[asterisk-users] REALTIME in 1.2
I am trying to change a 1.6 realtime statement into a 1.2 realtime
statement and I know much has changed. I wish I could just upgrade, but
alas not right now.
exten =x,n,Set(NULL1=${REALTIME(schedules,id,${SCHEDULE})})
comes back with
pbx.c:1371 ast_func_read: Function REALTIME not registered
I am not stuck with realtime, I just have a mysql database with info
that changes and needs to update the dialplan accordingly.
Jason
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[asterisk-users] Recording music in Queue
I know that this is a feature but I would like to have the hold music recorded while a person is on hold. So I know the agent put them on hold and not just muted. I have monitor-join=yes monitor-format=wav in my queues.conf any ideas? Per http://www.asteriskguru.com/tutorials/queues_conf.html The best part is no recording will be initiated while the people are listening to music on hold Jason Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] D-Channel Span Up without Down
I am getting a bunch of Primary D-Channel on span 1 up but there was not a down message before that. Is this normal? Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Issue
It seems that my realtime is not assigning channel variables correctly.
INFO
Asterisk 1.6.0.26
Exten.conf
exten = _X.,1,NoOp()
exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})
exten = _X.,4,NoOp(DEVICE is ${DEVICE})
exten = _X.,5,NoOp(USERNAME is ${USERNAME})
exten = _X.,6,NoOp(username is ${username})
CLI
-- Executing [...@default:1] NoOp(SIP/1156-55ce, ) in new
stack
-- Executing [...@default:2] Set(SIP/1156-55ce,
DEVICE=SIP/1156) in new stack
-- Executing [...@default:3] Set(SIP/1156-55ce,
NULL=username=john.smith,name=John
Smith,department=Dept_A,routable=no,extension=1234,device=SIP/1156,voice
mail=no,monitor=yes,visible=yes,date_modified=2010-02-09 14:12:01,) in
new stack
-- Executing [...@default:4] NoOp(SIP/1156-55ce, DEVICE is
SIP/1156) in new stack
-- Executing [...@default:5] NoOp(SIP/1156-55ce, USERNAME is )
in new stack
-- Executing [...@default:6] NoOp(SIP/1156-55ce, username is )
in new stack
So I can see it is getting info from the database in Line 3
But only the direct set variable command (Line 2) and Result (Line 4)
work
Lines 5 and 6 do not get the john.smith assigned
Help
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[asterisk-users] sip issue with one way audio
I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. any Ideas? Jason ___ --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] 2 Digit Issue
I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2
digits into the dialplan.
error
-- Invalid extension '81' in context 'impact' on
SIP/207.174.111.34-b77167f8
I pressed 8107
and ideas
my dial plan is (part of it)
[impact]
exten=s,1,Answer()
exten=s,n,Set(CALLERID(name)=Impact - ${CALLERID(number)})
exten=s,n,Background(IMPACT)
exten=s,n,WaitExten(10)
exten=s,n,Background(OP)
exten=s,n,Dial(Sip/office2,20)
exten=s,n,Voicemail([EMAIL PROTECTED],u)
exten=9,1,Background(IPM_MENU)
exten=9,n,WaitExten(10)
exten=9,n,Goto(0,1)
exten=0,1,Goto(inside,115,1)
exten=i,1,Goto(s,3)
exten=a,1,Goto(s,3)
include=inside
[inside]
exten=102,1,Macro(users,SIP,Comp4,[EMAIL PROTECTED])
exten=111,1,Wait(1)
exten=111,2,Playback(Randy)
exten=111,3,Dial(Sip/Randy,20)
exten=111,4,Goto(111-${DIALSTATUS},1)
exten=111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
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Re: [asterisk-users] 2 Digit Issue
I do have it in the inside context. It is also doing the circle dance.
I just gave an example. It seems as if it is just forgetting any digits
over 2. like that is in the dialplan but it is not.
Jason
Anthony Cennami wrote:
Looking at your dialplan I don't see extension 8 anything (8XXX) --
Are you sure you didn't have those extensions in another context that
you forgot to include?
According to the dialplan it is catching the invalid extension and
should be passing it to the i (invalid) handler to loop back into your
attendant.
On 8/1/07, *Jason Walker* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2
digits into the dialplan.
error
-- Invalid extension '81' in context 'impact' on
SIP/207.174.111.34-b77167f8
I pressed 8107
and ideas
my dial plan is (part of it)
[impact]
exten=s,1,Answer()
exten=s,n,Set(CALLERID(name)=Impact - ${CALLERID(number)})
exten=s,n,Background(IMPACT)
exten=s,n,WaitExten(10)
exten=s,n,Background(OP)
exten=s,n,Dial(Sip/office2,20)
exten=s,n,Voicemail([EMAIL PROTECTED],u)
exten=9,1,Background(IPM_MENU)
exten=9,n,WaitExten(10)
exten=9,n,Goto(0,1)
exten=0,1,Goto(inside,115,1)
exten=i,1,Goto(s,3)
exten=a,1,Goto(s,3)
include=inside
[inside]
exten=102,1,Macro(users,SIP,Comp4,[EMAIL PROTECTED])
exten=111,1,Wait(1)
exten=111,2,Playback(Randy)
exten=111,3,Dial(Sip/Randy,20)
exten=111,4,Goto(111-${DIALSTATUS},1)
exten=111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
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Re: [asterisk-users] 2 Digit Issue
Thanks for the help it was a provider issue
Jason
Anthony Cennami wrote:
Looking at your dialplan I don't see extension 8 anything (8XXX) --
Are you sure you didn't have those extensions in another context that
you forgot to include?
According to the dialplan it is catching the invalid extension and
should be passing it to the i (invalid) handler to loop back into your
attendant.
On 8/1/07, *Jason Walker* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2
digits into the dialplan.
error
-- Invalid extension '81' in context 'impact' on
SIP/207.174.111.34-b77167f8
I pressed 8107
and ideas
my dial plan is (part of it)
[impact]
exten=s,1,Answer()
exten=s,n,Set(CALLERID(name)=Impact - ${CALLERID(number)})
exten=s,n,Background(IMPACT)
exten=s,n,WaitExten(10)
exten=s,n,Background(OP)
exten=s,n,Dial(Sip/office2,20)
exten=s,n,Voicemail([EMAIL PROTECTED],u)
exten=9,1,Background(IPM_MENU)
exten=9,n,WaitExten(10)
exten=9,n,Goto(0,1)
exten=0,1,Goto(inside,115,1)
exten=i,1,Goto(s,3)
exten=a,1,Goto(s,3)
include=inside
[inside]
exten=102,1,Macro(users,SIP,Comp4,[EMAIL PROTECTED])
exten=111,1,Wait(1)
exten=111,2,Playback(Randy)
exten=111,3,Dial(Sip/Randy,20)
exten=111,4,Goto(111-${DIALSTATUS},1)
exten=111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
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[asterisk-users] Web User control
I am looking to allow some users to login to a website and change where their ext is forwarded to. any ideas? It can be very simple or I can install a full package and then allow certain users certain access. Thanks in advance Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys not Ringing
I do not have any answer int he dialplan. what I mean is that when I call any other SIP phone is does the answer in the CLI. Even if I put and answer() in the dialplan still no ringing Jason Luki wrote: shouldn't there be an answer in there somewhere?... like... No... you can (and probably should) Dial() an extension before answering the incoming call. Do a sip debug and see if the Sipura is getting the INVITE message (and responding with an ACK), and if it sends back a RINGING message. Something strange is going here, and my bet is on some kind of NAT screw-up. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys not Ringing
I have 2 linksys SIP phones SPA-942 I have a dialplan of exten = 144,1,Wait(1) exten = 144,2,Dial(Sip/phil,20) exten = 144,3,Voicemail([EMAIL PROTECTED],u) The CLI looks like this when I dial 144 -- Executing Wait(IAX2/JASONSERVER-9, 1) in new stack -- Executing Dial(IAX2/JASONSERVER-9, Sip/phil|20) in new stack -- Called phil -- Nobody picked up in 2 ms -- Executing VoiceMail(IAX2/JASONSERVER-9, [EMAIL PROTECTED]|u) in new stack -- Playing 'vm-theperson' (language 'en') It is registered and will make calls but I never get the -- SIP/phil is ringing This happening on my 2 linksys phones only Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Polycom reject button
Good Idea, but when the user has to do nothing is better for my users!
Thanks
JAson
Mojo with Horan Company, LLC wrote:
Another option is to have the user hit the forward button on their
phone and manually type in their cellphone number when they're going
to be out of the office.
Jason Walker wrote:
exten = 111,1,Wait(1)
exten = 111,2,Playback(Randy)
exten = 111,3,Dial(Sip/Randy,20)
exten = 111,4,Goto(111-${DIALSTATUS},1)
exten = 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten = 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)
works GREAT
Thanks a lot
Jason
Doug Lytle wrote:
Mike wrote:
Jason,
If you do test if JR's tip works, please share your finding with
us. I am
interested in this as well.
It'll work fine, the Polycom responds with BUSY when the DND button
is pressed. Using DIALSTATUS, it'll drop to voicemail and play the
busy message if recorded if that's what you have it programmed to do.
Doug
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Re: [asterisk-users] RE: Polycom reject button
exten = 111,1,Wait(1)
exten = 111,2,Playback(Randy)
exten = 111,3,Dial(Sip/Randy,20)
exten = 111,4,Goto(111-${DIALSTATUS},1)
exten = 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten = 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)
works GREAT
Thanks a lot
Jason
Doug Lytle wrote:
Mike wrote:
Jason,
If you do test if JR's tip works, please share your finding with us.
I am
interested in this as well.
It'll work fine, the Polycom responds with BUSY when the DND button is
pressed. Using DIALSTATUS, it'll drop to voicemail and play the busy
message if recorded if that's what you have it programmed to do.
Doug
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[asterisk-users] Polycom reject button
I have users in my dialplan that go from SIP to Cell When they are at their desk and they hit reject call, it goes to the next thing in the dialplan, thus transferring to their cell. Not what they want. Is it possible to change the reject button to make it go to voice mail or a new ext? Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 Call locations
I have a SIP user and a remote IAX device I want both to ring 3 times then if neiter pick up it to go to the next thing in the dialplan. Can you do this from the dialplan or do I need to set a hunt group up? Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Provistioning Issue
From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x1 error Any Ideas? 1005195711|so |4|00|-- Initial log entry -- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry 1005195711|cfg |4|00|Initial log entry 1005195711|copy |3|00|Initial log entry 1005195711|cdp |4|00|Initial log entry 1005195711|cdp |5|00|CDP is DISABLED. 1005195711|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1005195711|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1005195711|so |3|00|Platform: Board=2345-11500-040 A 1005195711|so |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, Subnet Mask=255.255.255.0 1005195711|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1005195711|so |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 1005195711|so |3|00|Application, main: P/N=3150-11069-322 1005195711|app1 |4|00|Initial log entry. 1005195711|app1 |3|00|DNS resolver server is '192.168.15.10' 1005195711|app1 |3|00|DNS resolver search domain is '' 1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= tn=CircaIP 1005195712|so |3|00|Link status is Net up Speed 100 full Duplex, PC down. 1005195722|cfg |3|00|Beginning to provision phone 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from '192.168.15.52' 1005195722|cfg |3|00|Image bootrom.ld has not changed 1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1005195722|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg' from '192.168.15.52' 1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on attempt 1 (addr 1 of 1) 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from '192.168.15.52' 1005195724|cfg |3|00|Image sip.ld has not changed 1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1005195724|cfg |3|00|Downloaded application image is identical to current version 1005195724|cfg |3|00|Phone successfully provisioned 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). 1005195755|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Provistioning Issue
Fixed that issue but it does not change the error 0126204105|cfg |3|00|Image sip.ld has not changed 0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 0126204105|cfg |3|00|Downloaded application image is identical to current version 0126204105|cfg |3|00|Phone successfully provisioned 0126204136|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0126204136|app1 |6|00|Uploading boot log, time is FRI JAN 26 20:41:36 2007 William M. Conlon wrote: Looks like the network time server isn't provisioned. -- Bill 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Provistioning Issue
?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=jason.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY= OVERRIDES_DIRECTORY= CONTACTS_DIRECTORY=/ is my mac IP Darryl Dunkin wrote: This is typically an error in one of your config files, either 0004f2023ecc.cfg or sip.cfg. What does your 0004f2023ecc.cfg file look like? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, January 26, 2007 12:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Provistioning Issue From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x1 error Any Ideas? 1005195711|so |4|00|-- Initial log entry -- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry 1005195711|cfg |4|00|Initial log entry 1005195711|copy |3|00|Initial log entry 1005195711|cdp |4|00|Initial log entry 1005195711|cdp |5|00|CDP is DISABLED. 1005195711|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1005195711|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1005195711|so |3|00|Platform: Board=2345-11500-040 A 1005195711|so |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, Subnet Mask=255.255.255.0 1005195711|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1005195711|so |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 1005195711|so |3|00|Application, main: P/N=3150-11069-322 1005195711|app1 |4|00|Initial log entry. 1005195711|app1 |3|00|DNS resolver server is '192.168.15.10' 1005195711|app1 |3|00|DNS resolver search domain is '' 1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= tn=CircaIP 1005195712|so |3|00|Link status is Net up Speed 100 full Duplex, PC down. 1005195722|cfg |3|00|Beginning to provision phone 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from '192.168.15.52' 1005195722|cfg |3|00|Image bootrom.ld has not changed 1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1005195722|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg' from '192.168.15.52' 1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on attempt 1 (addr 1 of 1) 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from '192.168.15.52' 1005195724|cfg |3|00|Image sip.ld has not changed 1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1005195724|cfg |3|00|Downloaded application image is identical to current version 1005195724|cfg |3|00|Phone successfully provisioned 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). 1005195755|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tone Issues
I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound Operator then go to a SIP phone. I would like it to write Caller ID Time to a file I can read and find out exactly how many people are getting to that point. #3. If it is Voicepulses fault. Who else might you suggest for my numbers to be ported to and handle my phone calls Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail issues
I put my voicemail groups into different contexts so that I can use Dial
by name and escape.
I had set ext 500 as
exten = 500,1,VoiceMailMain(${CALLERID(number)[EMAIL PROTECTED]|s)
but now that the contexts are different. this does not work
#1 how do I have everyone use an ext to get the voicemail regardless of
context.
#2 can I get the mail buttons to work on my polycom 501s and swissphones
#3 where do I put the i ext to allow the caller to go from the
voicemail back to a ext in the dialplan
Thanks
Jason
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[asterisk-users] DTMF over IAX
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the next dialplan fall through. It happens 60-70% of the time.
extentions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
;OEM
exten = _12125551212,1,Goto(OEM,s,1)
[OEM]
exten = s,1,Answer()
exten = s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)})
exten = s,n,Background(Outsource)
exten = s,n,WaitExten(10)
exten = s,n,Goto(inside,133,1)
exten = 9,1,Background(OEM_Menu)
exten = 9,n,WaitExten(10)
exten = 9,n,Goto(0,1)
exten = 0,1,Goto(inside,133,1)
IAX.conf
[general]
jitterbuffer=yes
forcejitterbuffer=no
maxjitterbuffer=500
autokill=yes
; -
; IAX INCOMING USER
;
; This is the user for incoming calls from:
; connect02.voicepulse.com
; -
[voicepulse] ; -- Name must be [voicepulse]
context=voicepulse-in ; -- Should match the context you
; are using in extensions.conf
; to handle incoming calls
type=user
host=connect02.voicepulse.com
qualify=yes
notransfer=yes
disallow=all
allow=g729 ; -- List supported codecs
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=g726
allow=adpcm
allow=lpc10
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Ken, Also stay away from Swissvoice phones I have found several ways to do the second thing. http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers It works great. Jason Tom Vile wrote: I tend to stay away from the Grandstream phones for business use because they simply break to easily. I would suggest using Snom phones like the Snom 300 for around $99. 2 Asterisk boxes in different locations? Sure, you can do that and its quite easily. On 11/1/06, Ken Williams [EMAIL PROTECTED] wrote: Thanks everyone for the input.After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over.I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All that being said, any comments on the Grandstorm phones?I've ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERY inexpensive for a business solution.I see it has room for 4 lines with 7 programmable buttons.I assume I can put a few more lines on the programmable buttons (we have 6 lines at our main location). One last newbie question, I assume if I have an Asterisk PBX at 2 locations in different states, I'll be able to transfer a call that comes into location1 to a user at location2. Thanks again for the quick responses help. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with "extra signaling".The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do.Always remember that it would cost the companies a ton to make every system totally closed That being said, the entry price for IP phones or ADSI phones can be much lower than you think.Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection.That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available. Thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels but I am running CentOS4. I also do not have any cards installed. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Escape from Voicemail
I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF / Silence issues
I am now running 1.4 beta3 I have an ongoing issue that it does not recognize my DTMF key press. I will call and press as many numbers and the background message still plays. I am also having an issue with transfers NOTICE[30930]: chan_sip.c:13289 handle_request_invite: Unable to create/find SIP channel for this INVITE happens everytime Any ideas. I tried to go back to 1.2 and the modules would not show up. Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 downgrade
I am having a bunch of issues with 1.4 and want to go back to 1.2 any ideas on the best way I saw someone say apt-get remove will this work for asterisk or do I need to do it for each libpri, addons, zaptel and asterisk? Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues with Asterisk 1.4 Beta
I thought I would list my issues so all of you that know more than me might be able to help. 1. I have 6 Swissphone ip10 they disconnect calls at either 70 seconds, 120 seconds or 180 seconds I have polycom Phones that go forever 2. When I try and transfer calls I have a LONG delay before the seconds line is usable. Call1 on hold then make second call and 1 minute passes before it attempts a connect 3. I have many Polycom 501s and I cannot seem to get the tick server to work. I change settings but it does nto fetch the time 4.I get-- Got SIP response 500 Internal Server Error back from 192.168.0.XXX from all my Polycom 501 phone every 2 mintues or so 5. I get [Oct 12 08:49:56] NOTICE[29165]: rtp.c:708 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.141 on my Swiss phones Any help would be great. I am a little new to asterisk and so if I posted this incorrectly please let me know Jason Walker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial String Questions
Some phones do not send DTMF automatically. What soft phone
are you using?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
LopezSent: Wednesday, January 25, 2006 9:23 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Dial String Questions
MCI should be expecting
them as DTMF on the B channel instead of the D.
You can try setting
this up in your extensions.conf:
exten =
1XX,1,Dial(Zap/g2/${EXTEN})
that will bridge the call
once you dial.
It may also be that
your SIP phones are not sending DTMF tones, What are
the DTMF settings in sip.conf and on the soft
phone.
-Original
Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Schochet, WesSent: Wednesday, January 25, 2006 5:24
PMTo: 'Asterisk
Users'Subject:
[Asterisk-Users] Dial String Questions
Hi
all-
My TDM long distance is provided by
MCI. We use account codes where MCI sends a challenge tone after receiving
1NXXNXX. Anyone have any suggestions of how to accomplish this?
I can't get the soft phones to send the DTMF (the other digits go down the
d-channel of our PRI). I also have not bee able to get the dial or the
outgoing queue command to work. Anyone run into
this?
Wesley A.
SchochetSenior Telecommunications EngineerSelect Comfort
Corporation(763-551-7757651-592-5441*[EMAIL PROTECTED]
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RE: [Asterisk-Users] CentOS vs. Vanilla Kernel
Julian - What hardware are you using? Proc, RAM, SCSI or IDE, etc. The reason I ask is that I have multiple hardware platforms, all on FC1 or FC4, and none of them hit 100% for each IRQ. I am usually in the high 98% with the occasional 100% on P3 servers (give or take 1 Gig RAM, 1 Gig CPU). Two servers are dual p3 1.2 with 2 Gigs Ram. Since CentOS is brought up, maybe my OS is the culprit...far fetched? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, November 10, 2005 12:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CentOS vs. Vanilla Kernel Not a problem that I've had :) Linux foxtrot.tessera.co.uk 2.6.9-22.0.1.EL #1 Thu Oct 27 12:26:11 CDT 2005 i686 i686 i386 GNU/Linux Opened pseudo zap interface, measuring accuracy... 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% --- Results after 24 passes --- Best: 100.00 -- Worst: 100.00 -- Average: 100.00 [EMAIL PROTECTED] zaptel]# Julian. [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote on 11/07/2005 01:17:31 PM: HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Will you be using Zaptel hardware? The only way I can get zttest results of 100% is with a CentOS 2.4 kernel. Any CentOS 2.6 kernel I've tried (Uni, SMP, with IOAPIC enabled or disabled) gave me 99.99% at best... Tim Massey -- -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can't create iax channel
The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. Based on your post, seems that you have an issue with codecs more than creating an IAX trunk. What version of Asterisk are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Gemmell Sent: Thursday, November 10, 2005 12:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can't create iax channel Hi all Could somebody please give me an idea as to whats wrong here. I'm trying to connect 2 servers using IAX, I'm not trunking them because I read that you need zaptel hardware installed at both sides to do the trunking. Theregistration seems to have worked as the output of iax show peers on the side I'm working from is as follows Name/UsernameHost Mask Port Status wayne165.165.164.87 (D) 255.255.255.255 4569 Unmonitored and on the other side iax2 show users shows Username SecretAuthen Def.Context A/C Codec Pref waynepassword 001 default No Host When trying to call from this side to that side I get the following -- Executing Dial(SIP/301-2d50, IAX2/wayne:[EMAIL PROTECTED]/204) in new stack Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: chan_iax2.c:7745 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/wayne-5 -- Hungup 'IAX2/wayne-5' Nov 10 08:37:21 NOTICE[30785]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(SIP/301-2d50, ) in new stack == Spawn extension (from-internal, 204, 2) exited non-zero on 'SIP/301-2d50' Any ideas? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HDLC errors on PRI
I have looked through other postings to the user group for HDLC errors, went through what worked for other people, and still can not seem to get past this issue. For 3 days, I have been getting HDLC abort(6) errors in *. Prior to Tuesday, the circuits were clean...I had maybe 10 HDLC abort messages since August 10th. Here are my specs: 1 Gig IBM x300 w/ 1 Gig Ram 1 Quad TE405P card No errors on IRQs IRQs are separated with NO sharing hdparm for irq and dma are set to 'on' Software - FC1 with -1 updates to kernel, etc. Asterisk v 1.0.9, libpri 1.0.9, zaptel 1.0.9.2 1 T1 is a tieline to our Nortel Meridian 3 T1s are a PRI trunk group with D chans on 24 and 48. The third T1 only has b channels. No alarms from zttool. Calls go through, inbound and outbound. About every 5 seconds, I get the following on the console: Nov 4 21:10:37 NOTICE[9693]: PRI got event: Alarm (4) on Secondary D-channel of span 1Nov 4 21:10:37 NOTICE[9693]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 The errors seem to increase as calls come in and out. There is also a noticable "popping" when the error happens. Any suggestions are welcome. thank you Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I give up - Help with TE410P
My 2 cents: If you are running kudzu on RH or FC, new and remove hardware should be detected...in most cases. I assume other distros have something similar...? If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue. Can you swap cables from a bad circuit to a good circuit? Are all of the circuits the same configuration from the carrier? As far as support, Digium's email support has ALWAYS been helpful to me - from basic questions to systematic issues. They have always been helpful and responsive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, October 29, 2005 4:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. Actually my support from them has been great... Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. allowed linux to remove the missing cards ?? what distro are you using? Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Again, what distro, what version of asterisk and whatnot? Is this [EMAIL PROTECTED] Since it's working, I'm done - but only go to show you these cards are flaky. It sounds like your system is what's flaky here... Linux doesn't need to remove the cards... Definitely something nonstandard from my point of view. I am glad it's working for you though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I give up - Help with TE410P
I understand the loopback scenario. Have you swapped the loops between circuits? Are circuits on some of your T1s but loops on others? Can you swap them to see if the green leds follow the cabling? I have kudzu enabled and do not have any issues...although I do not put more than one card in a server. When you say some of the ports are working again, can you expand on that? How about an IRQ issue? Too many for your server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bart Fisher Sent: Saturday, October 29, 2005 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] I give up - Help with TE410P Yep, it CentOS 4.0 (RH) - Kudzu - also seems to be the root of my problem. I later rebooted and now back to some ports working again. I'm using a Loop-Back plug to test with - no real T1 attached until I can fix this. Swapping card does not seem to follow issues. Maybe I'll give support another :) Bart - Original Message - From: Jason Walker [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 5:09 PM Subject: RE: [Asterisk-Users] I give up - Help with TE410P My 2 cents: If you are running kudzu on RH or FC, new and remove hardware should be detected...in most cases. I assume other distros have something similar...? If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue. Can you swap cables from a bad circuit to a good circuit? Are all of the circuits the same configuration from the carrier? As far as support, Digium's email support has ALWAYS been helpful to me - from basic questions to systematic issues. They have always been helpful and responsive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, October 29, 2005 4:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. Actually my support from them has been great... Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. allowed linux to remove the missing cards ?? what distro are you using? Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Again, what distro, what version of asterisk and whatnot? Is this [EMAIL PROTECTED] Since it's working, I'm done - but only go to show you these cards are flaky. It sounds like your system is what's flaky here... Linux doesn't need to remove the cards... Definitely something nonstandard from my point of view. I am glad it's working for you though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Red Alarms, No D-Channels, and Crazy People
One thing to consider is if there were alarms on the T1 to SBC, they may have something in place to take the circuit down. Even if you get your configs right, the T1 just might not come up clean. MCI does this to us sometimes. Please post your /etc/zaptel.conf and your /etc/asterisk/zapata.conf Thanks Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Tillman Sent: Monday, October 24, 2005 4:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Red Alarms, No D-Channels, and Crazy People I am still getting up to speed on the Asterisk system in place at my new employer. Today we are getting a lot of this: Oct 24 17:21:33 WARNING[2828]: Detected alarm on channel 1: Red Alarm Oct 24 17:21:33 WARNING[2828]: Unable to disable echo cancellation on channel 1 [snip] Oct 24 17:21:33 WARNING[2828]: Detected alarm on channel 23: Red Alarm Oct 24 17:21:33 WARNING[2828]: Unable to disable echo cancellation on channel 23 Oct 24 17:21:33 NOTICE[2828]: PRI got event: Alarm (4) on Primary D-channel of span 1 Oct 24 17:21:33 WARNING[2828]: No D-channels available! Using Primary on channel anyway 24! Oct 24 17:21:41 WARNING[2828]: No D-channels available! Using Primary on channel anyway 24! Oct 24 17:21:55 NOTICE[2828]: PRI got event: No more alarm (5) on Primary D-channel of span 1 Oct 24 17:21:55 WARNING[2828]: No D-channels available! Using Primary on channe l anyway 24! Oct 24 17:21:55 NOTICE[2828]: Alarm cleared on channel 1 [snip] Oct 24 17:21:55 NOTICE[2828]: Alarm cleared on channel 23 Should I be looking priamrily at the telco as the cause of this? People here are prepping an old fashioned tar and feathering for me. Thanks, -dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Red Alarms, No D-Channels, and Crazy People
I have not read through the rest of your posts, but try some of the other variations of switchtype: ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; switchtype=national For example, we had a PRI with Global Comm (through a reseller) and they swore up and down that a 5ess was what was needed to for signalling - basically, we could not get the circuit up and clean. When I switched to national, my issues went away. Funny thing is that through our Dialogic (on the same circuit) we had to use Dialogic's 4ess drivers/config for that circuit to work on another telecom device. Have you tried that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Tillman Sent: Monday, October 24, 2005 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Red Alarms, No D-Channels, and Crazy People On 10/24/05, Gary Reuter [EMAIL PROTECTED] wrote: Why not attach /etc/zaptel.conf and /etc/asterisk/zapata.conf? Tacked on below. First thing I'd actually check is the wiring: if you jiggle the cable and the led changes, you've got a serious problem, but since you've only been there 3 weeks, you can blame on the previous guy! ;-) I don't think it is the cabling, but you should see some of the other stuff the guy did... -dave # # zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 fxoks=25-28 loadzone = us defaultzone=us # zapata.conf [channels] txgain=-5.5 language=en context=default usecallerid=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes signalling=pri_cpe switchtype=dms100 group=0 context=from-pstn faxdetect=incoming echocancel=yes echotraining=800 channel = 1-23 signalling=fxo_ks context=from-internal callerid=Internal Fax 1 (xxx) 555-5551 channel = 25 callerid=Internal Fax 2 (xxx) 555-5552 channel = 26 callerid=Internal Fax 3 (xxx) 555-5553 channel = 27 callerid=Internal Fax 4 (xxx) 555-5554 channel = 28 signalling=fxs_ks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to negotiate codec???
What codec are you using on the client and the server? From my understanding, you have to have a license for both ends of the G.729 call. Are you passing this through one server to another and the call is being rejected at the server level? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clint_in_sydneySent: Saturday, October 22, 2005 5:44 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Unable to negotiate codec??? Hi All,I get the following when trying to dial in to my asterisk box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, requested/capability 0x200/0xfe00 incompatible with our capability 0xf900.and I get the following when I try to dial out.Oct 22 14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate codecI'm using a brand new g729 codec from Digium.Any ideas on what my problem might be?Cheers,Clint ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to negotiate codec???
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clint_in_sydneySent: Saturday, October 22, 2005 7:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Unable to negotiate codec??? I use IAX and have a license for G729 at my end and OZTell, my provider, use G729 as their main codec. My box rejects connections from my provider due to incompatible codecs and vice versa. I'm waiting for them to get back to me on this. Clint. - Original Message - From: Jason Walker To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Sunday, October 23, 2005 10:52 AM Subject: [other] RE: [Asterisk-Users] Unable to negotiate codec??? What codec are you using on the client and the server? From my understanding, you have to have a license for both ends of the G.729 call. Are you passing this through one server to another and the call is being rejected at the server level? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clint_in_sydneySent: Saturday, October 22, 2005 5:44 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Unable to negotiate codec??? Hi All,I get the following when trying to dial in to my asterisk box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, requested/capability 0x200/0xfe00 incompatible with our capability 0xf900.and I get the following when I try to dial out.Oct 22 14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate codecI'm using a brand new g729 codec from Digium.Any ideas on what my problem might be?Cheers,Clint ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to negotiate codec???
Sorry for the blank response - before... From your output below, what looks weird are the hex values for the codecs: [snip] requested/capability 0x200/0xfe00 incompatible with our capability 0xf900. From one of my servers, when I do a 'show codecs' on the console, I get sfsip01*CLI show codecsDisclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audio gsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726 (G.726) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8) (0x100) audio g729 (G.729A) 512 (1 9) (0x200) audio speex (SpeeX) 1024 (1 10) (0x400) audio ilbc (iLBC) 65536 (1 16) (0x1) image jpeg (JPEG image) 131072 (1 17) (0x2) image png (PNG image) 262144 (1 18) (0x4) video h261 (H.261 Video) 524288 (1 19) (0x8) video h263 (H.263 Video) 0x200 would be speex. G.729 - in hex, from this display - would be 0x100. >From your output, I don't see 0x100 at all. Am I confused? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clint_in_sydneySent: Saturday, October 22, 2005 7:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Unable to negotiate codec??? I use IAX and have a license for G729 at my end and OZTell, my provider, use G729 as their main codec. My box rejects connections from my provider due to incompatible codecs and vice versa. I'm waiting for them to get back to me on this. Clint. - Original Message - From: Jason Walker To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Sunday, October 23, 2005 10:52 AM Subject: [other] RE: [Asterisk-Users] Unable to negotiate codec??? What codec are you using on the client and the server? From my understanding, you have to have a license for both ends of the G.729 call. Are you passing this through one server to another and the call is being rejected at the server level? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clint_in_sydneySent: Saturday, October 22, 2005 5:44 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Unable to negotiate codec??? Hi All,I get the following when trying to dial in to my asterisk box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800 formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22 13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19, requested/capability 0x200/0xfe00 incompatible with our capability 0xf900.and I get the following when I try to dial out.Oct 22 14:06:31 WARNING[3599]: Call rejected by 203.98.83.19: Unable to negotiate codecI'm using a brand new g729 codec from Digium.Any ideas on what my problem might be?Cheers,Clint ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax softphone
Tom - do you end up with that phone shutting down with an error on Windows XP? I downloaded the latest. After about 3 minutes on a call, the other end can no longer hear me and then the phone just dies. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom VileSent: Saturday, October 22, 2005 8:21 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] iax softphone Idefisk for me. I love how it does not clutter the screen and it works. On 10/22/05, Matt Florell [EMAIL PROTECTED] wrote: We use the Firefly ThirdParty softphone on our windows laptops. It'sfree, easy to configure and will do IAX2 and SIP:http://www.virbiage.com/firefly/download/firefly-thirdparty.exeMATT---On 10/22/05, Zoa [EMAIL PROTECTED] wrote: Idefisk is currently only for windows, but a native linux version is nearly ready and will be released soon, others i can also recommend : - iaxphone by ipsando - firefly by virbiage. Time Bandit wrote: can someone tell me about a good iax softphone ?? Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php works only on windows for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html there is also DIAX : http://www.laser.com/dante/diax/diax.html and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php hth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax softphone
Done - Joachim, I cc'd you on the email so you could see what I sent. Let me know if more info is needed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoa Sent: Saturday, October 22, 2005 10:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iax softphone Jason, i didn't hear about that problem before (several thousand people are using that version), could you please send a copy of your config files + the exact version and language localisation of windows xp to [EMAIL PROTECTED] Does it happen with one specific version of asterisk ? Whatever the problem is, it should not be there. Please help us find the bug. Joachim. Jason Walker wrote: Tom - do you end up with that phone shutting down with an error on Windows XP? I downloaded the latest. After about 3 minutes on a call, the other end can no longer hear me and then the phone just dies. -- -- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Tom Vile *Sent:* Saturday, October 22, 2005 8:21 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] iax softphone Idefisk for me. I love how it does not clutter the screen and it works. On 10/22/05, *Matt Florell* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We use the Firefly ThirdParty softphone on our windows laptops. It's free, easy to configure and will do IAX2 and SIP: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe MATT--- On 10/22/05, Zoa [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Idefisk is currently only for windows, but a native linux version is nearly ready and will be released soon, others i can also recommend : - iaxphone by ipsando - firefly by virbiage. Time Bandit wrote: can someone tell me about a good iax softphone ?? Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php works only on windows for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html there is also DIAX : http://www.laser.com/dante/diax/diax.html and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php http://www.asteriskguru.com/tools/idefisk_beta.php hth ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 --- - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com
RE: [Asterisk-Users] iax softphone
Are you running on XP SP2just curious? How about the version of *? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom VileSent: Saturday, October 22, 2005 10:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] iax softphone Nope, I do not have that issue. On 10/23/05, Jason Walker [EMAIL PROTECTED] wrote: Tom - do you end up with that phone shutting down with an error on Windows XP? I downloaded the latest. After about 3 minutes on a call, the other end can no longer hear me and then the phone just dies. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom VileSent: Saturday, October 22, 2005 8:21 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] iax softphone Idefisk for me. I love how it does not clutter the screen and it works. On 10/22/05, Matt Florell [EMAIL PROTECTED] wrote: We use the Firefly ThirdParty softphone on our windows laptops. It'sfree, easy to configure and will do IAX2 and SIP:http://www.virbiage.com/firefly/download/firefly-thirdparty.exe MATT---On 10/22/05, Zoa [EMAIL PROTECTED] wrote: Idefisk is currently only for windows, but a native linux version is nearly ready and will be released soon, others i can also recommend : - iaxphone by ipsando - firefly by virbiage. Time Bandit wrote: can someone tell me about a good iax softphone ?? Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php works only on windows for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html there is also DIAX : http://www.laser.com/dante/diax/diax.html and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php hth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax softphone
Do you have any issues with not being able to hear the called party after +3 minutes? That is pretty consistent thus far. Don't get me wrong, I am liking the phone so far. Small interface, easy to configure. Uses an XML derived config file - nice for deployment to multiple computers. And the portion of the calls I can hear sound very nice. I just lose the call and the phone bombs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoa Sent: Saturday, October 22, 2005 10:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iax softphone I'm running it on sp2 myself, never had a crash with it so far. Jason Walker wrote: Are you running on XP SP2just curious? How about the version of *? -- -- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Tom Vile *Sent:* Saturday, October 22, 2005 10:03 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] iax softphone Nope, I do not have that issue. On 10/23/05, *Jason Walker* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Tom - do you end up with that phone shutting down with an error on Windows XP? I downloaded the latest. After about 3 minutes on a call, the other end can no longer hear me and then the phone just dies. *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Tom Vile *Sent:* Saturday, October 22, 2005 8:21 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] iax softphone Idefisk for me. I love how it does not clutter the screen and it works. On 10/22/05, *Matt Florell* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We use the Firefly ThirdParty softphone on our windows laptops. It's free, easy to configure and will do IAX2 and SIP: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe http://www.virbiage.com/firefly/download/firefly-thirdparty.exe MATT--- On 10/22/05, Zoa [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Idefisk is currently only for windows, but a native linux version is nearly ready and will be released soon, others i can also recommend : - iaxphone by ipsando - firefly by virbiage. Time Bandit wrote: can someone tell me about a good iax softphone ?? Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php http://www.marccharbonneau.com/asterisk/mediaxphone.php works only on windows for one that works on Windows and Linux : http://iaxclient.sourceforge.net/iaxcomm/index.html http://iaxclient.sourceforge.net/iaxcomm/index.html there is also DIAX : http://www.laser.com/dante/diax/diax.html and idefisk : http://www.asteriskguru.com/tools/idefisk_beta.php http://www.asteriskguru.com/tools/idefisk_beta.php hth ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 845-652-4578 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856
[Asterisk-Users] Just a test...
I have not seen any posts for awhile. Just testing. thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple instances of asterisk showing from 'ps aux'
When I run 'ps aux' I get this: root 964 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 965 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 967 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 975 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 982 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 984 0.0 0.4 47836 8280 ? S 00:02 0:12 asterisk -vvvg -croot 986 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 987 0.1 0.4 47836 8280 ? S 00:02 1:10 asterisk -vvvg -croot 988 0.1 0.4 47836 8280 ? S 00:02 1:24 asterisk -vvvg -croot 989 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 993 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 994 0.0 0.4 47836 8280 ? S 00:02 0:17 asterisk -vvvg -croot 996 0.0 0.4 47836 8280 ? S 00:02 0:00 asterisk -vvvg -croot 997 0.0 0.4 47836 8280 ? S 00:02 0:02 asterisk -vvvg -croot 24202 1.2 0.4 47836 8280 ? S 09:04 6:52 asterisk -vvvg -croot 29417 1.6 0.4 47836 8280 ? S 11:07 6:54 asterisk -vvvg -croot 6555 1.0 0.4 47836 8280 ? S 14:44 2:04 asterisk -vvvg -croot 8463 1.1 0.4 47836 8280 ? S 15:29 1:53 asterisk -vvvg -croot 14405 1.0 0.4 47836 8280 ? S 17:47 0:15 asterisk -vvvg -c My question is, why are there 21 instances of asterisk running? I understand the concept of a multi-threaded app in Linux (such as httpd). I am just looking for possible avenues and explanations of where I could look to figure out what each instance (or some of the instances) are actually doing. * 1.0.9; FC1 Thanks in advance Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial 2 channels at onece: Not working anymore atCVS?
What if you force a hangup between the two steps? I have multiple destinations specified when my internal number is called at work using similar syntax. All of the SIP and SCCP extensions dial based on my setup - which again, is very similar to yours. I do not use CVS HEAD on the production boxes...I am somewhat stuck with 1.0.9 at work. I don't know if that is the difference or not. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Tello Abrego Sent: Wednesday, October 19, 2005 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dial 2 channels at onece: Not working anymore atCVS? Version smbserver*CLI show version Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-10-20 01:13:39 UTC exten = 1100,1,Dial(Zap/5,30,Ttr) exten = 1100,n,Dial(Zap/8Zap/5,30,Ttr) Doesn´t dial zap/8 zap/5... * output: -- Nobody picked up in 3 ms -- Hungup 'Zap/5-1' -- Executing Dial(Zap/7-1, Zap/8Zap/5|30|Ttr) in new stack -- Called 8 Oct 19 20:33:55 WARNING[547]: chan_zap.c:1819 zt_call: Unable to ring phone: Device or resource busy -- Couldn't call 5 This USED to work, not so long ago... Why is this not working anymore? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium TDM400P (11B) problems
As an FYI - here is the output of my TDM400P: Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) I do not have newt installed on this machine, so zttool bombs. Just sending this out as an example. Here are my zap[tel|ata] conf files: Zaptel.conf: fxols=1 fxsls=3-4 [channels] context=fxo1 signalling=fxs_ls ... channel = 3 context=fxo1 signalling=fxs_ls ... channel = 4 context=fxs1 signalling=fxo_ls ... channel = 1 The ... Represents more information that may or may not be useful...I didn't think it was necessary for this thread. And frankly, I have rambled on enough ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Murray Sent: Wednesday, October 19, 2005 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium TDM400P (11B) problems Hi Rich, On 20/10/2005, at 2:42 PM, Rich Adamson wrote: snip dmesg: PCI: Found IRQ 12 for device :00:0a.0 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Adjusting gain Module 3: Installed -- AUTO FXO (NEWZEALAND mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules) Assuming you copy/pasted the above accurately, the output from dmesg is missing the stuff related to Module 4. You might also try zttool to see what it thinks is going on. That is all the output in dmesg, is there any other output in particular you were expecting? It's zero-based so Module 3 is the 4th module on the card. zttool doesn't report any alarms or anything of note. What does the loop button do? Cheers Phil Murray ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail 2
Correct - but is the context defined in voicemail.conf? As mickeymouse? Or whatever...? ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Saturday, October 15, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail 2 well, is just the context. You could call it as you prefer, mickeymouse??? ;o) Bye 2005/10/16, Linc Fessenden [EMAIL PROTECTED]: FaberK wrote: [EMAIL PROTECTED] --- Some ideas? Only thing I have that even looks different is [EMAIL PROTECTED] -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD calls to busy agents
Have you tried the incominglimit parameter (or did she)? I have found this to work pretty well when limiting the number of calls. After monitoring the full log, I saw that incoming calls where incrementing or decrementing the active call parameter for SIP agents. By limiting the number of calls that the phone extension/user can accept at one time limited the calls going to an agent. I am still trying to figure out how to jump out of the dialplan when a call comes into queue -- if anyone has any suggestions for that, it would be greatly appreciated. But in any event, for similar situations, limiting the number of calls for a SIP agent seems to help in the calls coming in on top of another. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J Thomas Sent: Saturday, October 15, 2005 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ACD calls to busy agents One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another ACD call when his turn comes. This results in unnecessary delay in answering that call. Taking out call waiting is not an option, as an agent can also get a direct dialed call, and he should be able to pick up that call even when he is on another call. Is there a way so that a busy agent (whether busy because of an incoming call, or outgoing call) is not presented another ACD call? Thanks, -- jt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Perplexed - IAX trunk == jitterbuffer or no jitterbuffer
I have 4 * servers interconnected with IAX trunks. Three are on a local LAN, one is accessible over a VPN tunnel out of the office. The IAX peer status (iax2 show peers from the CLI) will sometimes show upwards of 300ms. Considering the lag and distance, I am not entirely surprised. Anyway - my question falls towards the jitterbuffer settings in the iax.conf. Should I or should I not? I seem to come across one document that says to do it to only find another document that says this is not the best option for my particular installation. So I am now perplexed. I did updated the MAX_TIMESTAMP_SKEW value in rtp.c to an increased value (found that in one of the bug trackers) and then recompile. But the other settings, let alone to use the jitterbuffer at all, is still a quandary. These are the latest values I am using: jitterbuffer=yes dropcount=2 maxjitterbuffer=200 maxexcessbuffer=40 minexcessbuffer=5 jittershrinkrate=1 I have changed bandwidth and tos to maximize bandwidth and reliability. What I end up with are calls that sound like the far end is in a helicopter. I can only assume that the packets are ending up out of order. Or...? Any help, assistance, guidance, and past experience is GREATLY appreciated! Thanks! Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Perplexed - IAX trunk == jitterbuffer or nojitterbuffer
Thank you for the reply. All of the serves are running 1.0.9. If jitterbuffer and the like are not available, why have those options available from the non-CVS/HEAD release and in a series that does not support such features. I don't seem to recall reading that anywhere else - not an argument against your reply, just a comment of documentation lacking a key element (versions, etc.) Do the bandwidth and TOS settings work for IAX? Any recommendations for those two variables? Are they independent of each other or are they interchangeable? Again, thanks for your reply, Steve. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, October 13, 2005 5:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Perplexed - IAX trunk == jitterbuffer or nojitterbuffer On Wed, 12 Oct 2005, Jason Walker wrote: I have 4 * servers interconnected with IAX trunks. Three are on a local LAN, one is accessible over a VPN tunnel out of the office. The IAX peer status (iax2 show peers from the CLI) will sometimes show upwards of 300ms. Considering the lag and distance, I am not entirely surprised. Anyway - my question falls towards the jitterbuffer settings in the iax.conf. Should I or should I not? I seem to come across one document that says to do it to only find another document that says this is not the best option for my particular installation. So I am now perplexed. Hi Jason, You need to tell us which Asterisk version you are using. In the 1.0 series, trunking and the jitter buffer won't work together - the trunking process mangles frame timestamps in a way that the jitter buffer can't handle. In CVS-HEAD/1.2, you can optionally have trunked frames include extra timestamp info so that the jitter buffer can still work. Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Mitel SX 200 Slip and Frame Errorscausing Major Ala rms
You may have already tried this, but in the past whenever slips come into the picture on my T1s, crimping a new end for the CAT5 cable seems to help. We run T1s to a 110 block. Every once in awhile, the 110 needs to be repunched. I have found that slips can clear up when we rerun the cable...strange, but it sometimes helps. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Monday, October 10, 2005 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Mitel SX 200 Slip and Frame Errorscausing Major Ala rms Geoff Manning wrote: We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been working with a Mitel tech to get all the configurations correct and we still haven't been able to resolve the issue. We are currently connecting via crossover so we'll try a straight through just for kicks. We have a spare TE110P so we are going to try that. I just don't know enough about these errors to know what to try next. Any other thoughts? span=1,1,0,d4,ami em=1-24 Looks like you have told Asterisk to get it's timing from the Mitel. I'll bet the Mitel is trying to get it's timing from Asterisk. Try span=1,0,0,d4,ami and run ztcfg -vvv ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: :SPAM: Re: [Asterisk-Users] RE: faxing to/from asterisk - newscripts
I would appreciate seeing the scripts as well. Nice job! Desktophero at gmail.com Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Troy Swaine Sent: Friday, October 07, 2005 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: :SPAM: Re: [Asterisk-Users] RE: faxing to/from asterisk - newscripts Can you please also send to phpkidathotmail.com. Rplace the at of course. Thanks. TRoy Rajesh kumar wrote: Please send them to me at [EMAIL PROTECTED] regards, Rajesh - Original Message - From: Technical Support To: asterisk-users@lists.digium.com ; 'Roman' Sent: Friday, October 07, 2005 9:54 AM Subject: [Asterisk-Users] RE: faxing to/from asterisk - new scripts Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to [EMAIL PROTECTED] for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules / virtual user tables). They also include error logs, parameter checking, etc. Let me know if you want them Michelle Dupuis Technical Support Specialist Oxford Consulting Group Ltd. Making IT work for your business... T: (519) 672-8238 E: [EMAIL PROTECTED] W: www.ocg.ca --- --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma DS3 cards + Asterisk
Has anyone used the DS3 card from Sangoma with Asterisk? I have read many posts from users that the Sangoma cards have better echo canceling and so forth. I guess I am just wondering if there are more benefits to using this brand. I currently am responsible for multiple Asterisk servers all with Digium hardware. The echo on both TDM and some of the TE405P cards can be prevalent on some calls. There are also issues with overall volume. This, of course, is an issue with the business I work for and can cause issues across the board. Thank you in advance for any feedback and input. Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't call
It looks like your * server is not able to see the destination (presumably sip.uni.it).No route to destination -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Fabio MontemaggioreSent: Friday, September 30, 2005 2:34 AMTo: asteriskSubject: [Asterisk-Users] Don't callI receive a call, but don't call...Asterisk show this message.Are codecs the problem?Sep 30 11:25:54 WARNING[4475]: chan_sip.c:1899create_addr: No such host: sip.uni.it,rSep 30 11:25:54 NOTICE[4475]: app_dial.c:1109dial_exec_full: Unable to create channel of type 'SIP'(cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) ___Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server
One key that I have found is the more RAM the better. I am not discounting the CPU by any means and with the number of registrations you are talking about, I have not set up a system for that many concurrent users. I do have a 2x1.266 PIII w/ 2 Gigs of RAM that handles 75-85 concurrent SIP (GSM) sessions + 3 IAX2 trunks for a total of about 100 calls at the same time. (Using * 1.0.9). I also have this server set up for a file server (authentication using LDAP in a Win2K ADS domain). I also have a Celeron 1Gh w/ 2 Gigs of RAM that handles 72 DIDs, 24 connections to our traditional PBX, and is our incoming FX server. (Also running 1.0.9) Many, if not all, of our issues seem to go away when the server a) has plenty of RAM (IMHO, max the Ram!) and b) I schedule a nightly reboot. Just my 2 cents. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrien Laurent Sent: Friday, September 30, 2005 6:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server Hi everyone, I'm looking to buy a server that could handle 100 IAX users (g711)-(about 300 registrations) simultaneously. No zap channels. My budget is 1000$ us, Is a fast (3ghz) single server more reliable than a double cpu (like 1ghz) ? Will asterisk take full profit of two cpus? Isn't better to get a second cpu to handle system processes (like stat generation, backups...)? so that the remaining cpu will always be free for asterisk? I have also a 500$ deal for a 4 pentium III 500 cpu server. Thanks, -- Adrien Laurent 514-284-2020 [EMAIL PROTECTED] www.modulis.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Revieving some fax problems
I have run into a similar situation. One of our older faxes at the office seems to not work with spandsp module. The newer faxes work just fine. When I watch the logs, there appears to be communication from * requesting the fax to slow down. When the fax machine does not respond, * seems to say forget it and fail on the retrieval. I have not come up with a fix...regardless of rx/tx gains on the zaptel cards. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexandre Leclerc Sent: Friday, September 30, 2005 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Revieving some fax problems Hi, We are recieving some faxes, but I would say that about 50% of them do not work. We don't know why... is it something with the faxes speed, volume, etc? Should we use a real fax machine? Using a TDM13B with a rxgain of about 5.0... Thank you for any help. -- Alexandre Leclerc ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Who is going to AstriCon (TheAsteriskConference)?
This would be super-fantastical!!! With all of the other conferences going on, I can only get away so much. I love the idea of a webcast... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Saturday, September 17, 2005 8:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Who is going to AstriCon (TheAsteriskConference)? Sounds like a great idea to me --I know of some software which would allow audio and shared web pages and even talkback using very little resources. Even a dialup user could use it. http://www.talkingcommunities.com . on Saturday 09/17/2005 Dean Collins([EMAIL PROTECTED]) wrote Well I'm stunned no one has suggested a webcast option. I mean we aren't talking a bunch of people unable to grasp the concepts of chat/voice/vision sessions with a log in/remote display capability. If you think this is an option let me know I have someone who has some software they wouldn't mind stress testing as a trial. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of CalebSent: Saturday, 17 September 2005 8:48 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Who is going to AstriCon (TheAsteriskConference)? I'd have to second what Craig mentioned. Begin based out of Singapore we brought up a couple of points for consideration on organising an AstriconAsia in an email to Olle sometime back. SE Asia (and generally Asia) as a whole is really seeing a large increase in the number of IP Service Providers and many of them which I know are using Asterisk as well. Maybe its time to consider having an AstriconAsia 2006? If you are interested in seeing something like this materialise, drop me an email and I'll try to consolidate a list of interested participants for Digium/IPSando to consider. Regards, Caleb On 9/18/05, Craig Guy [EMAIL PROTECTED] wrote: If you're wanting some of the Asian users how about somewhere in SE Asia such as Singapore for an Astricon? Is also good for us Australians. I went to Madrid this year for Astricon but I'm not sure I'd ever be able tomake it to the US. Besides, Singapore is only 5 hours flight from Australia rather than approx 20 hours for Europe or the US. Craig - Original Message - From: Steven Sokol [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, September 17, 2005 9:32 AM Subject: Re: [Asterisk-Users] Who is going to AstriCon (The AsteriskConference)? On 9/16/05, Brian Roy [EMAIL PROTECTED] wrote: On 9/16/05, Steven Sokol [EMAIL PROTECTED] wrote: Hi, I'm taking a straw-poll to see who out there is planning on going to AstriCon. Enjoyed it last year, but putting it on the west coast seems to be pretty restrictive. I won't be making it. Atlanta was a good compromise. Maybe consider moving it to a more central location next year and I'llbe back. Well, I know it's a bit of a flight from coast-to-coast, but the Californians, Oregonians, and Washingtonians (sp?) did it last year, so we figured it would be good to give them a break. We also hoped to get some Asterisk users from Asia to make the hop. We're thinking of a central location for 2006 -- Dallas and Denver (two places that are central and fairly easy to get to by air) are currently at the top of the list. What central city sounds good to you? Thanks, Steve -Brian -- Steven Sokol CEO/Manager Sokol Associates, LLC Ask Me About AstriCon 2005! http://www.astricon.net/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
RE: [Asterisk-Users] Grandstream
That's what I have used...works until you change it. ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rene Kluwen Sent: Friday, September 16, 2005 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream admin? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joshua Abbott Sent: zaterdag 17 september 2005 1:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Grandstream Where do I find or what is the default password for a GrandStream BT 101 for the web interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is digium supporting new te405p and te406p install?
I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM server just waiting and try it every so often. When I am using a card for timing (TE405P is what we pretty much use), I feel pretty comfortable with FC1 and 1.0.9. Are you using 1.0.9? Have you tried 1.2 beta? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 7:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is digium supporting new te405p and te406p install? Hi, I tried to install these cards using FC3 and FC4 on various motherbords, but to fail. I sent email to digium several times, but no response. I think these cards are not for production use yet. Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is digium supporting new te405p and te406pinstall?
I kept running into compile errors when dealing with my Compaq (it is an older quad 700 Xeon...not sure of the model number). Once I dropped to FC1, the install of 1.0.9 compiled and install without an issue. Is there some other process/app that you are running that requires the newer kernel? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 8:54 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Is digium supporting new te405p and te406pinstall? I tried both 1.0.9 and 1.2beta. I couldn't see any interrupt from /proc/interrupt. My email server has no spam filter. --- Jason Walker [EMAIL PROTECTED] wrote: I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM server just waiting and try it every so often. When I am using a card for timing (TE405P is what we pretty much use), I feel pretty comfortable with FC1 and 1.0.9. Are you using 1.0.9? Have you tried 1.2 beta? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 7:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is digium supporting new te405p and te406p install? Hi, I tried to install these cards using FC3 and FC4 on various motherbords, but to fail. I sent email to digium several times, but no response. I think these cards are not for production use yet. Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP reinvite asterisk and NAT
I am curious...are you saying to use SIP locally and IAX from point to point (over a WAN or VPN tunnel)? With that in mind, do you think that using a lesser compressed codec over the IAX trunk would give an okay amount of bandwidth savings? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, September 15, 2005 7:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP reinvite asterisk and NAT If these phones are all to be in a single location I'd deploy a remote Asterisk box and run an IAX trunk between remote and local sites. That'll save more bandwidth than having a potential 5 individual SIP sessions running over your link. Also, with the addition of an analogue card such as the TDM400 series you'll have survivability should your link go down. If you don't add a phone line to the remote site how will they be able to call 911 etc? Mark Damon Estep wrote: I would like to setup up a remote office with a half dozen or so SIP phones connected to an asterisk server via a WAN link. To conserve bandwidth I would like the phones to be able to re-invite when they call each other. The phones will be Polycom, Cisco, or Snom. I may or may not use NAT. Seems like the NAT would really mess up re-invites, any experience with that? Assuming no NAT, what should be expected in this setup? I know the transfer option in asterisk would not work, but I do not think that is a big deal since any re-invited calls would be user to user, with little or no need to transfer. As long as the SIP termination peers I am using are set to canreinvite=no then a call between the users and a remote party would not be re-invited, since the peer terminating the call is set to no, correct? Can someone share some experiences wit this type of setup? Are there other real issues to look out for or be aware of? I am really just trying to avoid having another asterisk box in the remote site to maintain, but do not want to waste bandwidth on calls going across the office. Thanks for taking the time to share your wisdom. -- -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Connection Problems
5000-600? Do you mean 5060? That is the port for 5060. 1-2 is for RTP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. Asterisk UsersSent: Sunday, September 11, 2005 12:46 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP Connection Problems Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 1-2). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P reset
PRI channels will reset when not in use throughout the day. A reset on a channel should not happen when that channel is in use. This happens all the time on my PRI circuits (TE110P and TE410P). From what I gather, it's somewhat like a handshake for the D chan between the cpe and net sides. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Saturday, September 10, 2005 3:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE110P reset My TE110P reset some times in the day. E this cause an interruption in the service. How I decide this problem? my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for E1 defaultzone=us loadzone=us my zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel = 1-23 -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/9 successfully restarted on span 1 -- B-channel 0/10 successfully restarted on span 1 -- B-channel 0/11 successfully restarted on span 1 -- B-channel 0/12 successfully restarted on span 1 -- B-channel 0/13 successfully restarted on span 1 -- B-channel 0/14 successfully restarted on span 1 -- B-channel 0/15 successfully restarted on span 1 -- B-channel 0/16 successfully restarted on span 1 -- B-channel 0/17 successfully restarted on span 1 -- B-channel 0/18 successfully restarted on span 1 -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/20 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 Thank you João Carlos Moura ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P reset
You are correct. I did not expand completely and stand corrected. An additional note...we have some Dialogic cards (not associated with *) that do the same thing on PRI. Question - is it somewhat standard to have b chans restart on PRI circuits when not explicitly configured to NOT reset? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, September 10, 2005 5:00 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TE110P reset On Saturday 10 September 2005 19:40, Jason Walker wrote: PRI channels will reset when not in use throughout the day. A reset on a channel should not happen when that channel is in use. This happens all the time on my PRI circuits (TE110P and TE410P). From what I gather, it's somewhat like a handshake for the D chan between the cpe and net sides. Not exactly. Digium's replicating the B channel resets someone noted in a particular situation. It's not required, but it shouldn't hurt. If it's causing trouble you can turn it off with resetinterval=0 in your zapata.conf. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
I installed/ran both MozPhone and DIAX but did not see in the debug any information of the URL I sent. Perhaps the real question is: if optionalurl is used, how is the url sent to the device(s)? Has anyone applied this within a solution and is willing to share their experience? Thanks! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Denis Girard Sent: Tuesday, August 30, 2005 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd Kevin Bockman a écrit : Does anyone have details on the devices that support the optionalurl method of the Queue application? I am wondering if there is a softphone that supports this. The only thing that seems to happen is the queue_log is updated with whatever is placed in the optionalurl location of the Queue command. For Windows, I only know of DIAX. For Linux, I'm not positive but I think either Kphone or linphone does it. MozPhone at moziax.mozdev.org for Linux, Windows (and soon Mac also). Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
Now I don't feel so inadequate ;) This is exactly what I am doing. Perhaps there is more to this particular option. Here is more information - I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another one with CVS HEAD). Is 1.0.7 too old? Is this command not applicable to ver 1.0.7. For the client side, I am testing MozPhone and DIAX. MozPhone ver 0.9.2-200507111326; IAXClient: CVS-2005/07/03; Jslib: 0.1.290 DIAX is version 0.9.15a; same IAXClient as MozPhone. Am I dealing with a compatibility issue more so than anything else? Thank you for your responses. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Denis Girard Sent: Wednesday, August 31, 2005 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd Jason Walker a écrit : I installed/ran both MozPhone and DIAX but did not see in the debug any information of the URL I sent. Perhaps the real question is: if optionalurl is used, how is the url sent to the device(s)? Has anyone applied this within a solution and is willing to share their experience? Just insert something like the following in extension.conf: ;--- File d'attente exten = 180,1,Queue(file_attente,tH,http://taina.sysnux.pf/) exten = 180,2,Hangup Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
Are you using the Queue(queue-name,options,URL) syntax to send a URL to the client? Do you have to configure any options on the iaxComm side for this to work properly? Or is the URL option interpreted and executed with the default browser on the PC? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Wednesday, August 31, 2005 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd I have used iaxComm successfully (http://iaxclient.sourceforge.net/ iaxcomm/). We worked with the author, Michael Van Donselaar, to enhance some of the features of this software, particularly the handling of URLs, for a fee, with the condition that any changes we financially supported would be released to the public. The version he created for us works great and I would encourage anyone to use it. As a side note, Michael is a great guy to work with and is extremely reliable in supporting this software. Thanks, Waldo On Aug 31, 2005, at 10:47 AM, Jason Walker wrote: I installed/ran both MozPhone and DIAX but did not see in the debug any information of the URL I sent. Perhaps the real question is: if optionalurl is used, how is the url sent to the device(s)? Has anyone applied this within a solution and is willing to share their experience? Thanks! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean- Denis Girard Sent: Tuesday, August 30, 2005 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd Kevin Bockman a écrit : Does anyone have details on the devices that support the optionalurl method of the Queue application? I am wondering if there is a softphone that supports this. The only thing that seems to happen is the queue_log is updated with whatever is placed in the optionalurl location of the Queue command. For Windows, I only know of DIAX. For Linux, I'm not positive but I think either Kphone or linphone does it. MozPhone at moziax.mozdev.org for Linux, Windows (and soon Mac also). Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
Is there a specific version of DIAX that I should use? I grabbed the latest release...Looking at the DIAX site, 910g has the URL feature fixed. Is it broken again in 915a? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Bockman Sent: Wednesday, August 31, 2005 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd Jason Walker wrote: I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another one with CVS HEAD). Is 1.0.7 too old? Is this command not applicable to ver 1.0.7. That's probably your problem there. I know most newer versions of DIAX will do this. There is one of the later versions where the feature is broken. You probably need to update Asterisk. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
I copied my exact queues.conf, agents.conf and sections of the dialplan over from my 1.0.7 * server to my 1.0.9 * server and the optionalurl is working! I had to use the DIAX 910g app though (MozPhone worked without an issue on 1.0.9). The 915a would not accept the URL. Are there any (dare I say) SIP phones that have the URL capabilities? Basically, I am shooting for a screen pop of the caller ID. I have attempted to use code examples from voip-info.org that work when you KNOW the Agent or extension being called. When the agent is in a queue, this is a bit tougher. The verbose messages in my full log show the SIP agent being called, but having some portion that is executable to that is tougher. I have gone through some iterations of extensions and dialplans and get that piece to work - but not without some other feature being affected (reporting, etc.). Thanks to everyone for their replies! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Denis Girard Sent: Wednesday, August 31, 2005 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd Jason Walker a écrit : Now I don't feel so inadequate ;) This is exactly what I am doing. Perhaps there is more to this particular option. Here is more information - I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another one with CVS HEAD). Is 1.0.7 too old? Is this command not applicable to ver 1.0.7. Queue + URL and Dial + URL have been in asterisk for a long time (well before 1.0) so that is not your problem. For the client side, I am testing MozPhone and DIAX. MozPhone ver 0.9.2-200507111326; IAXClient: CVS-2005/07/03; Jslib: 0.1.290 DIAX is version 0.9.15a; same IAXClient as MozPhone. Am I dealing with a compatibility issue more so than anything else? I don't know what's the trouble; it is quite straightforward, it should just work. Are the softphone working (make receive call) apart from URL? Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
From voip-info.org: Queue(queuename|options|optionalurl|announceoverride|timeout) 'optionalurl' allows you to send a URL to devices that support it. Does anyone have details on the devices that support the optionalurl method of the Queue application? I am wondering if there is a softphone that supports this. The only thing that seems to happen is the queue_log is updated with whatever is placed in the optionalurl location of the Queue command. Thank you in advance, Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk -rx (or remote connections in general)
Try setting your logger.conf to allow full output (uncomment the full section) and see if there is something specific to the CLI crash. Be careful though and do not let the logging get out of control, especially on a big system. The file can get huge. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Monday, August 22, 2005 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk -rx (or remote connections in general) Sherwood McGowan wrote: I haven't been able to find an answerand got no response whatsoever to my previous questions concerning it. Has anyone found a fix for the remote connections to the CLI causing crashes? Also, is there a known limit? I have a huge need for using asterisk -rx in scripts, which seems is kinda why the -x option as added anyway... It's never made my Asterisk crash. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk -rx (or remote connections in general)
Do you have 5 or 6 scripts running against the interface for one instance of an outside script? Or, do you have multiple connections (outside users) attempting to run multiple instances of a script that are pulling 5-6 CLI scripts? This would exponentially increase the real number of scripts being run. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Monday, August 22, 2005 2:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] asterisk -rx (or remote connections in general) Maybe I am, I don't doubt it. But why does asterisk deadlock then when about 5 or 6 scripts hang while getting output from *? --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Eric Wieling aka ManxPower -Sent: Monday, August 22, 2005 5:08 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] asterisk -rx (or remote -connections in general) - -Sherwood McGowan wrote: - Now, there actually is actually documented problems with too many - remote connections to the manager (CLI). . . I'm asking if -someone's - figured out how to fix that. - - -http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+experien - ce - - Since my scripts apparently get some funky data from the -manager, they - freeze, build up, then deadlock the asterisk server... - - No experiences with fixing this? Anyone heard something I -haven't on - when they plan to improve remote connections (or even just multiple - connections in general?) to asterisk? - - - - --Original Message- - -From: [EMAIL PROTECTED] - -[mailto:[EMAIL PROTECTED] On -Behalf Of Eric - -Wieling aka ManxPower - -Sent: Monday, August 22, 2005 3:15 PM - -To: Asterisk Users Mailing List - Non-Commercial Discussion - -Subject: Re: [Asterisk-Users] asterisk -rx (or remote -connections in - -general) - - - -Sherwood McGowan wrote: - - I haven't been able to find an answerand got no response - - whatsoever to my previous questions concerning it. - - - - Has anyone found a fix for the remote connections to the - -CLI causing - - crashes? Also, is there a known limit? - - - - I have a huge need for using asterisk -rx in scripts, which - -seems is - - kinda why the -x option as added anyway... - - - -It's never made my Asterisk crash. - -You are confused. asterisk -rx command does not use the -Manager Interface. -___ -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P dial out problem
Shot in the dark
Do you have to dial '9' on your outside line?
Perhaps if you changed your Dial command to this:
[outgoing]
exten = _9X.,1,NoOp(Call for ${EXTEN})
exten = _9X.,2,Dial(Zap/1/${EXTEN:1})
The :1 will drop the leading '9' when it hits the outside. If this is a
regular line, there should be no need for the '9'.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Piero Baudino
Sent: Wednesday, August 17, 2005 12:13 PM
To: Tzafrir Cohen
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] X100P dial out problem
Hi Tzafrir,
thanks for your reply...
Here is what happens when I make the call:
pbx*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudoincomingit
1incomingit
-- Executing NoOp(SIP/6601-5d39, Call for 91234567) in new
stack
-- Executing Dial(SIP/6601-5d39, Zap/1/91234567) in new stack
-- Called 1/91234567
-- Zap/1-1 answered SIP/6601-5d39
-- Hungup 'Zap/1-1'
== Spawn extension (x-lite, 912334567, 2) exited non-zero on
'SIP/6601-5d39'
-- Unregistered SIP '6601'
pbx*CLI exit
The Hangup happens when I hangup from XLITE.
Here is my conf:
/etc/asterisk/zapata.conf
[channels]
language=it
signalling=fxs_ks
context=incoming
channel=1
/etc/asterisk/extensions.conf
[incoming]
exten = s,1,Dial(SIP/6601SIP/6602SIP/6603,20,tr) ; corresponding
clients must be configured in sip.conf
exten = s,2,Playback(vm-goodbye)
exten = s,3,Hangup
[outgoing]
exten = _9X.,1,NoOp(Call for ${EXTEN})
exten = _9X.,2,Dial(Zap/1/${EXTEN})
[x-lite] ; Note: SIP extensions are defined here as 66 followed by any
two digits
exten = _66XX,1,NoOp(Call for ${EXTEN})
exten = _66XX,2,Dial(SIP/${EXTEN})
exten = _66XX,3,Congestion
include = outgoing
/etc/asterisk/sip.conf
port=5060
context=default
srvlookup=yes
dtmfmode=inband
allow=aLaw
allow=uLaw
allow=gsm
[6601]
type=friend
secret=password
host=dynamic
;dtmfmode=rfc2833
context=x-lite
callerid=Piero 6601
allow=aLaw
allow=uLaw
allow=gsm
Thanks.
PieroB
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RE: [Asterisk-Users] dialplan defenition
For ZAP cards, you can tell Asterisk to answer calls immediately across
trunks. Does CAPI have the same type of setting? I am not familiar with
Asterisk and CAPI so I am not sure of the options.
In Zapata.conf, setting immediate=yes will make the call drop into the 's'
extension of the context. Setting immediate=no is supposed to make Asterisk
wait until a valid extension is dialed (I have had little to no success with
this portion of the setting).
If you can change a similar setting for CAPI, you should be able to drop
into a non-variable extension in the context (ie. i, s, t, etc.).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Wednesday, August 10, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] dialplan defenition
But to have a transparent integration with VoIP and legacy, I cant make
users dial twice... or having to whait for Asterisks dialtone, and dial
the number.
I whant to dial the 74XXX from a PBX extension (74118 for example) and
the IP phone rings.
Asterisk just need to forward the 74XXX calls, thats why I think the
solution is close to this:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
... but it always answers:
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid
extension 's' in context 'default', but no invalid handler
Why is CAPI sending it to 's' if I explicitly write
Dial(SIP/[EMAIL PROTECTED],30,r) ??
João
Matt Riddell wrote:
Joao Pereira wrote:
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and
has more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I
wrote this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
What is happening is that capi is sending it to s.
You will need to either set up an IVR, asking which number to send it to.
So, you would do the following:
exten = s,1,Answer()
exten = s,2,Background(pls-entr-extn)
exten = _74XXX,1,Dial(SIP/${EXTEN})
exten = _74XXX,2,Goto(s|1)
exten = _74XXX,102,Goto(s|1)
You will obviously need to record the pls-entr-extn sound.
You can do this by making an exten like this:
exten = 678,1,Record(pls-entr-extn)
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RE: [Asterisk-Users] ZAP bchan and dchan HELP!!
Did you setup your T1s as trunk groups? What channels are set up as d chans from the carrier? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Wednesday, August 10, 2005 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ZAP bchan and dchan HELP!! We have install a DS3 with 28 DS1's we have an Adtran MUX breaking out the DS1's, we are trying to setup the system with 2 dchannels for each 4 DS1's. Everything looks fine when modprobe zaptel and wct4xxp and ztcfg -vv but when I asterisk asterisk it says: Aug 10 16:33:32 ERROR[8954]: chan_zap.c:6750 mkintf: Channel 24 is reserved for D-channel. Aug 10 16:33:32 ERROR[8954]: chan_zap.c:9990 setup_zap: Unable to register channel '1-190' Aug 10 16:33:32 WARNING[8954]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 Do I HAVE to have a dchannel on every DS1? They want an extra $100 per dchannel. PLEASE HELP. I can have set the cards as follows: zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs span=5,0,0,esf,b8zs span=6,0,0,esf,b8zs span=7,0,0,esf,b8zs span=8,0,0,esf,b8zs bchan=1-190 dchan=191 dchan=192 loadzone = us defaultzone=us zapata.conf: switchtype = 5ess signalling = pri_cpe group = 2 context=internal channel = 1-190 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco IP Phone 30 VIP
The SEP file should be SEPMACADDR.cnf.xml You can also use XMLDefault.cnf.xml These have worked for me w/ 7960. What phone are you using? Here is some more information for reference: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Wednesday, August 10, 2005 5:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco IP Phone 30 VIP Sergio Chersovani wrote: Jason ha scritto: Could someone assist me in configuring this phone. It is saying in the CLI that its registered and saying its capabilities are recieved but i got no dialtone on the phone. Thanks are you using chan_skinny or chan_sccp? Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users chan_skinny atm but it dont matter to me, I think the main problem lies in a couple areas. 1. I cannot seem to find any kind of decent documentation on this phone with * and 2. I dont have the firmware or the SEPDefault.cnf binary file that is refered to on voip-info under the Cisco 12sp+/30VIP page. I am open to anything needed to get these phones working as the company i work for is non-profit and dont have much of a tech budget and I can get them for around $30 each. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ZAP bchan and dchan HELP!!
Where are the d chans in the trunk group? Which chan? Here is the example from the zapata.conf.sample ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ;group = trunkgroup,dchannel[,backup1...] ; ;trunkgroup is the numerical trunk group to create ;dchannelis the zap channel which will have the ;d-channel for the trunk. ;backup1 is an optional list of backup d-channels. ; ;trunkgroup = 1,24,48 ; ; Spanmap: Associates a span with a trunk group ;spanmap = zapspan,trunkgroup[,logicalspan] ; ;zapspan is the zap span number to associate ;trunkgroup is the trunkgroup (specified above) for the mapping ;logicalspan is the logical span number within the trunk group to use. ;if unspecified, no logical span number is used. ; ;spanmap = 1,1,1 ;spanmap = 2,1,2 ;spanmap = 3,1,3 ;spanmap = 4,1,4 Based on your info below, are 191 and 192 the d chans - primary and backup respectively? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Wednesday, August 10, 2005 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ZAP bchan and dchan HELP!! They want to setup a 1 primary and 1 backup dchannel for 4 T1's No can you give me an example on how to do this? I'm looking through the files now. trunkgroup = 1, 191, 192 spanmap = 1,1 spanmap = 2,1 spanmap = 3,1 spanmap = 4,1 spanmap = 5,1 spanmap = 6,1 spanmap = 7,1 spanmap = 8,1 something like this? Kyle Did you setup your T1s as trunk groups? What channels are set up as d chans from the carrier? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Wednesday, August 10, 2005 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ZAP bchan and dchan HELP!! We have install a DS3 with 28 DS1's we have an Adtran MUX breaking out the DS1's, we are trying to setup the system with 2 dchannels for each 4 DS1's. Everything looks fine when modprobe zaptel and wct4xxp and ztcfg -vv but when I asterisk asterisk it says: Aug 10 16:33:32 ERROR[8954]: chan_zap.c:6750 mkintf: Channel 24 is reserved for D-channel. Aug 10 16:33:32 ERROR[8954]: chan_zap.c:9990 setup_zap: Unable to register channel '1-190' Aug 10 16:33:32 WARNING[8954]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 Do I HAVE to have a dchannel on every DS1? They want an extra $100 per dchannel. PLEASE HELP. I can have set the cards as follows: zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs span=5,0,0,esf,b8zs span=6,0,0,esf,b8zs span=7,0,0,esf,b8zs span=8,0,0,esf,b8zs bchan=1-190 dchan=191 dchan=192 loadzone = us defaultzone=us zapata.conf: switchtype = 5ess signalling = pri_cpe group = 2 context=internal channel = 1-190 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco IP Phone 30 VIP
Misread the type of phone...sorry about that -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Wednesday, August 10, 2005 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco IP Phone 30 VIP those phones don't use .xml like the 7960s http://voip-info.org/tiki-index.php?page=Configuring%20Cisco%2012SP%20phones %20with%20Asterisk On Wed, 2005-08-10 at 16:49, Jason Walker wrote: The SEP file should be SEPMACADDR.cnf.xml You can also use XMLDefault.cnf.xml These have worked for me w/ 7960. What phone are you using? Here is some more information for reference: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Wednesday, August 10, 2005 5:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco IP Phone 30 VIP Sergio Chersovani wrote: Jason ha scritto: Could someone assist me in configuring this phone. It is saying in the CLI that its registered and saying its capabilities are recieved but i got no dialtone on the phone. Thanks are you using chan_skinny or chan_sccp? Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users chan_skinny atm but it dont matter to me, I think the main problem lies in a couple areas. 1. I cannot seem to find any kind of decent documentation on this phone with * and 2. I dont have the firmware or the SEPDefault.cnf binary file that is refered to on voip-info under the Cisco 12sp+/30VIP page. I am open to anything needed to get these phones working as the company i work for is non-profit and dont have much of a tech budget and I can get them for around $30 each. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P Cable Pin Out
I have had to create two different types of connections depending on what I connect any of the TE4XX cards and the TE1XX card. What are you connecting this to? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Dracevich Sent: Tuesday, July 26, 2005 7:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE110P Cable Pin Out I have just got a TE110P card, and I need the cable pin out. Thanks -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 7/25/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbiew extensions.conf question
If all of your extensions are in the same schema (i.e. 7## or 7###) you
could do this:
Exten = _7XX,1,Dial(DEVICE/${EXTEN})
Exten = _7XX,2,Voicemail(u${EXTEN})
This would allow for any 7## number to call into the extension. ${EXTEN} is
the variable for the extension dialed. I am using DEVICE in case you
decide to use other methods or protocols - IAX/2, Zap, etc.
Hope that helps.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kenny Kant
Sent: Thursday, August 04, 2005 3:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server. I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc
in my from-sip context I began doing entries such as:
exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)
exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)
..etc ..etc
This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck. Is there anyway around
this?
Thanks!
Kenny
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
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RE: [Asterisk-Users] Outbound Extension problem
Can you post your macro? Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim King Sent: Thursday, August 04, 2005 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Outbound Extension problem New problem, I figured out how to get the extension working and internally it works just fine. If I pick up a phone and hit 501 my cell starts ringing. However if an inbound caller dials that extension Everything seems to stop when it trys to bridge the two trunks together. Sound familiar to anyone? exten = 501,1,Macro(dialout-trunk,1,5551212) exten = 501,2,Wait,1 exten = 501,3,Voicemail(300) Thanks Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbiew extensions.conf question
That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.
I don't follow your logic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question
Right track, but it can be simplified even more
exten = 720,1,macro(sipexten,${EXTEN})
On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:
We handled it by creating a macro which dials the exten, then sends
the call to voicemail.
You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc
or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})
then the macro would look something like
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)
exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to
voicemail, play unavailable message
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,VoiceMail(b${ARG1});Send to
voicemail, play busy message
exten = s-BUSY,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
Depends on your needs which way would work better. We define
extension by extension individually, then have a wildcard at the
end that plays a message that says the extension is not in use and
then puts them in our main menu. In case we have to remove or
change an extension individually.
Louie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny
Kant
Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server. I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc
in my from-sip context I began doing entries such as:
exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)
exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)
..etc ..etc
This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck. Is there anyway around
this?
Thanks!
Kenny
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
___
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RE: [Asterisk-Users] newbiew extensions.conf question
But why do it that way?
Wouldn't:
exten = _72X,1,Dial(SIP/${EXTEN},50)
Be ideal? Or at least an easier way to expand the dialplan without mucho
administration?
Just a question...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie
Sent: Thursday, August 04, 2005 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.
You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})
and so forth.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.
I don't follow your logic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question
Right track, but it can be simplified even more
exten = 720,1,macro(sipexten,${EXTEN})
On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:
We handled it by creating a macro which dials the exten, then sends
the call to voicemail.
You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc
or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})
then the macro would look something like
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)
exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to
voicemail, play unavailable message
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,VoiceMail(b${ARG1});Send to
voicemail, play busy message
exten = s-BUSY,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
Depends on your needs which way would work better. We define
extension by extension individually, then have a wildcard at the
end that plays a message that says the extension is not in use and
then puts them in our main menu. In case we have to remove or
change an extension individually.
Louie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny
Kant
Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server. I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc
in my from-sip context I began doing entries such as:
exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)
exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)
..etc ..etc
This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck. Is there anyway around
this?
Thanks!
Kenny
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
___
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RE: [Asterisk-Users] ip phones
Soft phones or hard phones? There are many free VOIP soft phones out there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, August 04, 2005 9:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ip phones Hello, I want to setup asterisk and do VOIP. Somebody from US has offered to get me ip phones. Can anybody suggest a few good and resonably priced phones models. Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_rxfax errors
Up until today, I have had no issues with receiving faxes in *. One
change I made was that I now have the incoming DIDs "macro"'d since
they all start with 3 (3###).
>From /var/log/asterisk/messages
Aug 2 10:26:58 NOTICE[14938]: Unable to find
a path from unknown to unknown
Aug 2 10:26:58 WARNING[14938]: Unable to restore read format on
'Zap/41-1'
>From the console:
Aug 2 11:07:20 NOTICE[14938]: channel.c:1736
ast_set_read_format: Unable to find a path from unknown to unknown
Aug 2 11:07:20 WARNING[14938]: app_rxfax.c:256 rxfax_exec: Unable to
restore read format on 'Zap/41-1'
Has anyone come across the errors above, and if so what did you do
to correct?
When I explicitly set up a fax line to receive calls, no problems -
here is the dialplan:
exten =
3417,1,Macro(fax-receive,${EXTEN},${CALLERIDNUM})
exten = 3417,2,Hangup
[macro-fax-receive]
; $ARG1 is the extension called
; $ARG2 is the caller ID number
exten = s,1,Answer
exten = s,2,Ringing
exten = s,3,Wait(2)
exten = s,4,NoOp(${ARG1} ${ARG2})
exten = s,5,SetVar(FAXUNIQ=${ARG2}_${ARG1}_${UNIQUEID})
exten = s,6,SetVar(FAXFILE=/var/spool/asterisk-fax/${FAXUNIQ}.tif)
exten = s,7,rxfax(${FAXFILE})
exten = s,8,GotoIf($["${CALLERIDNUM}" != ""]?9:11)
exten = s,9,System(/usr/local/sbin/convertSendPDF.sh ${FAXUNIQ}
${ARG2} ${ARG1})
exten = s,10,Goto(macro-fax-receive,12)
exten = s,11,System(/usr/local/sbin/convertSendPDF.sh ${FAXUNIQ}
"NOCALLERID" ${ARG1})
exten = s,12,Hangup
This works without any issues.
Now when I do this:
exten 1-5 do some Mysql stuff to translate DNIS to DID
exten = _3XXX,6,Answer
exten = _3XXX,7,Ringing ; If a fax, dialplan redirects to the
fax extension in this context
exten = _3XXX,8,NoOp(${DNID})
exten = _3XXX,9,Wait(1)
exten = _3XXX,10,Dial(${TIE1}/${OutDID},150)
exten = _3XXX,11,Hangup
; Fax detected
exten = fax,1,Macro(dual-did-fax,${DNID},${CALLERIDNUM})
exten = fax,2,Hangup
[macro-dual-did-fax]
; $ARG1 is the extension called
; $ARG2 is the caller ID number
exten = s,1,Wait(1)
exten = s,2,NoOp(${ARG1} ${ARG2})
exten = s,3,SetVar(FAXUNIQ=${ARG2}_${ARG1}_${UNIQUEID})
exten = s,4,SetVar(FAXFILE=/var/spool/asterisk-fax/${FAXUNIQ}.tif)
exten = s,5,rxfax(${FAXFILE})
exten = s,6,GotoIf($["${CALLERIDNUM}" != ""]?9:11)
exten = s,7,System(/usr/local/sbin/convertSendPDF.sh ${FAXUNIQ}
${ARG2} ${ARG1})
exten = s,8,Goto(macro-fax-receive,12)
exten = s,9,System(/usr/local/sbin/convertSendPDF.sh ${FAXUNIQ}
"NOCALLERID" ${ARG1})
exten = s,10,Hangup
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[Asterisk-Users] Issue with zapata.conf immediate setting
I currently have two channel groups in my zapata.conf file. I would like one group to be immediate=yes and the other immediate=no Does not seem to matter which way I go, the first entry in overrides my explicit setting for the second group. I am running * 1.0.9 on FC1 [trunkgroups] ;trunkgroup = 1,24 trunkgroup = 1,48,72 ;spanmap = 1,1,0 spanmap = 2,1,0 spanmap = 3,1,1 spanmap = 4,1,2 [channels] ; Tie line to Nortel context=tie_line_01 signalling=em_w rxwink=300 usecallerid=yes hidecallerid=no usecallingpres=yes rxgain=0 txgain=0 overlapdial=yes transfer=yes immediate=no group=1 callgroup=1 pickupgroup=1 amaflags=billing accountcode=tie_line_01 callprogress=yes busydetect=yes channel = 1-24 ; Qwest DID Lines context=qw_pri_01 switchtype=national signalling=pri_cpe pridialplan=national callerid=XX nsf=sdn rxwink=300 usecallerid=yes immediate=no hidecallerid=no usecallingpres=yes rxgain=0 txgain=0 group=2 callgroup=2 faxdetect=both pickupgroup=2 amaflags=billing accountcode=qwe_pri_01 callprogress=yes channel = 25-47,49-71,73-96 the purpose of this is to bridge our traditional voice PBX and connected digital phones to our * box with a tieline, as well as allow incoming DIDs to flow through the * box into the traditional PBX using the same tieline. In extensions.conf, I have a dialplan set up for the qw_pri_01 circuit/context for calls coming in to hit the tie line device. This works fine. Going from the PBX to *, I have an issue. We have an ACOD of 777 to hit that trunkgroup. After I dial 777, a simple switch is started (I can see it on the console). As soon as I dial any other number (like 83028 as a SIP phone), the 8 is usually the only number that gets picked up. There is no 8 extension in the tie line context, so I get a not in service message. If I set immediate to yes, I COULD default the call to the 's' extension and attempt to handle the additional characters/digits after answering (perhaps a Read cmd). If I have immediate set to yes for this channel group, than the qw_pri_01 group also acts like I set immediate yes in that group - regardless of immediate=no being set. This screws everything up as the calling party does not get anything returned to them for an extension to dial. I suppose I could set up a forced call - but I think setting up immediate=yes on my tieline and immediate=no on my DIDs is a better plan. Perhaps there is a better way? Something I am missing? Thank you in advance Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue/Agents
Joseph - I would love to see something like this if you are willing to share. Thanks. Joseph wrote: Hall, Eric M. wrote: Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. I can share mine. Shows a list of callers and agent status. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help Windows messenger configuaration
Are you calling an IP or an extension? JASON WALKER - Original Message - From: someshwarak To: asterisk-users@lists.digium.com Sent: Thursday, July 28, 2005 7:37 AM Subject: [Asterisk-Users] help Windows messenger configuaration Hi, I am trying to register windows messenger to Asterisk. My windows messenger gets registered succesfully but I am continously getting a error response of SIP 481 Call Leg/Transaction Does not exist to the NOTIFY sip message. here is the SIP error response packet Sip read: SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 10.50.25.119:5060;branch=z9hG4bK286a047c From: "Unknown" sip:[EMAIL PROTECTED]:5060<SIP:[EMAIL PROTECTED]>;tag=as18fde308 To: sip:10.50.25.25:6931;tag=2fc7d57b0537473d85b3be02644724a2 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: RTC/1.3 Content-Length: 0 <SIP:[EMAIL PROTECTED]> <SIP:[EMAIL PROTECTED]>1.Is there anyway I can stop this. Help will be very much appreciated. <SIP:[EMAIL PROTECTED]> <SIP:[EMAIL PROTECTED]>2.Did Asterisk also supports IM. If yes how to configure my dial plan for the same. <SIP:[EMAIL PROTECTED]><SIP:[EMAIL PROTECTED]> <SIP:[EMAIL PROTECTED]>thanks, <SIP:[EMAIL PROTECTED]>Somesh <SIP:[EMAIL PROTECTED]> ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MozIAX phone on FC4/Firefox 1.6
Thank you for the reply. You are correct with the version #. I mistyped... Regarding the network_client...I have checked the services file and there is no other appl or daemon using 9998. I did add IAX as 4569/tcp to that file to see if that would help. After reviewing the output of the command you mention below, the fatal error is that iaxclient can not initialize. Apparently this is required. I must have missed that set of instructions from the moziax website. I will attempt to get iaxclient installed - however I was not able to get that application fully installed before. Make errors abound. If you have any suggestions, I would appreciate any assistance. Thank you, Jason -Original Message- From: Jean-Denis Girard [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 26, 2005 9:31 PM To: Jason Walker; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MozIAX phone on FC4/Firefox 1.6 Jason Walker a écrit : Has anyone had any luck with MozIAX (Mozphone) on FC4 with Firefox 1.6? jslib and moziax install through Firefox correctly - at least that is the message I get. I guess this is Firefox-1.0.6, or I must have been sleeping to long ;) I am able to log into the IAX Phone on Windows, however I get an error stating: -- FATAL ERROR: no connection to network_client. MozPhone will stop now! -- I am able to connect with the same connection settings on a Windows 2000 PC running Firefox 1.6 with MozPhone. Calls are successful in that environment. Any ideas? network_client uses port 9998, maybe there is already a service running on that port which prevents network_client from starting. Could you start firefox -phone from a console ? You should see some messages like the following: Setting up network input ProxIAX network_client waiting on port 9998 Client connection... Greeting client with Hello CVS-2005/07/03-13:20 now... Connecting|Looking up asterisk.sysnux.pf gui_hide_doing Connecting...|Connecting to asterisk.sysnux.pf gui_hide_doing Logging in|Logging in astman '... gui_hide_doing device,0,/dev/dsp,63 . I use Mozphone all day long with Firefox-1.0.6 on Mandriva-10.2. Thanks for using MozPhone, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MozIAX phone on FC4/Firefox 1.6
Has anyone had any luck with MozIAX (Mozphone) on FC4 with Firefox 1.6? jslib and moziax install through Firefox correctly - at least that is the message I get. I am able to log into the IAX Phone on Windows, however I get an error stating: -- FATAL ERROR: no connection to network_client. MozPhone will stop now! -- I am able to connect with the same connection settings on a Windows 2000 PC running Firefox 1.6 with MozPhone. Calls are successful in that environment. Any ideas? Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should this work?
Have you defined the context "default" in the
extensions.conf for outbound dialing in the globals section?
For example, I have my ZAP channels identified as
OUTBND1 not ZAP in the global section. This new global identifier is pointed to
ZAP/g1
[globals]
OUTBND1=Zap/g1
Instead of ZAP in my dial plan to call out, I use
${OUTBND1}.
Yours:
; for dialing outbound - over ISDN line - this bit
does not work
exten =
_9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten = _9XX.,2,Hangup
Mine would look like
this
exten =
_9XX.,1,Dial(${OUTBND1}/${EXTEN},##)
exten = _9XX.,2,Hangup
This helps me to keep track of inbound T1s and
outbound T1s.
Also, you have 2 (2) priorities listed in your
example. You can't really do this.
JASON WALKER
- Original Message -
From:
Angus
Comber
To: asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 8:11 AM
Subject: [Asterisk-Users] Should this
work?
Hello
I am using a Junghans quadBRI ISDN card and it is
loaded and working. In Asterisk if I connect to ISDN line it is detected
and tells me so.
In my zapata.conf I have
(abbreviated):
[channels]
switchtype=euroisdn
signalling = bri_cpe
context=default
group=1
channel = 1-2
;plus group 2 - 4
zaptel.conf:
loadzone=ukdefaultzone=uk# qozap span
definitions# most of the values should be bogus because we are not really
zaptelspan=1,1,3,ccs,amispan=2,0,3,ccs,amispan=3,0,3,ccs,amispan=4,0,3,ccs,ami
bchan=1,2dchan=3bchan=4,5dchan=6bchan=7,8dchan=9bchan=10,11dchan=12
Then in extensions.conf I have:
[default]
; this below for internal extensions - works
OK
exten =
_2XX,1,Dial(SIP/${EXTEN},20,Ttm)
; for dialing outbound - over ISDN line - this
bit does not work
exten =
_9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten = _9XX.,2,Hangup
Error I get is:
-- Executing
Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stackJul 25 11:56:33
NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create channel of type
'ZAP' == Everyone is busy/congested at this
time -- Executing Hangup("SIP/200-e433", "") in new
stack == Spawn extension (default, 902088787367, 2) exited non-zero
on 'SIP/200-e433'
I am dialing with sip phones. They work if
dialing extensions internally but not if try to dial outside - eg dial 9
followed by number.
What have I not done right?
Angus
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Re: [Asterisk-Users] Soft Phone
Any suggestions for IAX phones on Linux (without Wine preferred)? Thanks, JASON WALKER - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 25, 2005 11:05 AM Subject: RE: [Asterisk-Users] Soft Phone On Mon, 2005-07-25 at 17:17 +0200, Alex Ongena wrote: Any recommendation for Linux environments (without WINE) ? Thanks Alex Xten runs on linux. http://xten.com/index.php?menu=productssmenu=download -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queues and roundrobin/rrmemory
Round robin is designed to alternate between, in this case, the two agents. At least that is how I understand the comment in the queues.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Thursday, July 21, 2005 11:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] queues and roundrobin/rrmemory I have a queue setup using Asterisk CVS and roundrobin, however calls seem to be distributed in the same way as rrmemory (round robin with memory), ie, it is alternating between the two people in the queue rather than always calling the first available person in the queue first. I am using agents with agentcallbacklogin and addqueuemember to dynamically add the agent to the queue. asterisk version: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 2005-06-07 07:34:45 Does anyone use agents + agentcallbacklogin and use roundrobin queues with a recent CVS and have it working (or have the same problem ??) Thanks, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival questions
Has anyone had any luck in changing the voices for Festival and Asterisk? I have Festival installed and working, but can not get the voice different from the default. Thanks, Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Archer Sent: Wednesday, July 13, 2005 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Festival questions I'm working on this now. I don't expect it to be too useful though. --On Wednesday, July 13, 2005 3:47 PM -0400 [EMAIL PROTECTED] wrote: Hi, Is it possible to setup an Asterisk system that can allow someone to dial in using a DID and listen to their e-mail? Has anyone done this? Thanks, Mike C. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SpanDSP rxfax, no tiff.
I may be a little late on this, but what permissions are on
/usr/local/sbin/mailfax?
I have a similar set up to execute a mysql query to grab
the email address based on DNIS (PRI T1 with multiple numbers on one circuit)
and then email the fax to the destination. I set the perm to 755 on the script
so everyone/thing can execute.
Also, what are the perms on
/var/spool/asterisk/asterisk-fax?
Can you run the script from the command line by passing it
the appropriate values (i.e. /usr/local/sbin/mailfax
/var/spool/asterisk/asterisk-fax/#.#.tiff [EMAIL PROTECTED]?
In the event that something weird is going on with the
command line parameters, here are some considerations:
If the folder for the TIFFs is always the same, you could do a Set (or
SetVar depending on your Asterisk build) to have the UNIQUEID passed only to
the script
If the email reciepients are always on the same domain, you need to
only pass the name portion of the email address
For example:
Extensions.conf
section ---
[fax]
exten =
s,1,Answer
exten =
s,2,Macro(faxreceive)
;exten =
h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR})
This line could go away
[macro-faxreceive]
exten =
s,1,Set(FAX_OUT=${UNIQUEID})
exten = s,2,Set(FAXFILE=/var/spool/asterisk/asterisk-fax/${FAX_OUT}.tif)
exten = s,3,rxfax(${FAXFILE})
exten = s,4,system(/usr/local/sbin/mailfax
${FOX_OUT} MyName)
;exten =
s,3,Set([EMAIL PROTECTED])
This line could go away
Assuming #!/bin/bash
;)
#!/bin/bash
# $ARG1 is the
TIFF file name
# $ARG2 is the
name of the domain email user
EMAIL_ADDR=$2"@mycompany.com"
FAX_FILE="/var/spool/asterisk/asterisk-fax/"$1
# Do the sendmail
thing here
#--
Just my 40
cents.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob
DanzSent: Wednesday, July 13, 2005 8:18 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SpanDSP
rxfax, no tiff.
Hello,
Let me start by saying I have
checked the wiki and the archives and did find some relative information.
I tried the suggestions in those threads, but still have the same
problem.
Im using the CVS Asterisk from July
11, 2005.
Redhat
FC2
SpanDSP
0.0.2pre18
Libtiff
3.5.7
Digium PCI card 1 FXO,
1FXS.
I have a single POTS line coming,
but I have 2 numbers and am using distinctive ring detection in
*.
When you call my fax number, the
ring detection does work, and does send it to the fax context
correctly.
The debugs show the call is
answered, rxfax is invoked and it is trying to write to the fax file.
After the sending party hangs up, it tries to execute a script that will
ultimately mail me the fax file. But since the tiff file isnt there to
begin with, that fails. The permissions on that folder are 777 for now so
permissions arent the problem.
I saw a post by Steve Underwood from
last year on a similar problem, but it was looking like timing slips on the
T1/E1 for that user Im just using a POTS line though. Ive also done
ztmonitor to look at the Rx and Tx levels. Rx is a little hotter than Tx,
but theyre both well on the right hand side of the scale.
Any help is appreciated.
Debugs extensions.conf excerpt are below.
Thanks,
Rob
Debug output
---
Jul 13 10:04:34 NOTICE[7975]:
chan_zap.c:5759 ss_thread: Got event 2
(Ring/Answered)...
-- Detected ring
pattern: 93,0,0
-- Distinctive
Ring matched context fax
-- Executing
Answer("Zap/4-1", "") in new stack
-- Executing
Macro("Zap/4-1", "faxreceive") in new stack
-- Executing
Set("Zap/4-1", "FAXFILE=/var/spool/asterisk/asterisk-fax/1121267067.12.tif") in
new stack
-- Executing
RxFAX("Zap/4-1", "/var/spool/asterisk/asterisk-fax/1121267067.12.tif") in new
stack
-- Executing
System("Zap/4-1", "/usr/local/sbin/mailfax
/var/spool/asterisk/asterisk-fax/1121267067.12.tif ") in new
stack
Jul 13 10:05:03 WARNING[7975]:
app_system.c:75 system_exec_helper: Unable to execute '/usr/local/sbin/mailfax
/var/spool/asterisk/asterisk-fax/1121267067.12.tif
'
-- Hungup
'Zap/4-1'
Extensions.conf section
---
[fax]
exten =
s,1,Answer
exten =
s,2,Macro(faxreceive)
exten =
h,1,system(/usr/local/sbin/mailfax ${FAXFILE}
${EMAILADDR})
[macro-faxreceive]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/asterisk-fax/${UNIQUEID}.tif)
exten =
s,2,rxfax(${FAXFILE})
exten =
s,3,Set([EMAIL PROTECTED])
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RE: [Asterisk-Users] Problems with a new box of [EMAIL PROTECTED] 1.3
Are you getting any messages from the CLI on * pertaining to a sip user not registering? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fabrizzio ValenciaSent: Sunday, July 10, 2005 7:45 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Problems with a new box of [EMAIL PROTECTED] 1.3 Hello, I've recently installed [EMAIL PROTECTED], i'm following step by step the "new user guide" but I cannot get my X-Lite SIP phone see my [EMAIL PROTECTED] proxy... I've installed in aviertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Thanks in advance. Fabrizzio Valencia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme recordings
I have a conference set up through MeetMe and I can record each call coming in with the Monitor command. What I would like to move away from is having to then generate multiple files for the final output of these calls. On voip-info.org, there is an 'r' option to record the conference. This does not work on my 1.0.7 version of Asterisk. I looked through the app_meetme.c file and the option is not there either. As a reference, here is a link to the page on voip-info.org that I am refering to: http://voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe I have also setup a separate extension to dial intoin an attempt to record all of the members from one source. What I have found is that the first monitor session records all subsequent members of the conference. For example: Three members log in Member one records all members Member two records two and three Member three records member three I guess my question is what happened to the 'r' recording option in meetme? Thanks, Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP to PRI
Ummm, yes. I am not quite following what your question is asking. However...I have a PRI line pushing inbound calls to SIP users. Can you expand on your question? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Equipe du Royaume Sent: Wednesday, June 15, 2005 7:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP to PRI Hi I can provide my customers one or several phone lines by using an ATA through the SIP protocol. Is there a similar box that would allow me to provide a PRI (23B +D) to a customer using SIP ? Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.3/15 - Release Date: 6/14/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [PRI] TE110P
I have setup a TE110P with a different carrier. The pri switch setting I used was national. I think this will work with NI1 or NI2. Interestingly enough, I have to use this against a ATT 4ESS carrier switch. The number of digits outpulsed is usually a ten digit number. The version of the card can be found in the messages log file when the card is activated with the ztcfg -v command. Hope that helps. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael L. YoungSent: Tuesday, June 14, 2005 7:07 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] [PRI] TE110P We are in the process of installing a PRI line and we are going to connect it to an Asterisk Box. Verizon called us today to find out some information. I am surprised that they have never heard of Asterisk or Digium. But anyways, they needed some information in order to set up the circuit. Does the TE110P support NI1 or NI2? (I think the answer is both)What is the number of digits outpulsed? Is there a version number on the TE110P card? Thank you in advanced to anyone who might be able to answer these questions for me. I tried to explain to Verizon that it would work and that the system is flexible but they want to make sure of it before they setup the circuit. Michael No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.2/14 - Release Date: 6/14/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ztcfg server crash
What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be partially screwed. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.9 - Release Date: 6/11/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
