 Jul 4 11:56:55 DEBUG[20042] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jul 4 11:56:55 DEBUG[20042] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jul 4 11:56:55 VERBOSE[20298] logger.c: -- SIP/700-09530a90 is ringing Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Local/[EMAIL PROTECTED],1 is ringing Jul 4 11:56:56 DEBUG[20042] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jul 4 11:56:59 DEBUG[20042] chan_sip.c: Acked pending invite 102 Jul 4 11:56:59 DEBUG[20042] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jul 4 11:56:59 DEBUG[20042] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Jul 4 11:56:59 DEBUG[20027] channel.c: Avoiding initial deadlock for 'SIP/700-09530a90' Jul 4 11:56:59 VERBOSE[20298] logger.c: -- SIP/700-09530a90 answered Local/[EMAIL PROTECTED],2 Jul 4 11:56:59 DEBUG[20295] app_queue.c: Dunno what to do with control type -1 Jul 4 11:56:59 VERBOSE[20295] logger.c: -- Local/[EMAIL PROTECTED],1 answered Zap/14-1 Jul 4 11:56:59 DEBUG[20295] chan_zap.c: Set option TONE VERIFY, mode: MUTECONF(1) on Zap/14-1 Jul 4 11:56:59 DEBUG[20295] chan_zap.c: No echo training requested Jul 4 11:56:59 DEBUG[20298] channel.c: Planning to masquerade channel SIP/700-09530a90 into the structure of Local/[EMAIL PROTECTED],1 Jul 4 11:56:59 DEBUG[20298] channel.c: Done planning to masquerade channel SIP/700-09530a90 into the structure of Local/[EMAIL PROTECTED],1 Jul 4 11:56:59 DEBUG[20295] channel.c: Got clone lock for masquerade on 'SIP/700-09530a90' at 0x952ab64 Jul 4 11:56:59 DEBUG[20298]
[asterisk-users] Asterisk console filtering and logging
Hi, Is it possible to filter messages on asterisk console, which was started with -, to see messages only for one extensions? By default there are all messages for any extensions displayed so dialplan debuging is very difficult. Is it possible to log such console messages: ... -- Executing Set(SIP/10.0.0.1-0061f5d0, CDR(userfield)=2422718) -- Executing Dial(SIP/10.0.0.1-0061f5d0, SIP/708,25,tT) ... to file. I can't find any suitable option in logger.conf -- Thanks, Eugene Prokopiev ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrade Asterisk
Hi, Try first installing latest release of libpri, then zaptel Try install asterisk after then. ope you will be able to compile it without any probs. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Stephen Bosch wrote: Your rep at Sangoma? Or your reseller? That wasn't very clear. Sorry. It was Sangoma. (I would be more verbose, but I don't want to spam the list) This is a real chicken-and-egg problem. More people would get BRI if there were affordable hardware for it. I would like to see them write a NAm driver for it. To get them to take the chance, there have to be enough people willing to purchase the card to make them consider it seriously. The other option is a bounty or community support to get it done. The hardware already exists. The more people make noise about this, the better the chances of that happening. If there was a driver available, I'm still not sure how many installs I could sell. Verizon wants to pretend the service doesn't exist, and the largest CLEC in my area doesn't even sell it. (I even offered to buy my CLEC rep dinner and she wouldn't sell it to me.) Without telco support I think that the only real market for this is the DIY crowd. Of course, as you point out, we'll never know how big the market is without a driver. I think that the only real incentive for Sangoma to write a driver for an unproven market would be if there were a community driver available, and the cards start selling. The addition of a manufacturer supplied and supported driver would likely increase sales. -- Jeff Davis Netsource Consulting Richmond, VA ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need advice to get wcte11xp and wcfxo to load
Anyone? -Original Message- From: [EMAIL PROTECTED] on behalf of Wai Wu Sent: Wed 7/4/2007 12:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Need advice to get wcte11xp and wcfxo to load I have a X100P and a TE110P in my Asterisk box. I can either get the X100P or the TE110P to work, but never both. Here's my zaptel.conf span=1,0,0,d4,ami em=1-24 fxsls=25 When I load wcte11xp and wcfxo, I will get this error. [EMAIL PROTECTED] etc]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed Anyway to get both cards working? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade Asterisk
Yes that is write order . libpri then zaptel then asterisk . Remember that zaptel compilation is not required if you are using asterisk for voip only environment .But it's always good to install it before asterisk if you want to use conferencing abilities of asterisk . Regards, Jaswinder Singh On 05/07/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Hi, Try first installing latest release of libpri, then zaptel Try install asterisk after then. ope you will be able to compile it without any probs. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
