Re: [Asterisk-Users] [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'
Before or after you compiled zaptel and asterisk? It needs to be installed before you build everything else. after :-( thanks for your reply! ___ --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 sec delay from callee answering to call established on dialout
Hi, I've been using Asterisk (now version 1.2.4) for quite a while, and I'm trying to switch from POTS lines to a VOIP termination service. I've had this problem with a few services I've tried, so the problem must be on my end. Here's what I see: When dialing out, I hear rings, and the call connects. But by the time the call connects, the callee has already said 'hello' and there's just silence while he waits for me to speak. (One of my friends has already learned that it's me calling when that happens!) I wish I had more information, but that's about it. I've seen some other posts about this that went unresolved. This didn't happen when I used to use the FXO card. If it's relevant, this happens with my soft phone, my cordless (and FXS card), and cell phone (dialing in then DISAing out). This also happens on incoming calls, but I care less since I have the Wait command. Any help greatly appreciated! John ___ --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] Grandstream GXP-2000
Worth saying that the Aastra 9133i with the 1.3.1 firmware is a pretty good phone. I used to run GXP-2000's, still have 10 new in a box and another 20 in demo/test circulation, but I also run a few dozen 9133i, 480i and 9112i phones and I think Aastra are getting their now. Biggest problem I had with GXP are the usual power flakyness, which you can't really do much about but apart from that no real problems. Now the GXP firmware is getting there might offer them as a cheaper phone to the 9133i. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 19 February 2006 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream GXP-2000 On Sat, 18 Feb 2006, Michael J. Liberatore wrote: Well the gxp-2000 has BLF, the polycom 501 does not correct? I had an astra 480i and it was prety bad, but I was going to test the 9133i for an inexpensive phone to compete with the gxp2000. The gxp2000 is not bad though, the new firmware helps a lot, but once they work out the echo bugs fully and the various minor stuff it will be a good sub $100 phone. I am yet to find a phone under $300 that's perfect... The snom 360 is nice, but I have lots of problems with those too. I havent tried any polycom's though and starting to think they might be some of th ebest... The GXP2000 is good value for the money. It is not a great phone but for your $80 you get a lot more than one would expect. 7 programmable buttons with BLF, Backlight, dual 100bt. Stuff you dont find on some phones over twice the price... All phones have their warts, even cisco. For $80 I can live with the GXP2000's warts, grandstream do seem to be actively improving the firmware and fixing what they can. Asterisk features (mwi, blf) just work out of the box without the gyrations one has to go through for other vendors phones. I have some $200+ phones which have some serious warts and the vendors do not seem terribly interested in fixing them. Big money does not always mean good value. -Dan ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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: Firmware version 1.3.1 released for AastraIPphones
OK, well the audio option was the last one I required for now. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Owen Sent: 19 February 2006 16:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones The short answer is all the officially supported configuration parameters are in the admin guide and release notes. Options that aren't documented aren't guaranteed to work between releases. So, sorry but the current documentation contains all the config options. Gareth -Original Message- From: [EMAIL PROTECTED] on behalf of Lee Archer Sent: Fri 2/17/2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones Nice one it works. Is there a complete list of all the options you can use in the config files? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Owen Sent: 17 February 2006 13:39 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IPphones The follow should work from the configuration files (aasta.cfg/MAC.cfg), although I haven't tried it... audio mode: mode Where mode is a number between 0 and 3 0 = speaker 1 = headset 2 = speaker/headset 3 = headset/speaker Gareth Lee Archer wrote: Any chance of getting a config option in that allows you set headset/speaker in the audio menu? Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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] spa3000
I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret= context=from-pstn auth=md5 [PSTN_2] username=line2 type=peer secret= qualify=yes port=5061 nat=no host=192.168.0.20 context=from-pstn canreinvite=no auth=md5 The sip debug says this: -- SIP read from 192.168.0.20:5061: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-fc209f89 From: sip:[EMAIL PROTECTED];tag=be447a1af149c461o1 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-3.1.7(GWg) Content-Length: 426 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 23156020 23156020 IN IP4 192.168.0.20 s=- c=IN IP4 192.168.0.20 t=0 0 m=audio 16478 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 19 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.20 : 5061 (non-NAT) Found peer 'PSTN_2' Reliably Transmitting (no NAT) to 192.168.0.20:5061: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-fc209f89;received=192.168.0.20 From: sip:[EMAIL PROTECTED];tag=be447a1af149c461o1 To: sip:[EMAIL PROTECTED]:5060;tag=as535b07db Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=5a2eee21 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms server*CLI -- SIP read from 192.168.0.20:5061: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-fc209f89 From: sip:[EMAIL PROTECTED];tag=be447a1af149c461o1 To: sip:[EMAIL PROTECTED]:5060;tag=as535b07db Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5061 User-Agent: Sipura/SPA3000-3.1.7(GWg) Content-Length: 0 --- (10 headers 0 lines)--- server*CLI -- SIP read from 192.168.0.20:5061: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-2e547cf5 From: sip:[EMAIL PROTECTED];tag=be447a1af149c461o1 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=linea2,realm=asterisk,nonce=5a2eee21,uri=sip:[EMAIL PROTECTED]:5060,algorithm=MD5,response=f18750c7e09707b6e76e0c6c08f10b77 Contact: sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-3.1.7(GWg) Content-Length: 426 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 23156020 23156020 IN IP4 192.168.0.20 s=- c=IN IP4 192.168.0.20 t=0 0 m=audio 16478 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 19 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.20 : 5061 (non-NAT) Found peer 'PSTN_2' Reliably Transmitting (no NAT) to 192.168.0.20:5061: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-2e547cf5;received=192.168.0.20 From: sip:[EMAIL PROTECTED];tag=be447a1af149c461o1 To: sip:[EMAIL PROTECTED]:5060;tag=as535b07db Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- server*CLI -- SIP read from 192.168.0.20:5061: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-2e547cf5 From: sip:[EMAIL PROTECTED];tag=be447a1af149c461o1 To: sip:[EMAIL PROTECTED]:5060;tag=as535b07db Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username=linea2,realm=asterisk,nonce=5a2eee21,uri=sip:[EMAIL PROTECTED]:5060,algorithm=MD5,response=104aa010d2f90b4a69c56b0ebf0991d3 Contact: sip:[EMAIL PROTECTED]:5061 User-Agent: Sipura/SPA3000-3.1.7(GWg) Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines Reliably Transmitting (no NAT) to
RE: [Asterisk-Users] Intro and first questions
I'm a newbye myself so beware! (1) http://www.voip-info.org is your friend. I've got most of my info off that site and it's a good place to start. (2) Download a Softphone like XLite (you'll also find info on softphones on voip-info) and start experimenting on site. When you'll be able to configure your softphone to call an other softphone on a different machine you'll be on your way to setting up the link with your daughter. (3) If you've got a bit of experience with Linux and it's style of configuration files stay away from automated GUI's like AMP and stuff as they add an other level of abstraction on top of an already complex thing. Resolving the issues that you'll probably run into will be a lot easier if you typed the whole configuration files your self (as opposed to having them generated by things like AMP). Out of my experience, after staring with a fresh install of [EMAIL PROTECTED] I had to basically DELETE everything in my extensions.conf (the dial plan) as I was unable to make any sense of it. It was a complex thing generated by AMP. I'm sure it was much better then my own but I was plain simply unable to understand it! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tom Poe Sent: Saturday, February 18, 2006 6:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Intro and first questions This is my first venture into VoIP from my Fedora Core 4 system. I came across a posting: http://atrpms.net/name/asterisk/ http://atrpms.net/name/asterisk-addons/ http://atrpms.net/name/asterisk-sounds/ http://atrpms.net/name/spandsp/ http://atrpms.net/name/libpri/ http://atrpms.net/name/zaptel/ on the Fedora list, added the repository to yum, and downloaded, installed, then typed: # asterisk -c , hit the return, and a bunch of stuff happened, before returning to the root prompt. My first goal(s) is to be able to configure the machine to make a PC to PC call to my daughter, who lives in Minnesota. If all goes well, I can set up her computer to receive the call, using Asterisk. Is this a realistic first experience project? If so, is there a tutorial out there that describes the steps I need to take? Any advice, suggestions, greatly appreciated. Tom -- 94% of returning troops suffer from trauma Open Studios http://www.ibiblio.org/studioforrecording/ ___ --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] MixMonitor and command
Yes, you need to remove the 'System' part.
You should only have:
exten =
s,n,MixMonitor(${CALLDIR}${CALLFILENAME}.wav||touch/tmp/test${UNIQUEID})
Garth
Alex Barnes wrote:
Has anyone had any success using the MixMonitor() plus command as
nothing I have tried works.
I am using 1.2.1 I did google the archive but couldn't see any mention
of anyone using this. What I am hoping to do is run a macro on hangup,
current method I am using seems to miss some calls 5% of calls fail to
mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.
exten = s,n,MixMonitor(${CALLDIR}${CALLFILENAME}.wav||System(touch
/tmp/test${UNIQUEID}))
exten = s,n,Answer
exten = s,n,SayDigits(1234)
exten = s,n,StopMonitor()
exten = s,n,Hangup()
Output:
-- Executing MixMonitor(Zap/1-1,
/tmp/callrec/20060217-212722-1-IN.wav||System(touch
/tmp/test1140211642.11373)) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SayDigits(Zap/1-1, 1234) in new stack
-- Playing 'digits/1' (language 'en')
== Begin MixMonitor Recording Zap/1-1
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/4' (language 'en')
-- Executing StopMonitor(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (macro-test-script, s, 14) exited non-zero on
'Zap/1-1' in macro 'test-script'
== Spawn extension (from-outside-547551-tl-allhours, s, 1) exited
non-zero on 'Zap/1-1'
== End MixMonitor Recording Zap/1-1
== Executing [System(touch /tmp/test1140211642.11373)]
-- Hungup 'Zap/1-1'
However listing /tmp reveals no files. Running macros that only print
NoOp's don't work either.
Thanks for the help
Alex
---
Alex Barnes
Engineering Support
Ubiquity Software
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RE: [Asterisk-Users] automatically start application from thecommandprompt
Thankx MC,
This is the solution.
Ive tried it and it works perfect.
But Ive got a question.
I want to set a variable with the command
SetVar
I place the following text file in the
directory /var/spool/asterisk/outgoing/
Channel:
Zap/g1/0655871460
MaxRetries:
0
RetryTime:
30
WaitTime:
30
Context:
call_outbound
Extension:
s
Priority:
1
SetVar:
call_outbound_id=0
When I tried to read the variable
call_outbound_id in the context call_outbound I can not see the value. ( exten
= s,6,NoOp(${call_outbound_id}) )
Is this the right solution, or do I have
to use the option Data?
Kind regards,
Arjan Kroon
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Collins
Sent: maandag 13 februari 2006
21:06
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
automatically start application from thecommandprompt
This can also be done with the use of
call files. Check this out:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
-MC
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Arjan Kroon
Sent: Monday, February 13, 2006
7:10 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
automatically start application from the commandprompt
Hello,
Is it possible to start an asterisk
application from the command prompt?
This application has to dial to a number.
When the calling party picks up the phone,
the asterisk application had to play certain voicefiles.
Kind Regards,
Arjan Kroon
Mobillion B.V.
Copernicuslaan 30
Postbus 554 / PO Box
554
6710 BN Ede
tel: +31 (0)318-648920
fax: +31 (0)318-648839
mobile: +31 (0)6-55871460
email: [EMAIL PROTECTED]
internet: www.mobillion.nl
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RE: [Asterisk-Users] Grandstream GXP-2000
On Mon, 20 Feb 2006, Lee Archer wrote: Worth saying that the Aastra 9133i with the 1.3.1 firmware is a pretty good phone. I used to run GXP-2000's, still have 10 new in a box and another 20 in demo/test circulation, but I also run a few dozen 9133i, 480i and 9112i phones and I think Aastra are getting their now. Biggest problem I had with GXP are the usual power flakyness, which you can't really do much about but apart from that no real problems. Now the GXP firmware is getting there might offer them as a cheaper phone to the 9133i. I think if grandstream spent a bit more on quality construction and parts they could have an awesome phone. Similar to the difference between the sipura 841 and the linksys 941. I would still like to know what they were smoking when they put two _10 meg_ ethernet ports on the linksys 942. -Dan ___ --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] Grandstream GXP-2000
On Mon, 2006-02-20 at 01:22 -0800, [EMAIL PROTECTED] wrote: I would still like to know what they were smoking when they put two _10 meg_ ethernet ports on the linksys 942. Probably the let's not cannibalize the 79xx series pipe. Wouldn't surprise me if the Ethernet chip is capable of doing 100Mb but is forced to 10Mb in the firmware. Regards, Patrick ___ --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] chan_capi setting ${DNIS}
On Mon, 20 Feb 2006, Nathan Alberti wrote:
Is there a reason the variable ${DNIS} does not get set with incoming calls
via chan_capi ?
I don't know any channel setting DNIS. What are you expecting with that
variable?
Is it related to the MSN=X in capi.conf ?
No. msn= is obsolete and does not exist in chan_capi since many releases.
version = chan_capi-cm-0.6.3
example;
exten = _9555XX,1,NoOp, ${EXTEN}, ${DNIS}
If you mean the caller number, then try ${CALLERID(number)}
Armin
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Re: [Asterisk-Users] Grandstream GXP-2000
I have Data Sheet for 942 from Linksys web site. It says this on page 4 (close to bottom): Physical Interfaces: 2 100baseT RJ-45 Ethernet Ports (IEEE 802.3) And that was one of the reasons I was considering 942. Do you think the data sheet may be wrong? - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 11:44 AM Subject: RE: [Asterisk-Users] Grandstream GXP-2000 On Mon, 2006-02-20 at 01:22 -0800, [EMAIL PROTECTED] wrote: I would still like to know what they were smoking when they put two _10 meg_ ethernet ports on the linksys 942. Probably the let's not cannibalize the 79xx series pipe. Wouldn't surprise me if the Ethernet chip is capable of doing 100Mb but is forced to 10Mb in the firmware. Regards, Patrick ___ --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] Loops and Variables
trixter aka Bret McDanel wrote: On Sun, 2006-02-19 at 21:05 -0500, Doug Lytle wrote: trixter aka Bret McDanel wrote: since you have had a little time to play with this, was this the problem? Haven't had a chance yet, will look at it when I get into work this morning. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call parking hint
Hi, Is it possible to use the hint priority to allow call parking slots to be monitored on (for example) Snom indicator lamps? How do you refer to the slots (i.e., what is the channel) in the hint? - Mike ___ --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] Where can I get the tar.gz sources of libnewt?
Where can I get the tar.gz sources of libnewt? Reg, Anthony Azzopardi. ___ --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] Grandstream GXP-2000
On Mon, 20 Feb 2006, Soner Tari wrote: I have Data Sheet for 942 from Linksys web site. It says this on page 4 (close to bottom): Physical Interfaces: 2 100baseT RJ-45 Ethernet Ports (IEEE 802.3) And that was one of the reasons I was considering 942. Do you think the data sheet may be wrong? every reseller that is selling the 942 lists two 10mb ports. also, rather disturbingly the linksys press release[1] implies the 942 is PoE only (like the aastra 480i), no external power supply. if anyone has a 942 and can authoritatively state it has 100meg ports and supports non-PoE power source, i'd definitely like to know. -Dan [1] http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayoutpackedargs=c%3DL_News_C2%26cid%3D1136499819516pagename=Linksys%2FCommon%2FVisitorWrapper ___ --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] Re: SIP groups
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You can not define groups in sip.conf But there are, as you hint, other ways to solve the problem, like using queues or implementing it in dialplan logic. Do you have any example how to do that? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Re: Voicemail - direct call
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... aHR0cDovL3d3dy52b2lwLWluZm8ub3JnL3Rpa2ktaW5kZXgucGhwP3BhZ2U9QXN0ZXJpc2srY21k K1ZvaWNlTWFpbAoKSXQncyBhbGwgaW4gdGhlcmUKCk9uIDIvMTMvMDYsIFRvbWlzbGF2IFBhcuhp bmEgPHRwYXJjaW5hQGxhbWEuaHI+IHdyb3RlOgo+Cj4gSGkgbGlzdCEKPgo+IEhvdyB0byBzZW5k IGEgY2FsbCBkaXJlY3RseSB0byB2b2ljZW1haWwgcmVjb3JkaW5nPwo+Cj4gV2hlbiBJIHB1dCB0 aGlzCj4gZXh0ZW4gPT4gMzEzLG4sVm9pY2VNYWlsLHUyMjEKPiBPciB0aGlzCj4gZXh0ZW4gPT4g MzEzLG4sVm9pY2VNYWlsLGIyMjEKPiBJbiBteSBkaWFsIHBsYW4sIGNhbGxpbmcgcGVyc29uIGZp cnN0IGhlYXJzIGdyZWV0aW5nIG1lc3NhZ2UgKGJ1c3kgb3IKPiB1bnZpYWJsZSkuIEkgd291bGQg bGlrZSB0byBhdm9pZCBncmVldGluZyBtZXNzYWdlIChJIHdvdWxkIHBsYXkgc29tZXRoaW5nCj4g d2l0aCBQbGF5YmFjayBhcHBsaWNhdGlvbikuIElzIGl0IHBvc3NpYmxlPwo+Cj4KPiAtLQo+IFRv bWlzbGF2IFBhcmNpbmEKPiB0cGFyY2luYSNsYW1hLmhyCj4gX19fX19fX19fX19fX19fX19fX19f X19fX19fX19fX19fX19fX19fX19fX19fX18KPiAtLUJhbmR3aWR0aCBhbmQgQ29sb2NhdGlvbiBw cm92aWRlZCBieSBFYXN5bmV3cy5jb20gLS0KPgo+IEFzdGVyaXNrLVVzZXJzIG1haWxpbmcgbGlz dAo+IFRvIFVOU1VCU0NSSUJFIG9yIHVwZGF0ZSBvcHRpb25zIHZpc2l0Ogo+ICAgIGh0dHA6Ly9s aXN0cy5kaWdpdW0uY29tL21haWxtYW4vbGlzdGluZm8vYXN0ZXJpc2stdXNlcnMKPgo=___ --Bandwidth and Colocation provided by Easynews.com -- Thank you, but this is how I see your mail. How can I see it right? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Re: segmentation fault
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Hi Patrick,
I'm new to Linux, so can you please tell me how do you check how did Asterisk
died?
Thank you.
--
Tomislav Parcina
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Re: spandsp 0.0.2pre25
I have used version 0.0.2 every version from pre8 bar pre23 with 1.0.x and pre23, 25 with 1.2.2 and 1.2.4. My libtiff is 3.5.7 with asterisk 1.0.x and libtiff 3.7.1-6 with asterisk 1.2.2 and 1.2.4 I am of the personal opinion through experience that txfax talking to rxfax does not work, and that in any case trying to do more than 3 concurrent txfax is unreliable. I am uncertain of the upper limit of concurrent rxfax, but it is in excess of 12 on TE110p and 1stgen TE4XXp PRI cards. Craig - Original Message - From: Jesse Guardiani [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 12:20 PM Subject: [Asterisk-Users] Re: spandsp 0.0.2pre25 Craig Guy cguy at bigpond.net.au writes: Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4 to receive from analog fax machines. I have never yet been able to get rxfax working with txfax - my debugs when I try look like the logs in your email. Craig Perhaps I'm just being nitpicky, but you don't mention what version of spandsp you're using. pre20 rtfax - pre20 rxfax works fine here with asterisk 1.0.10 and 1.2.4. I tried using an analog fax machine with pre25 and asterisk 1.2.4 with no luck whatsoever. Unfortunately, I don't have the debug output from those attempts, but I could generate some if it would help. Jesse ___ --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] GSM GATEWAY
well, does this gateway support SIP?? and does it generate its own CDR? could you send the devices brocure/tech spec.?? thanks On 2/19/06, Sam Tam [EMAIL PROTECTED] wrote: Why not get 30 GSM Gateway from us at £60 each and then get an asterisk or some voip gateway like A800 and then link it all up From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Dumpolid ExeplishSent: Sunday, February 19, 2006 10:54 PM To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] GSM GATEWAY Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSMgateway(channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM Any Ideas?? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] call parking hint
On 2/20/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote: Hi, Is it possible to use the hint priority to allow call parking slots to be monitored on (for example) Snom indicator lamps? How do you refer to the slots (i.e., what is the channel) in the hint? You're looking for the metermaid patch available with /trunk at http://bugs.digium.com/view.php?id=5779 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Re: Call centre - * hang's up
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I think it's a bit of a known fault - the attended transfer function does not work from the queue system. It would be nice if it did, though. Hi Paul! Is there any explanation about this? Is that something that will change? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Re: Call centre - * hang's up
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You'll have to use uattended transfers for CCs. l. I have read Paul's mail. Is this bug or feature? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: Voicemail - direct call
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... aHR0cDovL3d3dy52b2lwLWluZm8ub3JnL3Rpa2ktaW5kZXgucGhwP3BhZ2U9QXN0ZXJpc2srY21k K1ZvaWNlTWFpbAoKSXQncyBhbGwgaW4gdGhlcmUKCk9uIDIvMTMvMDYsIFRvbWlzbGF2IFBhcuhp bmEgPHRwYXJjaW5hQGxhbWEuaHI+IHdyb3RlOgo+Cj4gSGkgbGlzdCEKPgo+IEhvdyB0byBzZW5k IGEgY2FsbCBkaXJlY3RseSB0byB2b2ljZW1haWwgcmVjb3JkaW5nPwo+Cj4gV2hlbiBJIHB1dCB0 aGlzCj4gZXh0ZW4gPT4gMzEzLG4sVm9pY2VNYWlsLHUyMjEKPiBPciB0aGlzCj4gZXh0ZW4gPT4g MzEzLG4sVm9pY2VNYWlsLGIyMjEKPiBJbiBteSBkaWFsIHBsYW4sIGNhbGxpbmcgcGVyc29uIGZp cnN0IGhlYXJzIGdyZWV0aW5nIG1lc3NhZ2UgKGJ1c3kgb3IKPiB1bnZpYWJsZSkuIEkgd291bGQg bGlrZSB0byBhdm9pZCBncmVldGluZyBtZXNzYWdlIChJIHdvdWxkIHBsYXkgc29tZXRoaW5nCj4g d2l0aCBQbGF5YmFjayBhcHBsaWNhdGlvbikuIElzIGl0IHBvc3NpYmxlPwo+Cj4KPiAtLQo+IFRv bWlzbGF2IFBhcmNpbmEKPiB0cGFyY2luYSNsYW1hLmhyCj4gX19fX19fX19fX19fX19fX19fX19f X19fX19fX19fX19fX19fX19fX19fX19fX18KPiAtLUJhbmR3aWR0aCBhbmQgQ29sb2NhdGlvbiBw cm92aWRlZCBieSBFYXN5bmV3cy5jb20gLS0KPgo+IEFzdGVyaXNrLVVzZXJzIG1haWxpbmcgbGlz dAo+IFRvIFVOU1VCU0NSSUJFIG9yIHVwZGF0ZSBvcHRpb25zIHZpc2l0Ogo+ICAgIGh0dHA6Ly9s aXN0cy5kaWdpdW0uY29tL21haWxtYW4vbGlzdGluZm8vYXN0ZXJpc2stdXNlcnMKPgo=___ --Bandwidth and Colocation provided by Easynews.com -- Thank you, but this is how I see your mail. How can I see it right? http://lists.digium.com/pipermail/asterisk-users/2006-February/146742.html ___ --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] ~1 sec delay from callee answering to call established on dialout
John Morris wrote: Hi, I've been using Asterisk (now version 1.2.4) for quite a while, and I'm trying to switch from POTS lines to a VOIP termination service. I've had this problem with a few services I've tried, so the problem must be on my end. Here's what I see: When dialing out, I hear rings, and the call connects. But by the time the call connects, the callee has already said 'hello' and there's just silence while he waits for me to speak. (One of my friends has already learned that it's me calling when that happens!) I wish I had more information, but that's about it. I've seen some other posts about this that went unresolved. This didn't happen when I used to use the FXO card. If it's relevant, this happens with my soft phone, my cordless (and FXS card), and cell phone (dialing in then DISAing out). This also happens on incoming calls, but I care less since I have the Wait command. Any help greatly appreciated! John Hi John, Are you dialing SIP to your Voip termination service? I have had this before with a SIP device, where the SIP device sends CAlll Progress instead of ringing. Do a SIP debug ___ --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] GSM GATEWAY
On 19 Feb 2006, at 14:54, Dumpolid Exeplish wrote: Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM I haven't got one, but from their web site it looks like they make a 'pure' model with no ISDN. http://www.2n.cz/products/gsm_gateways/isdn_pri_gsm_gateways/ stargate_gsm_gateways.html Basic unit (CPU, PSU, AUX and VoIP) - NEW VoIP interface 5070002E Is there something missing from that ? I'm curious, since I may need such a thing in a month or 2. T. Any Ideas?? ___ --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 http://www.westhawk.co.uk/ ___ --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] Problems mixing audio in queues and playing queue positions
Hi folks, Over the weekend I finally decided to upgrade one of our Asterisk systems from 1.0.9 to 1.2.4 I had no significant problems and all is well in general - as usual Asterisk rules! However, I did run into two small issues. Can anyone help me solve them please? The first one involves queue position announcements, and the second one is regarding monitor-join. A) In 1.0.9, as soon as a caller enters a queue they are played the position announcement (which is what I want) and then it is replayed every X seconds depending on what I have for announce-frequency in queues.conf This is not the case in 1.2.4 though. Effectively the queue position is not played until after the sum of times set for timeout and retry. e.g. from queues.conf: [myqueue] timeout = 10 retry = 5 wrapuptime=5 maxlen = 0 musiconhold = default strategy = ringall announce-frequency = 60 announce-holdtime = yes announce-round-seconds = 0 monitor-format = wav49 monitor-join = yes member = sip/phone1 member = sip/phone2 member = sip/phone3 With this queues.conf configuration, in 1.2.4 the caller won't get their queue position played until after they have been in the queue for 15 seconds, while in 1.0.9 they got it immediately. Any suggestions? I really think it makes more sense for it to be played immediately when the caller joins the queue rather than waiting for the first timeout, which for many configurations might be much longer than the 15 seconds in mine if timeout and retry are set to higher values. B) My second issue is that monitor-join = yes in queue.conf does not seem to work for me - I still get individual -in and -out files for calls in the queue. Admittedly I had this problem in 1.0.9 too, but not in 1.0.7 I don't think. A very significant bit of information here is that using the m option in Monitor() in extensions.conf does not work for me either (I still get individual -in and -out files). The correct soxmix command gets executed (at least it appears on the console) but does not actually have any effect on the files. Manually running the exact same command on the command line does work, and joins the files correctly, so sox and soxmix are there, and are in the path, and work correctly in theory. Any suggestions would be appreciated! Faris. ___ --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] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, Mark! At 06:33 AM 02/20/2006, you wrote: Please forgive the question, but what is the rationale behind using Solaris over Linux as an asterisk hosting platform? Because of a few reasons, actually: (1) The remote hardware management options available for the X2100 work better (or only, I'm not sure which) under Solaris, and they seem to *really* kick ass. Plus, being Sun-engineered, the X2100 should keep working until it's completely obsolete, and then some. (2) I know someone who knows Solaris inside-out and backwards, blindfolded, while hung upside-down, and codes Bourne shell and C in his sleep; this is vaguely reminiscent of www.chucknorrisfacts.com. I'm quite sure this will come in handy when (not if) something breaks, giving him the opportunity to make some money and giving me the opportunity to reduce my downtime. :) (3) I'd like to learn Solaris, and being SysV-based like Linux, it shouldn't be too much of a stretch. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --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] co-location providers in Ottawa, Canada
Thank you very much.I will contact Sprint, Magma, and Unlimitel about their service.For those who want to participate in Ottawa Asterisk Users Group, please send me email off-list at oss_richard at rogers dot comso I can update you and share ideas on activities.I can ask Carleton University to use one of their facilities on weekends (free parking) for group meetings.richardVirTERM [EMAIL PROTECTED] wrote: You can use Sprint (Group Telecom) and/or Magma. Keep us posted about the group meetings.. Thanks,Wojtek- Original Message - From: Richard OSS To: asterisk-users@lists.digium.com Sent: Sunday, February 19, 2006 12:03 AM Subject: [Asterisk-Users] co-location providers in Ottawa, CanadaAnybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server.< DIV>One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it.Thanks.richard___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] GSM GATEWAY
Dumpolid Exeplish wrote: Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM Any Ideas?? try Quescom , www.quescom.co.za the Q400, they connect to the LAN, then you can dial SIP or H323, using g711, g723, g729. Thet take 12 sim cards eeach ___ --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] GSM GATEWAY
yusuf wrote: Dumpolid Exeplish wrote: Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM Any Ideas?? try Quescom , www.quescom.co.za the Q400, they connect to the LAN, then you can dial SIP or H323, using g711, g723, g729. Thet take 12 sim cards eeach thats so supposed to be www.quescom.com :) ___ --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] Asterisk on Solaris 10 (AMD Opteron, SunFire X2100)
Ah! There you go - I knew Chuck Norris had something to do with it... ;-) Mark -Original Message- From: Alexander Burke [mailto:[EMAIL PROTECTED] Sent: Monday, 20 February 2006 11:17 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, SunFire X2100) Hello, Mark! At 06:33 AM 02/20/2006, you wrote: Please forgive the question, but what is the rationale behind using Solaris over Linux as an asterisk hosting platform? Because of a few reasons, actually: (1) The remote hardware management options available for the X2100 work better (or only, I'm not sure which) under Solaris, and they seem to *really* kick ass. Plus, being Sun-engineered, the X2100 should keep working until it's completely obsolete, and then some. (2) I know someone who knows Solaris inside-out and backwards, blindfolded, while hung upside-down, and codes Bourne shell and C in his sleep; this is vaguely reminiscent of www.chucknorrisfacts.com. I'm quite sure this will come in handy when (not if) something breaks, giving him the opportunity to make some money and giving me the opportunity to reduce my downtime. :) (3) I'd like to learn Solaris, and being SysV-based like Linux, it shouldn't be too much of a stretch. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --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] Loops and Variables
Doug Lytle wrote: trixter aka Bret McDanel wrote: since you have had a little time to play with this, was this the problem? Haven't had a chance yet, will look at it when I get into work this morning. This works correctly now. Doug ___ --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] spa3000
I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret= context=from-pstn auth=md5 [PSTN_2] username=line2 type=peer secret= qualify=yes port=5061 nat=no host=192.168.0.20 context=from-pstn canreinvite=no auth=md5 Try something like this instead: [linea2] username=linea2 type=friend secret= context=from-pstn [line2] username=line2 type=friend secret= qualify=yes nat=no context=from-pstn canreinvite=no Note that I changed this to type=friend, removed the host=, and removed the auth=md5. The above works just fine here with a spa3k. In the [linea2] section, you want the fxs port to be able to place calls as well as receive calls, therefore use type=friend. The same with section [line2]. Note that I also changed the [PSTN_2] to [line2] to match the username= line. Watch the upper/lower case matching to your spa3k configuration. After you get the above working correctly, then you can play around with auth=md5, etc. ___ --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] spa3000
2006/2/20, Rich Adamson [EMAIL PROTECTED]: In the [linea2] section, you want the fxs port to be able to place calls as well as receive calls, therefore use type=friend. The same with section [line2]. Note that I also changed the [PSTN_2] to [line2] to match the username= line. Watch the upper/lower case matching to your spa3k configuration. After you get the above working correctly, then you can play around with auth=md5, etc. Ok thanks, I'll try the authentication now. My other problem now if that spa picks-up the call just after the first ring, even when I specified Off Hook While Calling VoIP: NO. Is this a problem of spa3000 or a problem with asterisk? Who is deciding to answer the call? -- Alejandro Vargas ___ --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] Asterisk on Solaris 10 (AMD Opteron, SunFire X2100)
On Tue, Feb 21, 2006 at 12:17:43AM +1100, Mark Edwards wrote: At 06:33 AM 02/20/2006, you wrote: Please forgive the question, but what is the rationale behind using Solaris over Linux as an asterisk hosting platform? Solaris is also a supported OS (well if you pay for it). It's also 64 bit and any program written for earlier versions will just work. It's 32 bit layer also works out the box (trying to use 32 bit apps on 64 bit Linux can be a PITA). It's also very fast and debugging stuff can be much easier. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Loops and Variables
I usually do the same for IVRs, but I always make sure not to use
itself as the increment, and I use a tempvar instead, like this:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(100);this is where we start the loop
exten = s,100,Set(TCOUNT=${COUNT})
exten = s,101,Noop(${COUNT})
exten = s,102,GotoIf($[${COUNT} 5]?150);exit if more than 5 esle start again
exten = s,103,Set(COUNT=$[{TCOUNT} + 1])
exten = s,104,Goto(100)
exten = s,150,Noop(${COUNT})
I think the above is a bit cleaner, it might be a matter of taste.
On 2/20/06, Doug Lytle [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
trixter aka Bret McDanel wrote:
since you have had a little time to play with this, was this the
problem?
Haven't had a chance yet, will look at it when I get into work this
morning.
This works correctly now.
Doug
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[Asterisk-Users] queue behaviour
Hi folks, need some help on queue behaviour. What Im trying to do is accepting a call from pstn, put it into a queue, while callee is waiting contact some numbers till one responds, then bridge the two calls. What I cant manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I also tried AGI and Asterisk Manager, with the same result. I think Id need some kind of multi-threading. Any ideas? Thanks, _fangi_ ___ --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] SIP ATA gives no ring tone on IAX2 route
Hello everybody, I have this problem where I can't get a ring tone when SIP devices dial an IAX2 route. I get the ring tone using IAX2 devices to dial the same route. The call completes normally in both cases... Facts: - Asterisk 1.0.9 - The Dial command is within an AGI. - ATA (grandstream) and firefly (SIP mode) would not give me the ring tone at all - Switching to a SIP route works ok - Dial with -r option did not do it, same result - I tried progressinband=yes in sip.conf but with no success. So my question is, where should I look for to make sure that my SIP devices will always ring ? Even Asterisk ring tone would do it for now. Any particular settings ? Is sip reload full proof can I use being certin all settings will take effect ? Thanks a lot folks ! Fred ___ --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] Re: Re: RE: virtual extension per user ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You have to use AgentCallbackLogin for that. If a phone logs in that way, it's reachable as Agent/200 You can also use AgentCallbackLogin to logout the agent. You don't have to worry about an agent that forgets to logout on phone X when they walk to phone Y, cause AgentCallbackLoging will overwrite asterisk database entry for that agent so it's only reachable on the phone where they last login (asuming they didn't logout there) This is cool. Another thing, how can I limit outgoing phone calls form IP phone, if no agent isn't logged on that phone? And, in CDR, does it say which agent has made specific phone call? When I get home later today I will put an example in my system and post it here. Now I understand, but (as you can see) now I have new questions :)) Thank you for your time! -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] spa3000
2006/2/20, Rich Adamson [EMAIL PROTECTED]: In the [linea2] section, you want the fxs port to be able to place calls as well as receive calls, therefore use type=friend. The same with section [line2]. Note that I also changed the [PSTN_2] to [line2] to match the username= line. Watch the upper/lower case matching to your spa3k configuration. After you get the above working correctly, then you can play around with auth=md5, etc. Ok thanks, I'll try the authentication now. My other problem now if that spa picks-up the call just after the first ring, even when I specified Off Hook While Calling VoIP: NO. Is this a problem of spa3000 or a problem with asterisk? Who is deciding to answer the call? I'd suggest reading over the info at www.voxilla.com as the interface from the pstn to asterisk is a little different from what one would consider normal. As I recall from various firmware versions on the spa3k, incoming pstn calls are forwarded to asterisk meaning the incoming call is answered and then forwarded. Later versions did something a little different. Look for the spa3k configuration wizard on voxilla as it will assist you with the config process. ___ --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] Strange SIP registration situation
I have 2 Polycom SP 500's attached to my system. Both are behind NATs, but both seem to work fine, for the most part. A few weeks ago, I started to notice that I get an error message from one of them: Feb 20 08:54:58 NOTICE[10663]: chan_sip.c:7691 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for 'zzz.aaa.103.75' However, trying to call the phone works fine, so I know it's registered. Turning on SIP debug for the phone shows that it attempts to register and is rejected with unauthorized, then another attempt is rejected with forbidden, and finally a registration succeeds. The other phone doesn't exhibit this behavior. I am running asterisk 1.0.7. Has anyone used Polycoms remotely from behind a NAT enough to have insight as to what is going on? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] Handset phone to replace Flash Operator Pane l
I have that set up, but I cannot get some of the phones to change the hint State. The SNOM phone show State:InUse, but Swissvoice phones show State:Idle even when on a call. I use 'show hints' to see this. Kind Regards Garth Colin Anderson wrote: Breeze to set up, too. To monitor and transfer to SIP/1000 / ext 1000: 1. Insert exten = 1000,hint,SIP/1000 into your default context (the context the extension is in) 2. In the monitoring phone's web interface, click Function Keys, pick a key, change it to Destination and type in SIP/1000. Once you submit the form it will change to a SIP URL, that's OK. 3. There is no step 3. Only drag is you can't daisy-chain expansion modules, although there is a daisy-chain port on the module. So 54 keys max. -Original Message- *From:* Rob Lith [mailto:[EMAIL PROTECTED] *Sent:* Wednesday, February 08, 2006 10:56 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] Handset phone to replace Flash Operator Panel Garth The SNOM 360 with extension panel is one of the best options, it handles all the extension indication status and has enough line extensions to cover up to 54 extensions. Only the Polycom 601 comes close. Regards Rob On 2/8/06, *Garth van Sittert* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi All Has anyone come across a handset that can somehow replace FOP? Some users don't like FOP unless it is on a dedicated PC. Thanks Garth ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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
Re: [Asterisk-Users] spa3000
I don't mess with configuring these, the wizard on voxilla.com does everything except set the right context. Try using default for everything to get it working then separate as needed. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spa3000
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Tuesday, 21 February 2006 1:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spa3000 I don't mess with configuring these, the wizard on voxilla.com does everything except set the right context. Try using default for everything to get it working then separate as needed. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: spa.jpg ___ --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] Where can I get the tar.gz sources of libnewt?
I think this would help you. http://packages.debian.org/unstable/perl/libnewt-perl Anthony Azzopardi wrote: Where can I get the tar.gz sources of libnewt? Reg, Anthony Azzopardi. ___ --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] chan_capi setting ${DNIS}
On 20/02/2006, at 12:08 PM, Andrew Furey wrote:
On 2/20/06, Nathan Alberti [EMAIL PROTECTED] wrote:
Is there a reason the variable ${DNIS} does not get set with incoming
calls via chan_capi ?
Is it related to the MSN=X in capi.conf ?
Just a guess, are you thinking of ${DNID} instead? There's no direct
mention of ${DNIS} on the wiki variables page, but ${DNID} works for
me with a BRI...
Andrew
Thanks Andrew, DNID was what I was meant.
Nathan.
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Re: [Asterisk-Users] Application Faxing using SIP
Phil Blundell wrote: I suspect that the datapath through our regular network switches is probably close enough to lossless for this purpose as well. You could be surprised. I know that I have been suprised by how easy it is for a UDP packet to get dropped or lost. I've had a few customers approach me that had both an Asterisk server and a HylaFAX server running on separate networked systems and wanted to use iaxmodem (running on the HylaFAX box, channels from the Asterisk box)... and in tests after configuration it has never worked well enough for me to sign-off on it, audio packets were getting dropped, and we've always gone the route of using termnetd+ttyd to use the modems remotely... so iaxmodem would run on the Asterisk server and HylaFAX would use a ttyd-created remote modem device. (I haven't completely enjoyed this, either, but it's more acceptable to me than the arrangement of dropping UDP audio packets.) In fact, the only environment where I have seen a suitable arrangement where iaxmodem communicates with an Asterisk server that is not running on the same host is at my own home-office... and the traffic on my home network is certainly more than in some of my customers'... so I'm not so sure that it always has to do with traffic volume... and I'm more inclined to think that it either has to do with the hardware involved (the ethernet switches, for example) or it has to do with other specifics of the network configuration. Losing an audio packet here or there wouldn't normally be so bad for fax. Normally I would expect the fax protocol, especially ECM protocol, to be able to recover from it. However, Asterisk seems to not work in an ideal fashion for this purpose. Whenever Asterisk encounters a lost audio packet something called packet loss concealment is performed by placing a PLC frame there as a placeholder. When the audio is retransmitted the PLC frame is supposed to be converted into synthesized audio. Between what I have been told and from what I have observed, this conversion of PLC frames into synthesized audio does not happen with uLaw, alaw, or slinear codecs (the only codecs suitable for fax). Consequently the PLC frame is converted into zero-data... or 20 ms of silence... which is probably the worst-possible thing that could happen. A 20 ms period of silence will make the modem detect carrier loss. In fax protocol carrier loss is used to synchronize the communications... when carrier loss is detected the fax device knows that it's time to move on to the next step in the protocol. So, depending at the timing of the packet loss things can go awry enough to cause the fax session to fail outright. So... because a mere 20 ms gap in audio can cause so much trouble for faxing, it's very important to make sure that the lossless communication medium between Asterisk and the fax device is truly lossless. Lee. ___ --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] q931 85
Anyone know if asterisk supports q931 85 in the uk? Thanks Bails ___ --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] A good SIP VB6.0 component to use?
Hi there We are wanting to build our own SIP soft phone using VB6. What is a good component to use for this? We have done research and have only found very expensive ones offered by VaxVoip or Radvision. Anyone know of a good component that does the basics that doesn't cost two arms and both legs? We have used Microsoft RTC in the past but that didn't work for us since it uses silence suppression by default which Asterisk doesn't support. Cheers Hagen ___ --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] q931 85
On Mon, Feb 20, 2006 at 03:15:52PM +, bails wrote: Anyone know if asterisk supports q931 85 in the uk? Nope, it only supports Q.931 110 (which is EuroISDN). 85 is UK ISDN, most providers can set the line to 110, but you may have to ask for it. Marconi System X switches (as used by BT, THUS and others) will default to 85. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Adam Robins wrote: Hi Adam After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls breaking up from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. I am interested to know why you are using ilbc, n why not g729 ot g723 or speex. What is the size of the WAN connection. How many calls are you running over this link. I just need to see how others are fairing with IAX2 over WAN links, as I am the final stages of testing on my side thanks, yusuf ___ --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] Grandstream GXP-2000
Page 3 of the same data sheet reads: Optional 5 volt DC Universal (100-240 Volt) Switching Power Adaptor And, Package Contents section on the same page reads: Important Note: Power Supply is Ordered Separately -- Models: PA100-NA, PA100-EU, PA100-UK, PA100-AU This explains the PoE issue, I think. For 100bit issue, I tend to believe in the data sheet, but I would also like to hear a first-hand verification. (But I guess we have to wait, because voipsupply accepts pre-sale orders for now, they don't ship them yet.) - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 1:01 PM Subject: Re: [Asterisk-Users] Grandstream GXP-2000 On Mon, 20 Feb 2006, Soner Tari wrote: I have Data Sheet for 942 from Linksys web site. It says this on page 4 (close to bottom): Physical Interfaces: 2 100baseT RJ-45 Ethernet Ports (IEEE 802.3) And that was one of the reasons I was considering 942. Do you think the data sheet may be wrong? every reseller that is selling the 942 lists two 10mb ports. also, rather disturbingly the linksys press release[1] implies the 942 is PoE only (like the aastra 480i), no external power supply. if anyone has a 942 and can authoritatively state it has 100meg ports and supports non-PoE power source, i'd definitely like to know. -Dan [1] http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayoutpackedargs=c%3DL_News_C2%26cid%3D1136499819516pagename=Linksys%2FCommon%2FVisitorWrapper ___ --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] calling from SIP to a h.323 device with oh323
Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday). = Connected to Asterisk SVN-branch-1.2-r10487 currently running on sip (pid = 29977) nip*CLI Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Hold and Call Waiting - Budgetone 100
Hello, I didnt exactly find what the problem was but I built a new Asterisk server, copied the conf files over from the original server and now the phones work fine. Thanks, Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Peters Sent: Friday, February 17, 2006 5:00 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hold and Call Waiting - Budgetone 100 Hello, I have been working on getting Asterisk up and running with a couple Grandstream hard phones (Budgetone 100s) and a couple softphones. Things work pretty well but we have discovered an odd problem. A calls B and then A can put the call on hold and pick it back up. A calls B and B can put the call on hold but cannot pick it back up. These tests are internal on the hard phones. The same also happens to calls over the PSTN. Also, another issue that is probably related is with call waiting. A calls C and then B calls C. C can flash over to pick up Bs call and A goes on hold. C cannot flash back over to pickup As call. B does get put on hold but the call to A never comes off hold. These tests are internal on the hard and soft phones. Thank you, Dan Peters ___ --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] Landmark digital key systems and Asterisk
Anyone integrated Landmark (formerly Southwestern Bell) Digital Key System phones into an Asterisk installation? The phones are model DKS930 and the main CPU for the system is a DKS1224. I'm hoping to reuse some of the phones with a new Asterisk install I'm building. Thanks! John Cornell's True Value www.cornells.com ___ --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] Waiting for your help...
You have to make all of your manual changes in the _custom.conf files. [EMAIL PROTECTED] overwrites the xxx.conf files -I think this happens every time you restart the app. Log files are usually in /var/log/asterisk and you can see them in the maintenance screen on AMP From: yrving rivas [mailto:[EMAIL PROTECTED] Sent: Monday, February 13, 2006 8:37 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Waiting for your help... Thanks Tzafrir: I am apologize because of my language problems writing the subject. I am not good at english. So thanks for telling me I am doing it the wrong way, and I will be more carefully next time. Help me if it is possible to you. The Asterisk version is 2.1 wich I downloaded trhough http://asteriskathome.sourceforge.net/. To install de fax to email support I followed the instructions in http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+8 How do I trace and how do I post my configuration files?. I have some about asterisk programming, I know some in programming, and a good learner Plz. YrvingTzafrir Cohen [EMAIL PROTECTED] escribió: On Mon, Feb 13, 2006 at 05:25:37AM -0600, yrving rivas wrote: Hello every one. This is a question done by me, not yet answered. Please, help.How about a decent subject for your message? I: 1. Run install-pdf from linux to support faxes on my asterisk.Of what software package, exactly?What version?What version of Asterisk?What OS/distro? What version of it? 2. Made the configurations throuhg AMP in a. Setup-Inbound Routing-(the only route I have)-fax extension-System b. Setup-Inbound Routing-(the only route I have)-fax email-(my email) c. Setup-Inbound Routing-(the only route I have)-Immediate Answer- yes d. Setup-Inbound Routing-(the only route I have)-pause after answer- 2 e. Setup-General Settings-fax machine for receiving faxes-system f. Setup-General Settings-Email address to have-(my email) 3. as a good boy made a test call from a fax, and it reports that couldn´t send the fax ( what means the aste risk couldn´t receive it). I didn´t receive any fax. What can I do to receive them? Tips: 1- In my configuration I have a TDM04B. 2- I receive via email the voice mail messages left to any extension.Looks like a CLI trace would come in handy. In other hand (and not related to this case, as you will see):No, I'm not sure. AMP's dialplan is a mess, and there's no telling whata naive change to it will do. I made changes to the extensions.conf file through AMP to construct a call forward on no answer, but at the next day all programming was like at beggining. What should I do to make the changes for ever? amp normally does not override extensions.conf (except, maybe on upgradetime).Anyway, posting your modified extentions.conf may help. Yourextensions_additional.conf may help as well. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | bestICQ# 16849755 | | friend___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] calling from SIP to a h.323 device with oh323
Hi, Can you post your working config, I'm wasting my time to config h323-sip Thanks Guillermo Salas M wrote: Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday). = Connected to Asterisk SVN-branch-1.2-r10487 currently running on sip (pid = 29977) nip*CLI Best regards, -- Marc Patino Gómez Dpto. Sistemas Claranet España. Servicios Internet C/General Almirante 2-28, Torres Cerdá 08014 Barcelona Tel: +34 93 445 26 50 Fax: +34 93 445 19 20 www.claranet.es Claranet Group: United Kingdom - Spain - France - Germany - Portugal - Netherlands - USA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loops and Variables
On Mon, Feb 20, 2006 at 08:56:41AM -0500, C F wrote:
I usually do the same for IVRs, but I always make sure not to use
itself as the increment, and I use a tempvar instead, like this:
Why?
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(100);this is where we start the loop
exten = s,100,Set(TCOUNT=${COUNT})
exten = s,101,Noop(${COUNT})
exten = s,102,GotoIf($[${COUNT} 5]?150);exit if more than 5 esle start
again
exten = s,103,Set(COUNT=$[{TCOUNT} + 1])
exten = s,104,Goto(100)
exten = s,150,Noop(${COUNT})
Why not simply:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(100);this is where we start the loop
exten = s,100,GotoIf($[${COUNT} 5]?150);exit if more than 5 esle start
again
exten = s,101,Set(COUNT=$[${COUNT} + 1])
exten = s,102,Goto(100)
exten = s,150,Noop(${COUNT})
Alternatively:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5 esle start
again
exten = loop,2,Set(COUNT=$[${COUNT} + 1])
exten = loop,3,Goto(1)
exten = next,1,Noop(${COUNT})
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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[Asterisk-Users] asterisk error
Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:+ 1^2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. Any ideas? a Thank you Dov ___ --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] co-location providers in Ottawa, Canada
Telecom Ottawa? Large, Ultra fast pipe with direct connections to TDM providers (Which may be at 151 Front St. in Toronto) but they should work for what you want. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard OSSSent: February 19, 2006 12:04 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] co-location providers in Ottawa, Canada Anybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server. One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it. Thanks. richard ___ --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] Loops and Variables
It was trying to perform looping in the dialplan that made me seriously look at
AGI. Gee, I wonder what's easier.
This:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5 esle start
again
exten = loop,2,Set(COUNT=$[${COUNT} + 1])
exten = loop,3,Goto(1)
exten = next,1,Noop(${COUNT})
or this...
loop = 0
while loop 5:
do-something
loop += 1
I really wasn't enthused about having to look at dialplan code months later and
try and work out what I did earlier.
Nasty!
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Monday, February 20, 2006 9:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Loops and Variables
On Mon, Feb 20, 2006 at 08:56:41AM -0500, C F wrote:
I usually do the same for IVRs, but I always make sure not to use
itself as the increment, and I use a tempvar instead, like this:
Why?
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(100);this is where we start the loop
exten = s,100,Set(TCOUNT=${COUNT})
exten = s,101,Noop(${COUNT})
exten = s,102,GotoIf($[${COUNT} 5]?150);exit if more than 5 esle start
again
exten = s,103,Set(COUNT=$[{TCOUNT} + 1])
exten = s,104,Goto(100)
exten = s,150,Noop(${COUNT})
Why not simply:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(100);this is where we start the loop
exten = s,100,GotoIf($[${COUNT} 5]?150);exit if more than 5 esle start
again
exten = s,101,Set(COUNT=$[${COUNT} + 1])
exten = s,102,Goto(100)
exten = s,150,Noop(${COUNT})
Alternatively:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5 esle start
again
exten = loop,2,Set(COUNT=$[${COUNT} + 1])
exten = loop,3,Goto(1)
exten = next,1,Noop(${COUNT})
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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RE: [Asterisk-Users] GSM GATEWAY
Why not get an asterisk and install software like a2billing on it. It has CDR and things like that From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid Exeplish Sent: Monday, February 20, 2006 7:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GSM GATEWAY well, does this gateway support SIP?? and does it generate its own CDR? could you send the devices brocure/tech spec.?? thanks On 2/19/06, Sam Tam [EMAIL PROTECTED] wrote: Why not get 30 GSM Gateway from us at £60 each and then get an asterisk or some voip gateway like A800 and then link it all up From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Dumpolid Exeplish Sent: Sunday, February 19, 2006 10:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] GSM GATEWAY Hi everyone, Can anyone give me suggestions on any equipment that can connect from VOIP to a GSMgateway(channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I am looking of a device that is purely VOIP to GSM Any Ideas?? ___ --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] asterisk error
Dov Bigio wrote: Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: What part of your dial plan is generating the error? Can you post it? Doug ___ --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] Loops and Variables
Douglas Garstang wrote:
It was trying to perform looping in the dialplan that made me seriously look at
AGI. Gee, I wonder what's easier.
This:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5 esle start
again
exten = loop,2,Set(COUNT=$[${COUNT} + 1])
exten = loop,3,Goto(1)
exten = next,1,Noop(${COUNT})
or this...
loop = 0
while loop 5:
do-something
loop += 1
I really wasn't enthused about having to look at dialplan code months later and try and work out what I did earlier.
Nasty!
If you think that's nasty.
[default]
exten = 3528,1,SetVar([EMAIL PROTECTED])
exten = 3528,2,SetVar(DIAL_DEST[1]=SIP/0004f201fbd8-a)
exten = 3528,3,SetVar(DIAL_DEST[2]=SIP/0004f201fbd8-b)
exten = 3528,4,Macro(std-exten)
[macro-std-exten]
exten =
s,1,AGI(callerid-fixup.agi,${CALLERIDNUM}${MACRO_EXTEN}00)
exten = s,2,Noop(AGI(set-ring))
exten = s,3,GotoIf($[${LEN(${FAX_DEST})} = 0]?9:4)
exten = s,4,Cut(TECHNOLOGY=CHANNEL,/,1)
exten = s,5,GotoIf($[${TECHNOLOGY} = Zap]?6:9)
exten = s,6,Answer
exten = s,7,Ringing
exten = s,8,NVFaxDetect(4,d)
exten = s,9,Goto(${MACRO_EXTEN},1)
exten = _,1,GotoIf($[${LEN(${DIAL_DEST[1]})} = 0]?2:4)
exten = _,2,GotoIf($[${LEN(${DIAL_DEST})} = 0]?14:3)
exten = _,3,SetVar(DIAL_DEST[1]=${DIAL_DEST})
exten = _,4,SetVar(INDEX=1)
exten = _,5,GotoIf($[${LEN(${DIAL_TIMEOUT[${INDEX}]})} = 0]?6:7)
exten = _,6,SetVar(DIAL_TIMEOUT[${INDEX}]=20)
exten =
_,7,Dial(${DIAL_DEST[${INDEX}]},${DIAL_TIMEOUT[${INDEX}]},${DIAL_OPTS[${INDEX}]}g)
exten = _,8,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?12:9)
exten = _,9,GotoIf($[${DIALSTATUS} = NOANSWER]?14:10)
exten = _,10,Noop(DIALSTATUS=${DIALSTATUS})
exten = _,11,Hangup
exten = _,12,SetVar(INDEX=$[${INDEX} + 1])
exten = _,13,GotoIf($[${LEN(${DIAL_DEST[${INDEX}]})} = 0]?14:5)
exten = _,14,GotoIf($[${LEN(${VOICE_MAILBOX})} = 0]?19:15)
exten = _,15,Voicemail(${VOICE_MAILBOX})
exten = _,16,Wait(2)
exten = _,17,Hangup
exten = _,18,GotoIf($[${DIALSTATUS} = NOANSWER]?19:22)
exten = _,19,Voicemail(u${EXTEN})
exten = _,20,Wait(2)
exten = _,21,Hangup
exten = _,22,Voicemail(b${EXTEN})
exten = _,23,Wait(2)
exten = _,24,Hangup
exten = _,116,AbsoluteTimeout(30)
exten = _,117,Playback(sorry-mailbox-full)
exten = _,118,Wait(2)
exten = _,119,Congestion
exten = _,120,Goto(116)
exten = _,123,Goto(116)
exten = talk,1,Goto(${MACRO_EXTEN},1)
exten = fax,1,Cut(FAX_TECH=FAX_DEST,/,1)
exten = fax,2,GotoIf($[${FAX_TECH} = Zap]?3:7)
exten = fax,3,Dial(${FAX_DEST},20)
exten = fax,4,AbsoluteTimeout(30)
exten = fax,5,Wait(2)
exten = fax,6,Congestion
exten = fax,7,RxFax(/tmp/fax-${UNIQUEID}.tiff)
exten =
fax,8,DeadAGI(/usr/local/bin/fax2email.pl,/tmp/fax-${UNIQUEID}.tiff)
exten = fax,9,Hangup
exten = fax,104,AbsoluteTimeout(30)
exten = fax,105,Busy
exten = a,1,Playback(/etc/asterisk/directvm)
exten = a,2,VoicemailMain()
exten = a,3,Wait(.5)
exten = a,4,Goto(1)
exten = o,1,GotoIf($[${LEN(${OPER_DEST})} = 0]?2:4)
exten = o,2,Goto(extensions,0,1)
exten = o,3,Hangup
exten = o,4,GotoIf($[${OPER_TIMEOUT} = 0]?5:6)
exten = o,5,SetVar(OPER_TIMEOUT=)
exten = o,6,GotoIf($[${LEN(${OPER_MESSAGE})} = 0]?8:7)
exten = o,7,Playback(${OPER_MESSAGE})
exten = o,8,Dial(${OPER_DEST},${OPER_TIMEOUT},${OPER_FLAGS})
exten = o,9,Voicemail(u${MACRO_EXTEN})
exten = o,10,Hangup
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RE: [Asterisk-Users] asterisk error
Dov Bigio wrote: Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: What part of your dial plan is generating the error? Can you post it? Doug This sounds a lot like the error Doug was getting when he tried to increment a variable before it actually was defined. Please post the part of the dialplan that causes this error and we will probably be able to figure it out. -MC ___ --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] Good VoIP providers that support Asterisk PBX's
Hi, Why not try http://www.voipjet.com Been with them and found them quite good... Try it out Dan# On 20/02/06, Joseph Tanner [EMAIL PROTECTED] wrote: I have voicepulse connect too. I had occassional problems with incoming calls, but not many and not recently. Have had more problems with outgoing calls which is fine for me, as I have more than one backup (I use voxee as my primary due to lowest price, then voicepulse, and failing that I can use my cellphone or my landline). I am a bit disappointed with the price, it was decent before they upped it to $11. Seems a bit high to me, for just an incoming line with no outgoing minutes. Many other places charge about that and give you a bunch of minutes, or an unlimited local calling plan (in-state, in-area code, etc.). But, it's been very reliable, no complaints about uptime. Joseph Tanner On 2/19/06, David Blomquist [EMAIL PROTECTED] wrote: I've been using voicepulce connect for several months with very few problems. Occasionally I get all circuits are busy messages when trying to dial out but no too often. d From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Sunday, February 19, 2006 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's I had voicepulse connect but had to transfer IAX2 had non stop drop outs in audio all the time. Tried everything to fix it, even with 14ms ping times it just didnt want to work right. I never figured out why, just canceled. Although i didnt like the no-name on incoming caller id either though, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthews Sent: Tuesday, February 14, 2006 8:52 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's http://connect.voicepulse.net They support astrisk, with iax2 :) On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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
Re: [Asterisk-Users] indications issues in Singapore?
Good to know about that Loopstart thing --- helped me quickly solve my problem of the phones not ringing :-) thank you for the input Chris - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 17, 2006 7:10 PM Subject: Re: [Asterisk-Users] indications issues in Singapore? Chris Earle (CBL) wrote: Hi all, haven't seen many posts about asterisk in Singapore... Getting a server going there and was wondering if TDM400Ps will be fine in FCC mode, and if there are indications / cadence values that I should be putting on there as in other international locations. Seen an unsettling post on voip-info about Singapore issues with Call Polarity/Hangup Detection -- crossing my fingers I don't run into that problem :-) Analog lines here are mostly loopstart, so you need to enable busydetect if you're using the zaptel FXO. A better option is to use a capi ISDN BRI card. I used the Fritz! PCI card with chan_capi, costs around S$160. The original poster on voip-info wrote about using kewlstart and CPC, which I have never encountered over here. I guess it was in vogue during the good old DID analog trunk days. But nowadays, you either use plain analog or move to BRI/PRI if you need MSN/DDI. ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Zap channels Deactivated with Bristuff-0.3.x after upgrade from 0.2.0
Hello, I have about 10 Asterisk PBX in production with Bristuff-0.2.0-RC8q (asterisk 1.0.10) and I want to use Bristuff-0.3 now for the new PBX I am going to set up. With Bristuff-0.2.0-RC8q the ISDN lines are working fine, but the new version of Asterisk add some nice features. All these PBX are in France with France Telecom lines. When I use the new version after about an hour with Euronumeris lines and almost instantly with Euronuméris+ line, TE lines goes Deactivated in /proc/zaptel/: Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED (F7) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) I have tested almost all the possible settings for /etc/zaptel.conf and /etc/asterisk/zapata.conf There is no IRQ sharing for the HFC-PCI cards. When cards are deactivated, I can receive calls, but not make call. The cards in NT mode seems to work fine (connected to an Alcatel PBX). All the cards are 1 port HFC-PCI cards. I also have an 8 ports card from Junghanns.net but this card is on a production server and it's hard to use it for test. Almost all the PBX are on Centos 3.4 with kernel 2.4.21. Does someone have this problem to? I can post the configuration files if it can help. Best regards ___ --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] problem with outgoingcallsUnabletocreatechannelof type 'ZAP' (cause 34 -Circuit/channelcongestion)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600 Sent: Saturday, February 18, 2006 2:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoingcallsUnabletocreatechannelof type 'ZAP' (cause 34 -Circuit/channelcongestion) On 2/17/06, Michael Collins [EMAIL PROTECTED] wrote: Nik, This definitely helps! Please check your dial command. You've got Dial(Zap/0/mynumber) and I think you might possibly want it to be something like this: Dial(Zap/1/mynumber) or Dial(Zap/g0/mynumber) I don't recall there being a zap channel zero, but it is common to have a group zero. I would recommend trying Zap channel 1 - Dial(Zap/1/mynumber) - before trying the group. Again, please get the debug info. The CHANUNAVAIL message made it easier to diagnose this issue. Don't give up! The education you are getting will help you in the long run and in a few months you'll be able to help a * newbie with the same issues! -MC ok, thanks for your help, please, be patient because now i've got many logs to post ... :-) so, i've made this new entry in extension.conf: exten = 444,1,Dial(Zap/0/0465670127) exten = 445,1,Dial(Zap/g0/0465670127) exten = 446,1,Dial(Zap/1/0465670127) exten = 447,1,Dial(Zap/g1/0465670127) and i've reloaded asterisk with: asterisk -r reload quit and then: tail -f /var/log/asterisk/full I CALL 445 or 446: snip Feb 18 04:53:20 VERBOSE[3608] logger.c: Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network s erving the local user (2) Feb 18 04:53:20 VERBOSE[3608] logger.c: Ext: 1 Cause: Unknown (99), class = Protocol Error (6) ] /snip now i'll read and search on the network about all this warnings and error and verbose report, but i hope that your experience will found where is the problem...thanks Nik, It looks like your exten 445 and exten 446 are communicating on the D-channel! Again, that's progress. It looks like there is a protocol error (see snipped portion of log) when dialing out. I can't quite figure out what the source of that protocol error is. However, at this point it would be good to review what ISDN settings your telephone carrier is providing. Specifically, can they tell you which protocol (sometimes we yanks will call it a protocol variant) they are using? If they are set for NI2 or National then perhaps there's something going on within their equipment. Hard to say without doing some testing. I've had some strange occurrences with my telephone carriers here in California. For example, I had the telco set their equipment to 4ess and I set mine to the same - it would NOT work, no matter what. I had them leave theirs on 4ess and I set mine to 5ess and it worked perfectly! Go figure. Are you in a position to have one of the carrier's engineers do some debugging? You can call them and let them know that you are making test calls but your equipment is showing protocol errors. They should be able to do a trace on the D-channel on their end. Hopefully you'll get an engineer who knows Q.931. (I've had technicians who couldn't even spell P-R-I and I've had to escalate the phone call to their respective supervisors!) There are several debugging options right now, but I wouldn't continue without getting your carrier involved. They may look at the D-Channel messages and make an adjustment on their end, or they might suggest changing protocols, at least for testing. Don't worry - if this is your first foray into the wacky world of PRI then you're just getting the obligatory baptism by fire. I've set up dozens of PRI's here in the states and at first it always took hours, even for an experienced PBX technician. But now that I've been through the wringer I know which questions to ask and what tinkering to do. Please don't give up - PRI is pretty nice once you getting it working. If you have any questions about talking to your carrier, please contact me offline. I'll be happy to help! -MC ___ --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] Asterisk start errors with TDM2413E
I do not think so by reading the documentation
however I have changed the settings and still get the same error when starting
Asterisk
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
- Message from [EMAIL PROTECTED] on Mon, 20 Feb 2006 07:08:05
+0100 -
To:
'Asterisk Users Mailing List - Non-Commercial
Discussion' asterisk-users@lists.digium.com
Subject:
RE : [Asterisk-Users] Asterisk start errors with TDM2413E
Hi,
I believe that you have inverted fxo_ks and
fxs_ks into your zapata.cong file signaling= declaration...
Invert and redo the tests.
Good Luck !
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
De la part de [EMAIL PROTECTED]
Envoyé : lundi 20 février 2006 04:34
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Asterisk start errors with TDM2413E
I get the following errors when starting Asterisk.
== Parsing '/etc/asterisk/zapata.conf': Found
Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open:
Unable to specify channel 1: No such device
Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable
to open channel 1: No such device
here = 0, tmp-channel = 1, channel = 1
Feb 19 21:14:35 ERROR[10440]: chan_zap.c:10264 setup_zap:
Unable to register channel '1'
Feb 19 21:14:35 WARNING[10440]: loader.c:414 __load_resource:
chan_zap.so: load_module failed, returning -1
Feb 19 21:14:35 WARNING[10440]: loader.c:554 load_modules:
Loading module chan_zap.so failed!
[EMAIL PROTECTED] ~]# Ouch ... error while writing audio
data: : Broken pipe
Software versions
asterisk-1.2.3
asterisk-addons-1.2.1
asterisk-perl-0.08
asterisk-sounds-1.2.1
libpri-1.2.2
zaptel-1.2.4
Output from modprobes
[EMAIL PROTECTED] asterisk]# modprobe -v zaptel
insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/zaptel.ko
[EMAIL PROTECTED] asterisk]# modprobe -v wctdm24xxp
install /sbin/modprobe --ignore-install wctdm24xxp
/sbin/ztcfg
insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/wctdm24xxp.ko
This takes at least 10 seconds to come back to
a prompt
ztcfg output
[EMAIL PROTECTED] asterisk]# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Kewlstart (Default) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Slaves: 08)
Channel 09: FXS Kewlstart (Default) (Slaves: 09)
Channel 10: FXS Kewlstart (Default) (Slaves: 10)
Channel 11: FXS Kewlstart (Default) (Slaves: 11)
Channel 12: FXS Kewlstart (Default) (Slaves: 12)
Channel 13: FXS Kewlstart (Default) (Slaves: 13)
Channel 14: FXS Kewlstart (Default) (Slaves: 14)
Channel 15: FXS Kewlstart (Default) (Slaves: 15)
Channel 16: FXS Kewlstart (Default) (Slaves: 16)
16 channels configured.
zaptel.conf
fxoks=1-4
fxsks=5-16
defaultzone=us
loadzone=us
zapata.conf
[channels]
signalling=fxo_ks
echocancel=yes
echocancelwhenbridged=yes
usecallerid=yes
context=outstation
channel= 1-4
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
usecallerid=yes
group=2
context=incomingpstn
channel= 5-16
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
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Re: [Asterisk-Users] Grandstream GXP-2000
One thing to keep in mind with PoE is that you can simply use an injector at the phone location. At least with the 480i you can easily order the phone with the power injector. Richard On 2/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Mon, 20 Feb 2006, Soner Tari wrote: I have Data Sheet for 942 from Linksys web site. It says this on page 4 (close to bottom): Physical Interfaces: 2 100baseT RJ-45 Ethernet Ports (IEEE 802.3) And that was one of the reasons I was considering 942. Do you think the data sheet may be wrong? every reseller that is selling the 942 lists two 10mb ports.also, rather disturbingly the linksys press release[1] implies the 942 isPoE only (like the aastra 480i), no external power supply.if anyone has a 942 and can authoritatively state it has 100meg ports and supports non-PoE power source, i'd definitely like to know.-Dan[1] http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayoutpackedargs=c%3DL_News_C2%26cid%3D1136499819516pagename=Linksys%2FCommon%2FVisitorWrapper___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Monday, February 20, 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Adam Robins wrote: Hi Adam After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls breaking up from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. I am interested to know why you are using ilbc, n why not g729 ot g723 or speex. What is the size of the WAN connection. How many calls are you running over this link. I just need to see how others are fairing with IAX2 over WAN links, as I am the final stages of testing on my side thanks, yusuf ___ --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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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] any doc/example for app_sms.so ?
Hi! is there any documentation or simple example around for app_sms.so to get started with it and do two simple tasks: 1. send a message to an sms-capable phone connected to an ATA 2. receive a message from an sms-capable phone and so something simple with it, even just write it to the debug screen... This has quite a lot of info, might be able to help you: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+sms Cheers, Philipp ___ --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] Grandstream GXP-2000
At 03:01 AM 02/20/2006, you wrote: also, rather disturbingly the linksys press release[1] implies the 942 is PoE only (like the aastra 480i), no external power supply. Well, the 480i CT comes with a wall wart if you don't want to use POE and their web site shows the optional PS available. I assume the 480i is similar but I don't have one so I can't say. Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.375 / Virus Database: 267.15.11/264 - Release Date: 02/17/2006 ___ --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] Asterisk start errors with TDM2413E
Issue resolved, Thanks Digium!
The module slot closest to the bracket
that contains the connector is the last slot. I assumed that the
first slot would be at the bracket.
Silly user error, never assume.
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
o. 602.532.3706
c. 602.692.0304
__
I do not think so by reading the documentation
however I have changed the settings and still get the same error when starting
Asterisk
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
- Message from [EMAIL PROTECTED] on Mon, 20 Feb 2006 07:08:05
+0100 -
To:
'Asterisk Users Mailing List - Non-Commercial
Discussion' asterisk-users@lists.digium.com
Subject:
RE : [Asterisk-Users] Asterisk start errors with TDM2413E
Hi,
I believe that you have inverted fxo_ks and
fxs_ks into your zapata.cong file signaling= declaration...
Invert and redo the tests.
Good Luck !
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
De la part de [EMAIL PROTECTED]
Envoyé : lundi 20 février 2006 04:34
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Asterisk start errors with TDM2413E
I get the following errors when starting Asterisk.
== Parsing '/etc/asterisk/zapata.conf': Found
Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open:
Unable to specify channel 1: No such device
Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable
to open channel 1: No such device
here = 0, tmp-channel = 1, channel = 1
Feb 19 21:14:35 ERROR[10440]: chan_zap.c:10264 setup_zap:
Unable to register channel '1'
Feb 19 21:14:35 WARNING[10440]: loader.c:414 __load_resource:
chan_zap.so: load_module failed, returning -1
Feb 19 21:14:35 WARNING[10440]: loader.c:554 load_modules:
Loading module chan_zap.so failed!
[EMAIL PROTECTED] ~]# Ouch ... error while writing audio
data: : Broken pipe
Software versions
asterisk-1.2.3
asterisk-addons-1.2.1
asterisk-perl-0.08
asterisk-sounds-1.2.1
libpri-1.2.2
zaptel-1.2.4
Output from modprobes
[EMAIL PROTECTED] asterisk]# modprobe -v zaptel
insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/zaptel.ko
[EMAIL PROTECTED] asterisk]# modprobe -v wctdm24xxp
install /sbin/modprobe --ignore-install wctdm24xxp
/sbin/ztcfg
insmod /lib/modules/2.6.14-1.1656_FC4smp/misc/wctdm24xxp.ko
This takes at least 10 seconds to come back to
a prompt
ztcfg output
[EMAIL PROTECTED] asterisk]# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Kewlstart (Default) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Slaves: 08)
Channel 09: FXS Kewlstart (Default) (Slaves: 09)
Channel 10: FXS Kewlstart (Default) (Slaves: 10)
Channel 11: FXS Kewlstart (Default) (Slaves: 11)
Channel 12: FXS Kewlstart (Default) (Slaves: 12)
Channel 13: FXS Kewlstart (Default) (Slaves: 13)
Channel 14: FXS Kewlstart (Default) (Slaves: 14)
Channel 15: FXS Kewlstart (Default) (Slaves: 15)
Channel 16: FXS Kewlstart (Default) (Slaves: 16)
16 channels configured.
zaptel.conf
fxoks=1-4
fxsks=5-16
defaultzone=us
loadzone=us
zapata.conf
[channels]
signalling=fxo_ks
echocancel=yes
echocancelwhenbridged=yes
usecallerid=yes
context=outstation
channel= 1-4
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
usecallerid=yes
group=2
context=incomingpstn
channel= 5-16
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
___
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RE: [Asterisk-Users] queue behaviour
What I'm trying to do is accepting a call from pstn, put it into a queue, while callee is waiting contact some numbers till one responds, then bridge the two calls. What I can't manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I've no idea if this'll work in practice, but the theory seems sound: 1) Create some extensions in your dialplan which dial the numbers you want the queue to try: exten = 1000,1,Dial(dialstring here) exten = 1001,1,Dial(second dialstring here) etc. 2) Assign members to your queue as follows: member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] etc. 3) Set the queue to ringall or round robin as required. 4) let the list know whether it worked or not :-) Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] Grandstream GXP-2000
On Mon, 20 Feb 2006, Richard Amerman wrote: One thing to keep in mind with PoE is that you can simply use an injector at the phone location. At least with the 480i you can easily order the phone with the power injector. Aastra does not really make it clear that the 480i is poe _only_. A lot of people are very suprised when I explain to them that the 480i is poe only. -Dan ___ --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] spa3000
As I recall from various firmware versions on the spa3k, incoming pstn calls are forwarded to asterisk meaning the incoming call is answered and then forwarded. Later versions did something a little different. I can definitely confirm that the SPA3000 here at home forwards the call to asterisk *without* answering the line, if that's any help. It's running the latest firmware. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] g729 quality at GSM bitrates
Greetings all, I'm trying to improve the codec selection on a few of the asterisk boxes we have to keep the g729 licences free for calls from ATAs that don't support anything apart from g711 and g729. GSM seems to offer noticably inferior call quality (at least when using a softphone + decent headphones), but it's about where I want the bitrate to be. I know there are lots of Speex options in codecs.conf - but has anyone done some research to detemine at what bitrates and other settings Speex offers comparable call quality with g729? Alternatively, has anyone done any subjective comparisons between iLBC and Speex at various bitrates? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] Linear Queues Strategies for 3rd Party Application
Does anyone know how to setup a linear type of queue strategy? By that I mean that agents will be tried in a particular order and the call will be routed to them unless they are on the phone or not logged in. I want a 3rd party app to be able to re-arrange this order on the fly based on sales and other metrics. Anybody setup something similar? Any pointers or products already out there open source or not? Thanks, Steve Totaro ___ --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] Linear Queues Strategies for 3rd Party Application
I would use an agi and the local channel with SQL running the logic from an AGI. Anybody setup something similar? Any pointers or products already out there open source or not? I have done this before. Thanks, Steve Totaro ___ --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] Re: Call centre - * hang's up
- Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 10:46 PM Subject: [Asterisk-Users] Re: Call centre - * hang's up In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I think it's a bit of a known fault - the attended transfer function does not work from the queue system. It would be nice if it did, though. Hi Paul! Is there any explanation about this? Is that something that will change? It probably should change - somebody different asks the question on the list here every month or so. Has anyone logged this onto bugs.digium.com??? PaulH ___ --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] Application Faxing using SIP
Losing an audio packet here or there wouldn't normally be so bad for fax. Normally I would expect the fax protocol, especially ECM protocol, to be able to recover from it. However, Asterisk seems to not work in an ideal fashion for this purpose. Whenever Asterisk encounters a lost audio packet something called packet loss concealment is performed by placing a PLC frame there as a placeholder. When the audio is retransmitted the PLC frame is supposed to be converted into synthesized audio. Between what I have been told and from what I have observed, this conversion of PLC frames into synthesized audio does not happen with uLaw, alaw, or slinear codecs (the only codecs suitable for fax). Consequently the PLC frame is converted into zero-data... or 20 ms of silence... which is probably the worst-possible thing that could happen. Turning ECM seems to cause most of my issues with FAX. Most newer machines have this on by default. However if there is any packet loss, then when ECM tries to resend and there is additional loss, then it gets in a loop and everything just fails. Whereas with ECM off, you may have an occasional extra or missing pixel, but most users never notice, and the speed is way faster. Most complaints are solved by jsut turning ECM off. Of course this does not necesarily help mortgage companies who seem to enjoy faxing 50page legal docs ___ --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: Call centre - * hang's up
On Tue, 21 Feb 2006, [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] says... I think it's a bit of a known fault - the attended transfer function does not work from the queue system. It would be nice if it did, though. Hi Paul! Is there any explanation about this? Is that something that will change? It probably should change - somebody different asks the question on the list here every month or so. Has anyone logged this onto bugs.digium.com??? it will probably get rejected as a feature request not a bug, post it on the voip-info bounty page -Dan ___ --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] Grandstream GXP-2000
On Mon, 20 Feb 2006, Soner Tari wrote: For 100bit issue, I tend to believe in the data sheet, but I would also like to hear a first-hand verification. (But I guess we have to wait, because voipsupply accepts pre-sale orders for now, they don't ship them yet.) The SPA-942 is $179.95, I would rather buy a polycom 501 ($169.95). -Dan ___ --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] Application Faxing using SIP
Jerry Jones wrote: Turning ECM seems to cause most of my issues with FAX. Most newer machines have this on by default. However if there is any packet loss, then when ECM tries to resend and there is additional loss, then it gets in a loop and everything just fails. Whereas with ECM off, you may have an occasional extra or missing pixel, but most users never notice, and the speed is way faster. Most complaints are solved by jsut turning ECM off. Of course this does not necesarily help mortgage companies who seem to enjoy faxing 50page legal docs A 50-page non-ECM fax is similar to an average 20-page ECM fax in that there are repeated sections of V.17/V.29/V.27ter modulation occuring. That high-speed data communcation is the really sensitive part about faxing and getting a mistaken carrier drop during that time is the thing that kills. If you're saying that a 50-page legal document will frequently have troubles, then you're talking about an error ratio that most businesses of my acquaintence would simply not tolerate. In a typical lossless audio environtment, fax speeds with ECM should be the same or better due to additional compression mechanisms that require a lossless image type. If you find that your fax speeds with ECM are frequently slower than without ECM... or if you find that ECM fails more frequently than non-ECM, then it would seem to indicate that the audio corruption that is occurring before audio gets to your fax machine is fairly severe. Lee. ___ --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] good voip
Can anyone recommend a good voip provider? 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
Re: [Asterisk-Users] Grandstream GXP-2000
At 11:21 AM 02/20/2006, you wrote: Aastra does not really make it clear that the 480i is poe _only_. A lot of people are very suprised when I explain to them that the 480i is poe only. I thought them made it really clear it was POE only and I was really surprised when I found the wall wart in the box and realized I didn't actually need the POE router I'd purchased. I'm using POE because it's neater, but the 480i CT comes with a power adapter. Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.375 / Virus Database: 267.15.11/264 - Release Date: 02/17/2006 ___ --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
[Asterisk-Users] Dial from AGI = no ring back ??
Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back from the script: exten = _0XX32316200,1,DeadAGI(fred.agi) exten = _0XX32316200,2,Hangup ;Dialing this one directly gives me the ring back exten = _10XX32316200,1,Dial(IAX2/provider/559132316200,60); exten = _10XX32316200,2,Hangup The fred.agi script: #!/usr/bin/perl use DBI; use Asterisk::AGI; $AGI = new Asterisk::AGI; $AGI-answer(); $dialstr = IAX2/provider/559132316200|60; $res = $AGI-exec(DIAL $dialstr); ___ --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] Dell PowerEdge 2850
Hello,Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardwareWe are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a 5V PCI slot. Does Dell PE 2850 has a 5V PCI slot? A person in our group tried to call Dell's customer support but they do not seem to know.We will also be using RHEL ES 4 as the OS.Anybody have experience (good/bad) for this type of configuration? We are going to use it primarily as a conferencing server serving 30-50 simultaneous users.Can anybody recommend an alternative server that works well with TE205P and RHEL ES 4?This is our fi rst time using Asterisk so we would like to have it pain free as much as possible.Thank you very much.richard___ --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] Asterisk Broadvoice Incoming Calls Problems
Hi, i'm having problems with broadvoice incoming calls. I can perfectly place calls but my Asterisk Box is having problems when registering with the SIP Proxy. Sometimes it register and the call gets into asterisk, but without sound (seems to be NAT problems) and sometimes its not possible for asterisk to receive the calls. Everything was working great exactly for a month, but a week ago it crashed and stop working. I dont know if someting happened with the broadvoice service or the problem is mine, cause i tried with the same setup it was working.Im behind a NAT Firewall, so i need a setup that can work in this condition. Here is part of my setup:sip.confexternip=200.42.xxx.xxxnat=yesregister = [EMAIL PROTECTED]:MyPassword1:[EMAIL PROTECTED]register = [EMAIL PROTECTED]:MyPassword2:[EMAIL PROTECTED][301]type=friendregexten=301username=301secret=xcallerid="Agent #1" 301host=dynamicnat=yescanreinvite=nodisallow=allallow=gsmallow=ulawallow=alaw[sip.broadvoice.com]type=peeruser=phonehost=sip.broadvoice.comfromdomain=sip.broadvoice.comfromuser=305x22secret=MyPassword1username=305x22insecure=verycontext=from-broadvoiceauthname=305x22dtmfmode=inbanddtmf=inband;Disable canreinvite if you are behind a NATcanreinvite=nodisallow=allallow=gsmallow=ulawallow=alaw[sip.broadvoice.com1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=305x66 secret=MyPassword2 username=305x66 insecure=very context=from-broadvoice authname=305x66 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw[broadvoice-incoming]context=from-broadvoicedtmf=inbanddtmfmode=inbandfromdomain=sip.broadvoice.comhost=sip.broadvoice.cominsecure=verynat=yessecret=MyPassword1type=useruser=305x22username=305x22[broadvoice-incoming2]context=from-broadvoicedtmf=inbanddtmfmode=inbandfromdomain=sip.broadvoice.comhost=sip.broadvoice.cominsecure=verynat=yessecret=MyPassword1type=useruser=305x66username=305x66extensions.conf[from-broadvoice]exten = s,1,Answerexten = s,2,Wait(2) ; Waits 2 Seconds Before Playing the Welcome Msgexten = s,3,Playback(welcome-message)exten = s,4,Dial(SIP/301,25,Tt) exten = s,5,Hangup/etc/hosts147.135.0.128 sip.broadvoice.com__Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.espanol.yahoo.com/ ___ --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] g729 quality at GSM bitrates
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, February 20, 2006 11:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] g729 quality at GSM bitrates I'm trying to improve the codec selection on a few of the asterisk boxes we have to keep the g729 licences free for calls from ATAs that don't support anything apart from g711 and g729. GSM seems to offer noticably inferior call quality (at least when using a softphone + decent headphones), but it's about where I want the bitrate to be. To my ear, ILBC sounds much better than GSM. It's slightly more efficient, and more tolerant of things like packet loss. Some folks, hate the sound of ILBC encoded calls. shrug Your other choice would be G.726/32. * supports it, as do many ATA's and softphones. It's a bit fatter, but sounds MUCH better than GSM. ___ --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] Trunk calls ring internal analog phone
I am having an issue where outbound external calls. Calls made using an analog line (connected to an FXS) route correctly out the trunk (connected to an FXO). However, when I make a similar outbound call using a SIP phone the analog phone connected to the FXS rings. I was having this problem intermittently with a manual asterisk install -- a reboot would fix the problem. I am now giving [EMAIL PROTECTED] a spin and it happens every time. I have included what I think are the relevant portions of the logs. If any more information would help, please let me know and I'll provide them. When calling from the analog phone (works): Feb 20 15:41:50 VERBOSE[3773] logger.c: -- Executing GotoIf(Zap/1-1, 0?16) in new stack Feb 20 15:41:50 DEBUG[3773] pbx.c: Not taking any branch Feb 20 15:41:50 VERBOSE[3773] logger.c: -- Executing Dial(Zap/1-1, ZAP/g0/5855961) in new stack Feb 20 15:41:50 DEBUG[3773] chan_zap.c: Dialing '5855961' Feb 20 15:41:50 DEBUG[3773] chan_zap.c: Deferring dialing... Feb 20 15:41:50 VERBOSE[3773] logger.c: -- Called g0/5855961 Feb 20 15:41:51 DEBUG[3773] chan_zap.c: Exception on 17, channel 4 Feb 20 15:41:51 DEBUG[3773] chan_zap.c: Got event Hook Transition Complete(12) on channel 4 (index 0) Feb 20 15:41:52 DEBUG[3773] chan_zap.c: Exception on 17, channel 4 When calling from the sip phone (doesn't work): Feb 20 15:43:48 VERBOSE[3810] logger.c: -- Executing GotoIf(SIP/226-b40f, 0?16) in new stack Feb 20 15:43:48 DEBUG[3810] pbx.c: Not taking any branch Feb 20 15:43:48 VERBOSE[3810] logger.c: -- Executing Dial(SIP/226-b40f, ZAP/g0/5855961) in new stack Feb 20 15:43:48 DEBUG[3810] chan_zap.c: FXO: setup deferred dialstring: 5855961 Feb 20 15:43:48 VERBOSE[3810] logger.c: -- Called g0/5855961 Feb 20 15:43:48 VERBOSE[3810] logger.c: -- Zap/1-1 is ringing Feb 20 15:43:50 DEBUG[3810] chan_zap.c: Exception on 16, channel 1 Feb 20 15:43:50 DEBUG[3810] chan_zap.c: Got event Ringer Off(11) on channel 1 (index 0) Feb 20 15:43:50 VERBOSE[3810] logger.c: -- Zap/1-1 is ringing Feb 20 15:43:50 DEBUG[3810] chan_zap.c: Exception on 16, channel 1 Feb 20 15:43:50 DEBUG[3810] chan_zap.c: Got event Ring/Answered(2) on channel 1 (index 0) Feb 20 15:43:50 DEBUG[3810] chan_zap.c: Enabled echo cancellation on channel 1 Feb 20 15:43:50 DEBUG[3810] chan_zap.c: Engaged echo training on channel 1 Feb 20 15:43:50 DEBUG[3810] chan_zap.c: channel 1 answered Feb 20 15:43:50 DEBUG[3810] chan_zap.c: Sent FXO deferred digit string: Tw5855961 Thanks in advance, Steven ___ --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] good voip
Where are you located? That makes a big difference! PaulH Melbourne, Australia - Original Message - From: CyberSource [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, February 21, 2006 7:37 AM Subject: [Asterisk-Users] good voip Can anyone recommend a good voip provider? 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 ___ --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] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I have now set the resyncthreshold to -1, to turn it off. I have also set the maxjitterbuffer to 2000. I still received 10 complaints of choppy calls today on Asterisk 1.2.4 versus only 1 complaint on Asterisk 1.07. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Monday, February 20, 2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning Adam Robins wrote: Hi Adam After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls breaking up from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. I am interested to know why you are using ilbc, n why not g729 ot g723 or speex. What is the size of the WAN connection. How many calls are you running over this link. I just need to see how others are fairing with IAX2 over WAN links, as I am the final stages of testing on my side thanks, yusuf ___ --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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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] CISCO 1760 with 1 BRI
will try that, thanks bob ! jl2006/2/16, Bob Goddard [EMAIL PROTECTED]: On Thursday 16 Feb 2006 22:20, Jean-Louis curty wrote: hi, My question is may be a bit out of scope but I don't know where to turn, I have a 1760 with a ccme 24 user licences 1 bri card. I want to configure a bri card in a cisco 1760 on port 0/0, the card is new, seen by the router, show isdn status gives layer 1 desactived , layer not activated, what ever I do, no shutdown command / shutdown command, etc , the green OK light never turn on, what doi miss,Assuming that your wires correct, no crossover, plugged in,under the bri interface, try the following command(s)isdn tei-negotiation first-call~or~ isdn tei-negotiation powerup[...]B--http://www.mailtrap.org.uk/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] good voip
[EMAIL PROTECTED] wrote: Where are you located? That makes a big difference! PaulH Melbourne, Australia - Original Message - From: CyberSource [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, February 21, 2006 7:37 AM Subject: [Asterisk-Users] good voip Can anyone recommend a good voip provider? 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 ___ --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 Buffalo New York USA -- cybersource.us 115 Richfield Road Williamsville, New York 14221 716-553-8525 ___ --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] seg fault when skinny phone answers
hello- i'm having trouble completing a connection between an older skinny phone (12sp+) and a soft sip phone (x-lite). the skinny phone appears to successfully register: -- Starting Skinny session from 192.168.1.50 Device SEP00D0BA03AB66 is attempting to register -- Device 'office' successfuly registered Requesting capabilities Version Request Received CapabilitiesRes Buttontemplate requested Sending 12SP template to [EMAIL PROTECTED] (12SP) Received Time/Date Request when a place a call from x-lite, the 12sp+ rings, and asterisk says: -- Executing Dial(SIP/ion-226a, Skinny/[EMAIL PROTECTED]|20|tr) in new stack Found device: office -- skinny_request([EMAIL PROTECTED]) -- Skinny cw: 0, dnd: 0, so: 0, sno: 0 skinny_new: tmp-nativeformats=4 fmt=4 -- skinny_call(Skinny/[EMAIL PROTECTED]) Trying to send: 2r ämó@' Displaying message 2r ämó@' Displaying Prompt Status 'Ring-In' -- Called [EMAIL PROTECTED] -- Skinny/[EMAIL PROTECTED] is ringing as soon as i answer the call (or hangup from x-lite, or wait for the timeout period), aterisk says: -- Skinny/[EMAIL PROTECTED] answered SIP/ion-226a Segmentation fault (core dumped) in addition, i can't make a call from the 12sp. when i dial x-lite from the 12sp, asterisk says: Attempting to Clear display on Skinny [EMAIL PROTECTED] skinny_new: tmp-nativeformats=4 fmt=4 -- Starting simple switch on '[EMAIL PROTECTED]' Collected digit: [8] -- Asked to indicate 'Stop tone' condition on channel Skinny/ [EMAIL PROTECTED] Collected digit: [1] -- Asked to indicate 'Stop tone' condition on channel Skinny/ [EMAIL PROTECTED] -- Asked to indicate 'Stop tone' condition on channel Skinny/ [EMAIL PROTECTED] Skinny [EMAIL PROTECTED] went on hook Skinny([EMAIL PROTECTED]): waitfordigit returned 0 skinny_hangup(Skinny/[EMAIL PROTECTED]) on [EMAIL PROTECTED] the Asked to indicate... message after repeats indefinitely until the phone is hung up, and x-lite never sees the call. i'm running asterisk 1.2.1 (debian testing package) - below are a few related sections of my config. my apologies if i've omitted something - this is my first experience with asterisk. thanks! -ben --sip.conf: [general] context=home bindport=5060 bindaddr=0.0.0.0 srvlookup=yes --skinny.conf: [general] port = 2000 bindaddr = 0.0.0.0 dateFormat = Y-M-D keepAlive = 120 [office] device=SEP00D0BA03AB66 host=192.168.1.50 context=home line = 1234 model=12SP version=P00203010003 callerid=office 84 --extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp IAXINFO=guest TRUNK=Zap/g2 TRUNKMSD=1 [home] exten = 81,1,Dial(SIP/ion,20,tr) exten = 82,1,Dial(SIP/quark,20,tr) exten = 83,Dial(SIP/proton,20,tr) exten = 84,1,Dial(Skinny/[EMAIL PROTECTED],20,tr)___ --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
