[Asterisk-Users] Asterisk 1.2.0 - TE210P - ...Control Frame 15...
Hi all, I have a TE210P connected to two E1 and everything seems fine. I create a script that originates a call from the first E1 to the second E1 and then starts playing an announcement, the extension that answers the call starts recording the announcement and place that in a directory. The recorded files have not the same size and at the Asterisk console i keep getting a NOTICE message saying: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15 Can somebody help with that? What is this control frame 15? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura 3000 Call waiting on the PSTN line
OK I have set the time and message Luki writes: Has anybody been able to get call waiting on the PSTN line? As far as I recall, you will only hear a tone in the audio stream when a second call comes in. The Sipura does not detect or handle it, but if you flash the line on the FXS interface after hearing the tone, the Sipura will forward the flash to the FXO interface and hence switch to the second call. I am positive this works when the call is picked up on the local FXS port but I am not sure if it also works when the call is picked up by a remote device. This is how I had set it up: PSTN - FXO on Sipura 3K - Asterisk (remote via DSL) - FXS on Sipura 3K - Phone The call would be re-invited in this case so no RTP traffic goes via DSL, only SIP traffic. Switching to second call with flash works in this scenario. Additionally I also allowed the call to be received by a remote device (RTP via DSL) but I am not sure if you can then use Call Waiting (never tried it). I don't think I'm expressing myself clearly here; if not, please ask. Or correct me if I'm wrong. Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failover Registration
Try something like this.
Note: I did not write these scripts. I would give credit to who did, but
unfortunately I do not remember where I got it.
Dan
[globals]
TRUNK1 = IAX2/user:[EMAIL PROTECTED]
TRUNK2 = IAX2/user:[EMAIL PROTECTED]
; Sets up the outgoing gateway according to availability
[macro-swap-priority]
exten = s,1,NoOp(Swapping trunk priority)
exten = s,n,SetGlobalVar(TRUNKBUF=${TRUNK1})
exten = s,n,SetGlobalVar(TRUNK1=${TRUNK2})
exten = s,n,SetGlobalVar(TRUNK2=${TRUNKBUF})
exten = s,n,NoOp(Swapped)
exten = s,n,NoOp(Priority 1 ${TRUNK1})
exten = s,n,NoOP(Priority 2 ${TRUNK2})
; calls the swap-priority macro to find out which gateway is set to the
default and dials the number.
[macro-outbound-dial]
exten = s,1,Wait(3)
exten = s,n,Set(TIMEOUT(response)=60)
exten = s,n,Dial(${TRUNK1}/${ARG1})
exten = s,n,NoOp(TRUNK1 failed)
exten = s,n,SetVar(A=2)
exten = s,n,NoOp(rolling over to TRUNK2)
exten = s,n,Playback(hang-on-a-second)
exten = s,n,Macro(swap-priority)
exten = s,n,Wait(2)
exten = s,n,Dial(${TRUNK1}/${ARG1})
exten = s,n,Playback(all-outgoing-lines-unavailable)
exten = s,n,Playback(please-try-again-later)
exten = s,n,Hangup()
;Call outbound-dial macro
[from-inside]
exten = _1XX,1,Macro(outbound-dial,${EXTEN})
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Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!
Remco Barende wrote: Weird, I checked with KPJ before and he mentioned it is normal behaviour for ISDN. My console is filled with messages like this : == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down Well, just wanted to share my experiences, over here in Belgium. We do not have this behaviour on ISDN lines. (I'm using a quadBRI from junghanns, but have also used plain hfc pci cards) My signalling type (for the quadBRI) in zapata.conf is: ; p2p TE mode (for connecting ISDN lines in point-to-point mode) signalling = bri_cpe cheers ___ --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] ZapHFC cards not maintaining sync?!
On Mon, December 5, 2005 7:22, Remco Barende said: Already HAVE Florz patch installed! :-( What version of * and BRIstuff are you using? Strange, sounds like the florz patch has not been effectively applied or it's broken. I'm using an old version of bristuff : Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n built by [EMAIL PROTECTED] on a x86_64 running Linux The patch applied fine, No errors, nor .rej files I had an issue compiling it for x86_64 though, but that's a different question. I assumed as much when I saw your last name... :-) Whereabouts in NL? I'm in Zoetermeer (ZH)... Amsterdam, but the ISDN setup I installed near Leiden (ZH) :) Close enough! ;-) 1) Every 10 seconds () the D channel gets torn down, which That's too slow, it should happen about every 1-2 seconds or so. The d channel going down and up again is normal behaviour. I know it is. Used to work for a Networking Competence Centre, and we had the same kind of issues with 3Com Netbuilders. The first call attempt after the D Channel was torn down always failed... The only solution was to get KPN to turn on the D Channel permanently... Strange, I never had that problem before. When the * box gets up I can immediately make calls. Also the standard KPN A/B equipment doesn't have this problem, sounds like it's more 3Com related. It was... 3Com didn't recover from the tear-down. Only solution was preventing the teardown ;-) One problem I have found with bristuff (and no solution yet), if you disconnect the ISDN line from the * box (or the ISDN line is out of order for a short while), bristuff will not re-establish the connection. It is then required to unload all the modules and re-load them or even worse, reboot the box. I guess that is a specific bristuff problem. All calls to the ISDN line fail and it's not possible to make any calls. Even after several hours bristuff doesn't setup the line connection. Not seen that yet... I can unplug the line, plug it in again and it'll bring it back up... 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... I noticed that the 'deactivated' issue doesn't happen for a while after a call has been placed. I am now testing placing a call every minute, with a 100 ms timeout using the manager api. This means it never actually gets a chance to get through, or be picked up, but it does cause activity on the D channel. This has been running for half an hour now, and I haven't seen the channel go down for extended periods since. I'm not sure whether the KPN will like it, but it's an interesting test to run! G Good luck with our Royal Dutch KPN, but I would try florz first :) Tell me about it! Like I said above, we had *extensive* experience with them over the D Channel issue! It is in NL, but that is because the KPN have decided to do it that way. *Normal* PSTN's keep D channel alive Weird, I checked with KPJ before and he mentioned it is normal behaviour for ISDN. My console is filled with messages like this : == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down and it doesn't cause me any issues. It would be nice to 'hide' these messages when not in very verbose mode to avoid cluttering up the console. The messages indeed do appear about every 10 seconds or so. Yep... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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.h
With asterisk-1.0.9 version when I wanted to enable logging of CDR in MySQL I needed to make softlink ln -s /usr/src/asterisk-1.0.9/asterisk.h /usr/src/asterisk-addons-1.0.9/asterisk.h Now, on version 1.2.0 I don't have that file. Do I need it? Thank you for your time. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Chat Lines/ Party Line Solutions for Asterisk
Does anyone know of any chat lines/party line programs/agi/add ons for asterisk to handle suck type of operations? ___ --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] AGI Problem
See:Dec3 01:16:52 WARNING[20212]: chan_iax2.c:2747 create_addr: No such host:24
Are u sure exists 24 iax device ?Try with ip
2005/12/3, Cyrille Demaret
[EMAIL PROTECTED]:Hi,Same result with dial:-- Executing DeadAGI(SIP/205-0231, b) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/b-- AGI Script Executing Application: (Dial) Options: (IAX2/24)Dec3 01:16:52 WARNING[20212]: chan_iax2.c:2747 create_addr: No such host:24Dec3 01:16:52 NOTICE[20212]: app_dial.c:1011 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 3 - No route to destination)== Everyone is busy/congested at this time (1:0/0/1)b: 200 result=0-- AGI Script Executing Application: (Dial) Options: (IAX2/24)
Dec3 01:16:52 WARNING[20212]: chan_iax2.c:2747 create_addr: No such host:24Dec3 01:16:52 NOTICE[20212]: app_dial.c:1011 dial_exec_full: Unable tocreate channel of type 'IAX2' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)b: 200 result=1-- AGI Script Executing Application: (Dial) Options: (IAX2/24)Dec3 01:16:52 WARNING[20212]: chan_iax2.c:2747 create_addr: No such host:
24Dec3 01:16:52 NOTICE[20212]: app_dial.c:1011 dial_exec_full: Unable tocreate channel of type 'IAX2' (cause 3 - No route to destination)== Everyone is busy/congested at this time (1:0/0/1)b: 510 Invalid or unknown command
-- AGI Script b completed, returning 03 different results...Regards,Cyrille-Message d'origine-De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] De la part de GiovanniMianoEnvoyé: vendredi 2 décembre 2005 16:18À: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: Re: [Asterisk-Users] AGI ProblemI thing u cant use ChanIsAvail with exec command... as use EXEC DIAL(SIP/40) .. it isnt work2005/12/2, Cyrille Demaret
[EMAIL PROTECTED]: Hi, I've changed that and it's the same problem. I've this problem with all applications. Results from agi are not correct. Regards, Cyrille
-Message d'origine- De: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] De la part de Giovanni Miano Envoyé: vendredi 2 décembre 2005 12:52 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [Asterisk-Users] AGI Problem
Try print EXEC ChanIsAvail IAX2/24\n; Channel type is IAX2 not IAX Cheers 2005/12/2, Cyrille Demaret [EMAIL PROTECTED]
: Hi, I'm running the last CVS asterisk version (I was running an olderversion before with the same problem) and I've a problem with agi scripts. Commands
results are not always correct. I've made a small agi test script that execute ChanIsAvail on an inexistent extension:
#!/usr/bin/perl $|=1; while(STDIN) { chomp; last unless length($_); if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2; } } # Check print EXEC ChanIsAvail IAX/24\n; $result = STDIN; print VERBOSE \$result\ 0\n;
# Check print EXEC ChanIsAvail IAX/24\n; $result = STDIN; print VERBOSE \$result\ 0\n; # Check
print EXEC ChanIsAvail IAX/24\n; $result = STDIN; print VERBOSE \$result\ 0\n;
Result is : -- Executing DeadAGI(SIP/200-60d2, b) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/b
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24) Dec2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type registered for 'IAX'
b: 200 result=-1 -- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24) Dec2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type registered for 'IAX'
b: 200 result=1 -- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24) Dec2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type registered for 'IAX'
b: 510 Invalid or unknown command -- AGI Script b completed, returning 0 The first result is ok (-1) but not the second and the third.
Why do I get different results for the same command? Thank you, Regards, Cyrille ___
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Re: [Asterisk-Users] Connecting 2 Asterisk using SIP
Try this
___
1st Machine sip.conf
[box2]
username=box1
type=friend
host=10.0.0.2
secret=*
in extensions.conf
exten = _XX,1,Dial(SIP/box2/${EXTEN})
__
2nd Machine sip.conf
[box1]
username=box2
type=friend
host=10.0.0.1
secret=*
in extensions.conf
exten = _X,1,Dial(SIP/box1/${EXTEN})
--xce
*** REPLY SEPARATOR ***
On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote:
I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
gateway. Currently they are connected using IAX2. I wanted to play
with SIP.
I setup a sip entry (type=friend) in the PSTN gateway box and a sip
entry (type=user) in the second box in order to send calls using SIP
to the second box. This works fine. However, when I setup the second
box as type=friend in order for it to be able to send calls back to
the gateway box, then calls no longer work from gateway box to the
second box. The reported error is:
Dec 5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite:
Failed to authenticate on INVITE to '2125551212 sip:
[EMAIL PROTECTED];tag=as0698b1b9'
In the gateway box, my sip.conf looks like this:
[general]
allowguest=yes
autocreatepeer=no
[secondbox]
type=friend
host=10.0.0.2
secret=mysecret
In the second box, my sip.conf looks like this:
[general]
allowguest=yes
autocreatepeer=no
[secondbox]
type=user
host=10.0.0.1
secret=mysecret
Any ideas on how to correctly set this up?
Thanks,
Waldo
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[Asterisk-Users] Re: [Asterisk-biz] UK DID 0208 £1 per month
UK, London Based DID £1 per month All number begin with 0208 0xx Sam, Please, if you are going to market London numbers, format them correctly! The code for London is 020, therefore your numbers are 020 80xx . [Blatent self-plug] If you or anyone wants to purchase numbers from the rest of the UK we can offer DID/DDI from all UK area codes, *but* in wholesale qualities only. Linus Magrathea ___ --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] Iax2 connection failed
On 4 Dec 2005, at 21:14, chawki hammoud wrote:Hi:Sorry,but i dont know what ethereal is,and for myasterisk version the iax is good on it because i madea lot of succesful iax connections with many voipproviders like "sixtel,voipjet..."Yep, I agree, your asterisk should be fine,but this provider is sending a message it doesn't understand.You can either ask the provider to tell you what it means by calling their support line, or you can try and find out yourself.If you decide to find out for yourself, the easiest way would beto use a network packet capture program that understandsIAX2 (e.g. ethereal) and inspect the HANGUP packet they sendto see what the cause code is.I'm guessing that they are sending a non-standardcause code that isn't in asterisk. This is a bit ofa grey area because the IAX2 protocol documentimplies that all the Q.931 cause codes are valid,but only lists a sub-set. It could be that you provider is sending a valid Q.931 code.Tim. 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] Sound problem
Hi all I had allread install asterisk server and two X-Lite softphones on two different machines. whole processa of calling is going fine. But I cann't able to hear ringing / any type of voice on both side. The asterisk sever give following worning. WARNING[1922]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' Dec 5 15:41:27 WARNING[1922]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player Can any one help? what is wrong with this. Thanks Vipul ___ --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] diax not working properly
Hi, - Original Message - From: amna saleem [EMAIL PROTECTED] I have been using Asterisk-1.0.3 for quite some time now.My main aim nowadays is to make iax-iax calls for which i am usin DIAX soft phone.Theproblem is that sometimes the phone doesn`t register and at others it gets out of the registration(after being registere for some time).Can anyone tell me what can be the problem ,what other iax phones are available ? Did you have this problems with older Asterisk versions too? There is anybody else having this issue with DIAX and the new Asterisk version? I have develop it and test it with Asterisk till version 1.2. Best regards, 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
[Asterisk-Users] (warning) iaxy.bin fails checksum
Hello, Since I've installed Asterisk 1.2 (from CVS) on a gentoo server, I've got this warning when loading chan_iax2: WARNING[10204]: chan_iax2.c:1254 try_firmware: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum There is no problem after, IAX works well. Is that a problem? How can I remove this warning? -- Benoit Merouze Network Software Developer [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] SIP INVITE with no 'Contact' field and RealTime support.
Hi, When sip device sends to Asterisk INVITE with no 'Contact' field, the server should respond with all information it holds about client. When reading database fields, 'fullcontact' is empty. So, whole procedure ends with 'chan_sip.c:6393 register_verify: Failed to parse contact info'. Interesting thing, internal database (CLI databse show SIP/Registry x) holds all valid information about this client, so why it's not used? ___ --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] Calls to DISA over ISDN PRI don't get CONNECT ACKNOLEDGE
Hi ! I'm having a problem with calls that come over ISDN PRI and go to DISA
app. Problem doesn't happen with calls from SIP phones to DISA or for a
calls over ISDN PRI to SIP phones. Asterisk is 1.2.0
This renders DISA completely unusable when call comes over PRI, since every
call gets hunged up, about 10 seconds after it is answered by Answer or
DISA.
Here is relevant part of extensions.conf:
[default]
exten = 209,1,Macro(superdial, SIP/209,,rtT,,${EXTEN},1)
exten = 299,1,Answer() ;tried without this also, but with same results
exten = 299,2,DISA(no-password)
---
Here is PRI debug of call to DISA:
Protocol Discriminator: Q.931 (8) len=37
Call Ref: len= 2 (reference 67/0x43) (Originator)
Message type: SETUP (5)
[a1]
Sending Complete (len= 1)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
[6c 04 80 31 30 39]
Calling Number (len= 6) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Presentation permitted, user
number not screened (0) '109' ]
[70 04 80 32 39 39]
Called Number (len= 6) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0) '299' ]
[7c 03 80 90 a3]
IE: Low-layer Compatibility (len = 5)
[7d 02 91 81]
IE: High-layer Compatibility (len = 4)
-- Making new call for cr 67
-- Processing Q.931 Call Setup
-- Processing IE 161 (cs0, Sending Complete)
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
-- Processing IE 124 (cs0, Low-layer Compatibility)
-- Processing IE 125 (cs0, High-layer Compatibility)
Protocol Discriminator: Q.931 (8) len=10
Call Ref: len= 2 (reference 67/0x43) (Terminator)
Message type: CALL PROCEEDING (2)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
-- Accepting call from '109' to '299' on channel 0/1, span 1
-- Executing Answer(Zap/1-1, ) in new stack
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 67/0x43) (Terminator)
Message type: CONNECT (7)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
[1e 02 81 82]
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Private network serving the local user (1)
Ext: 1 Progress Description: Called
equipment is non-ISDN. (2) ]
-- Executing DISA(Zap/1-1, no-password) in new stack
Timed out looking for connect acknowledge
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 67/0x43) (Terminator)
Message type: DISCONNECT (69)
[08 02 81 90]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user (1)
Ext: 1 Cause: Unknown (16), class = Normal Event (1) ]
Protocol Discriminator: Q.931 (8) len=5
Call Ref: len= 2 (reference 67/0x43) (Originator)
Message type: RELEASE (77)
-- Channel 0/1, span 1 got hangup
== Spawn extension (default, 299, 2) exited non-zero on 'Zap/1-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
Request
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 67/0x43) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 02 81 90]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user (1)
Ext: 1 Cause: Unknown (16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
---
For a comparation purposes here is PRI DEBUG for a call to SIP phone, over
same ISDN PRI, where the CONNECT message is ACK'ED.
---
Protocol Discriminator: Q.931 (8) len=37
Call Ref: len= 2 (reference 68/0x44) (Originator)
Message type: SETUP (5)
[a1]
Sending Complete (len= 1)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931
Re: [Asterisk-Users] New to [EMAIL PROTECTED]
Thanks all! You've been very helpful!!! - Original Message - From: Tom Vile [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 04, 2005 10:36 PM Subject: Re: [Asterisk-Users] New to [EMAIL PROTECTED] Better yet get a Telasip account at telasip.com On 12/4/05, Kerry Garrison [EMAIL PROTECTED] wrote: You have to have a Digium TDM400, a clone X100P (eBay, about $10), or Sipura SPA-3000. A regular phone modem will not work. Another way is to get a cheap VOIP account like Broadvoice for $10 a month, then you don't need a regular phone line for the kids and you can have two calls going at the same time. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Sunday, December 04, 2005 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] New to [EMAIL PROTECTED] I'm thinking of Installing [EMAIL PROTECTED] on my PC, to contol and route calls to each of my children's computer via SoftPhone X-lite. I downloaded the ISO image, and familiar with the process to install Asterisk now. However I don't have a digium modem. Can I use any regular phone modem for incoming line on the Asterisk server? ---Dakota the Newbie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 Call waiting on the PSTN line
Luki wrote: Has anybody been able to get call waiting on the PSTN line? As far as I recall, you will only hear a tone in the audio stream when a second call comes in. The Sipura does not detect or handle it, but if you flash the line on the FXS interface after hearing the tone, the Sipura will forward the flash to the FXO interface and hence switch to the second call. I am positive this works when the call is picked up on the local FXS port but I am not sure if it also works when the call is picked up by a remote device. This is how I had set it up: PSTN - FXO on Sipura 3K - Asterisk (remote via DSL) - FXS on Sipura 3K - Phone The call would be re-invited in this case so no RTP traffic goes via DSL, only SIP traffic. Switching to second call with flash works in this scenario. Additionally I also allowed the call to be received by a remote device (RTP via DSL) but I am not sure if you can then use Call Waiting (never tried it). The flash in the above is intercepted and used by the spa3k, and is typically used by the sipura for its implementation of special features. The flash is not forwarded to the pstn line unless one programs the spa3k to operate in a different mode. Sipura added a configuration option (I think it came out in v3.1.7 code) that allows one to program a double-hook-switch-flash that is forwarded to the pstn line. In the OP's config, asterisk is in the middle of the call, therefore something would need to be configured in asterisk to handle generate the flash to the pstn interface. But, I've not tried or even researched how that might be accomplished. ___ --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] oh323 installation
Hi all, I am getting some error while i am trying to install oh323. I already installed Pwlib and openh323 library, But i do not know from where the following error is apearing. asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/local/other/h323/asterisk-oh323-0.7.3/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED] asterisk-oh323-0.7.3]# Please help me to solve this issue. -- Thank You, Code Lover ___ --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] Sipura 3000 Call waiting on the PSTN line
Luki wrote: Has anybody been able to get call waiting on the PSTN line? As far as I recall, you will only hear a tone in the audio stream when a second call comes in. The Sipura does not detect or handle it, but if you flash the line on the FXS interface after hearing the tone, the Sipura will forward the flash to the FXO interface and hence switch to the second call. I am positive this works when the call is picked up on the local FXS port but I am not sure if it also works when the call is picked up by a remote device. This is how I had set it up: PSTN - FXO on Sipura 3K - Asterisk (remote via DSL) - FXS on Sipura 3K - Phone The call would be re-invited in this case so no RTP traffic goes via DSL, only SIP traffic. Switching to second call with flash works in this scenario. Additionally I also allowed the call to be received by a remote device (RTP via DSL) but I am not sure if you can then use Call Waiting (never tried it). The flash in the above is intercepted and used by the spa3k, and is typically used by the sipura for its implementation of special features. The flash is not forwarded to the pstn line unless one programs the spa3k to operate in a different mode. Sipura added a configuration option (I think it came out in v3.1.7 code) that allows one to program a double-hook-switch-flash that is forwarded to the pstn line. In the OP's config, asterisk is in the middle of the call, therefore something would need to be configured in asterisk to handle generate the flash to the pstn interface. But, I've not tried or even researched how that might be accomplished. ___ --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] Sipura 3000 Call waiting on the PSTN line
cp wrote: I have configured my SPA-3000 to direct all calls received over PSTN interface to Asterisk. That is set up via a dial plan with S0 and works fine. CID information is passed along as well. But I find it impossible to setup call waiting on the PSTN line. I do subscribe to this service from my PSTN provider (SBC). Once a PSTN call comes in, is forwarded to asterisk and then to a phone at line 1, Sipura seems oblivious to any other calls coming in on PSTN. When I dial my PSTN number with a second call I don't see any activity on Sipura. I have set up remote logging and the level is set to 3. But I still get no activity in the log for the second call. Even the first call logs very little, but it logs something. Has anybody been able to get call waiting on the PSTN line? The syslog feature in the spa3k does not take affect until after the spa3k has been power cycled. It's not a dynamic parameter change like most of the other config parameters are. See other post on the call waiting topic. ___ --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] Include a variable from another file in config files
I would like to know if it is possible to include a variable in sip_nat.conf. I have a file with my network configuration and I want to parse it and to use LAN IP in sip_nat.conf. Is there a way to parse a file and include variables in a .conf file. Amaury ___ --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] Sound problem
Attention:Your mp3s arent higher than 128 bit/s2005/12/5, Vipul Patel [EMAIL PROTECTED]: Hi all I had allread install asterisk server and two X-Lite softphones on two different machines. whole processa of calling is going fine. But I cann't able to hear ringing / any type of voice on both side. The asterisk sever give following worning. WARNING[1922]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' Dec 5 15:41:27 WARNING[1922]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player Can any one help? what is wrong with this. Thanks Vipul ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --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] Include a variable from another file in config files
You can use shell script to generate sip_nat.conf file2005/12/5, Amaury BOSSE [EMAIL PROTECTED]: I would like to know if it is possible to include a variable in sip_nat.conf. I have a file with my network configuration and I want to parse it and to use LAN IP in sip_nat.conf. Is there a way to parse a file and include variables in a .conf file. Amaury ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --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] Include a variable from another file in config files
Amaury BOSSE a écrit : I would like to know if it is possible to include a variable in sip_nat.conf. I have a file with my network configuration and I want to parse it and to use LAN IP in sip_nat.conf. Is there a way to parse a file and include variables in a .conf file. Amaury In your sip.conf #include /path/to/the/file/you/want/to/include In this file Asterisk will find the command, eg localnet=your LAN IP -- Daniel ___ --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] [EMAIL PROTECTED] with a2Billing
Hi i have installed [EMAIL PROTECTED] and working as of with my extensions and Sip provider now iam looking to deploy prepaid application with a2billing does any one successfully integrated or any other docs make me to integrate i dont see full docs even at [EMAIL PROTECTED] site iam still googling to get some good info help will be appriciated ram ___ --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: sound problem in X-Lite phone with asterisk server
Hi, Two things does your codec set in X-lite match what is set in the sip file and have you rebooted since setting up music on hold. I should also ask if ran a make and make install in the asterisk-addons directory, this installs a mp3 player (among other things) in Asterisk 1.2? Vipul Patel wrote: Hi all I am a newbie to the asterisk. I just installed asterisk server and two X-Lite softphones. I allready configured sip.conf and extension.conf. Now when i call from one softphone to other , sip signaling is going perfect. Both phone are in ringing mode. But i can't able to hear ring. When i pickup call, there is not any sound at all. The asterisk server give following output during call: Dec 5 12:49:57 NOTICE[1931]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Dec 5 12:49:57 WARNING[1931]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' Dec 5 12:49:57 WARNING[1931]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player Can any one pls tell me where i am going wrong. Thanks Vipul ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.11/191 - Release Date: 12/2/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys SPA-841 Missing Calls
Thanks all for the replies. I've narrowed it down to the phones dislike for my older 3COM switch. I noticed on the weekend that when these missed calls occur, if I ping the phone, the first few packets are dropped..almost like it's gone to sleep.. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Sent: Saturday, December 03, 2005 1:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Linksys SPA-841 Missing Calls I experienced a similar situation with the SPA-841, it turned out to be that the calls I was missing didn't have caller ID (outside calls with caller ID Blocked), found that the SPA841 phone has an option to ignore calls without caller ID. Turned this option off and it fixed the problem. Sorry, I no longer use the SPA841 and I can't remember the exact menu setting on the SPA841 that fixed it, so you will have to go through the manual. c Message: 1 Date: Fri, 02 Dec 2005 21:43:01 -0800 From: Wolfgang S. Rupprecht [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Linksys SPA-841 Missing Calls To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Might the SPA-841 be crashing and rebooting? With the current firmware (v. 3.1.4) I often see my phone hang and flash all its lights Really? For me the 841 is a quite stable phone. Out of the 15 we have in the office neither one crashed in the past 3 months. And they are used heavily. The phone has weaknesses, but stability in my opinion is not one of them. Phone info: Software Version: 3.1.4(a) Hardware Version: 1.0.0(1813) Elapsed Time: 50 days and 09:48:10 I only have 1 phone so it is hard to tell if the crashing is a hardware or software problem. I never noticed the phone having problems previous to this. I did resync asterisk to HEAD a month ago. Thats also about the time the phone started crashing (or at least I started noticing it). Come to think of it, I've been running the current firmware in the phone since July 20th. The only think that changed in recently was asterisk. I wonder if there is something the newer asterisk is doing that the phone really hates... Asterisk CVS HEAD built by [EMAIL PROTECTED] on a amd64 running OpenBSD on 2005-11-02 00:58:42 UTC Software Version: 3.1.4(a) Hardware Version: 1.0.0(700b) Elapsed Time: 1 day and 05:54:03 (crashed during a call) People have been reporting a finicky ethernet connector, so maybe that is the reason the phone does not answer to any traffic? Yea, this phone has that problem too. ;-) Some cables just don't work. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.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 ___ --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] Streaming MOH
Title: Message Hi, Have someone successfully configured the streaming MOH in Asterisk 1.2.0 using streamplayer? 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
[Asterisk-Users] Problem with a second incoming call on a BRI Zap Channel
Hi, I'm using Asterisk with a BRI Card (HFC Chipset) using the zaphfc driver. I'm encountering the following problem : when the first line is in use and a second incoming call arrive, the console shows the following message : Dec 5 14:40:52 WARNING[2323]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! Does anyone have an idea of what this is ??? FYI : the Asterisk is located in France (so France Telecom as carrier) The worst is that it doesn't do it all the time Any help will be welcome Best regards David ___ --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] diax not working properly
Hi! I have been using Asterisk-1.0.3 for quite some time now.My main aim nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The problem is that sometimes the phone doesn`t register and at others it gets out of the registration(after being registere for some time).Can anyone tell me what can be the problem ,what other iax phones are available ? I don't think your problem is DIAX, Dan is making a great phone and he test it carefully. But anyway, since you asked, here is a short list : - MediaX (my own) : http://www.marccharbonneau.com/asterisk/mediaxphone.php - Idefix : http://www.asteriskguru.com/tools/idefisk_beta.php - IAX phone : used to be at this address : http://www.sokol-associates.com/IaxPhone.htm but the site changed and I lost track of it - MozIAX : plugin for Firefox/Mozilla : http://moziax.mozdev.org/ - iaxComm : http://iaxclient.sourceforge.net/iaxcomm/ hth Thanx and Regards, Amna ___ --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] SNOM and 1.0.9
This feature has worked for us since ver 1.0 (not cvs) Alvaro Parres wrote: Josheph: I had have that problem, and it get solve when i take out the incominglimit from my sip.cfg Also if you send you sip.cfg and extensions.cfg will be easier to help you Tray it. Alvaro Parres On 11/28/05, *BJ Weschke* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 11/28/05, Kevin Hanson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Joseph Rothstein wrote: Greetings to all, I am trying to get the line lights on a SNOM 320 to work using 'hint' in extensions.conf. Unfortunately I have not been able to get it to work properly. Does anyone know for sure if the hint function works properly in 1.0.9? If anyone has gotten this to work properly under 1.0.9 please post a sample. This is definitely a 1.2 only feature. It is not in 1.0.9. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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 ___ --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] VegaStream 400
Hi All Apologise if this has been previously asked but I am fairly new to the list. I have a VegaStream 400 and have succesfully connected the asterisk to the box to make outgoing calls with no problems. I cannot for the life of me work out how to recieve incoming calls. I have looked around and cannot find any information regarding this, can someone help? Thanks Scott Pinhorne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys SPA-841 Missing Calls
Dave Morrow wrote: Thanks all for the replies. I've narrowed it down to the phones dislike for my older 3COM switch. I noticed on the weekend that when these missed calls occur, if I ping the phone, the first few packets are dropped..almost like it's gone to sleep.. Not likely to be the switch if everything continues to function through that switch. It is entirely possible for the ping function to miss one or two attempts while your system conducts the normal arp discovery process; that's fairly normal, particularly for older equipment. ___ --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] SIP INVITE with no 'Contact' field and RealTime support.
lokotes wrote: When sip device sends to Asterisk INVITE with no 'Contact' field, the server should respond with all information it holds about client. When reading database fields, 'fullcontact' is empty. So, whole procedure ends with 'chan_sip.c:6393 register_verify: Failed to parse contact info'. Interesting thing, internal database (CLI databse show SIP/Registry x) holds all valid information about this client, so why it's not used? This is completely wrong; if the SIP peer sends an INVITE with no Contact information, the request is invalid. Are you talking about REGISTER? If so, that's a known problem, that Asterisk does not currently support registration queries. ___ --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] warning message
Hi I got this warning message repeating itself in the log this morning Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position I had to disable logging to be able to use the console Anybody seen this one ? 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
[Asterisk-Users] Transfer/take call to/from other phone
hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, but my boss is not there, how can i take the phonecall from 401 to 400? Do i need special options in my extensions.conf or is that feature from the isdn phone? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --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] h323 vs oh323
Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] h323 vs oh323
Prely subjective, but I first installed h323 and it worked. Somewhere along the line something happened and it no longer worked. Recompiling it etc seemed to have no effect. I then tried oh323 and it worked first time and has stayed working. I probably did soemthing wrong, but oh323 seems to work for me. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Innocent Evil Sent: 05 December 2005 14:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] h323 vs oh323 Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] Ambient Modem
Hi to all i'm finding the procedures for install the ambient md 3200 chipset modem to make tests, anybody have a link or the procedure to do that?? thanks to all Vladimir __ Visita http://www.tutopia.com y comienza a navegar m�s r�pido en Internet. Tutopia es Internet para todos. ___ --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] Transfer/take call to/from other phone
Hi, Push the '#' key followed by the extension for a blind transfer. Thanks Denny Schierz wrote: hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, but my boss is not there, how can i take the phonecall from 401 to 400? Do i need special options in my extensions.conf or is that feature from the isdn phone? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --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] Connecting 2 Asterisk using SIP
This worked perfectly.
Thanks,
Waldo
On Dec 5, 2005, at 4:32 AM, xcel wrote:
Try this
___
1st Machine sip.conf
[box2]
username=box1
type=friend
host=10.0.0.2
secret=*
in extensions.conf
exten = _XX,1,Dial(SIP/box2/${EXTEN})
__
2nd Machine sip.conf
[box1]
username=box2
type=friend
host=10.0.0.1
secret=*
in extensions.conf
exten = _X,1,Dial(SIP/box1/${EXTEN})
--xce
*** REPLY SEPARATOR ***
On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote:
I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
gateway. Currently they are connected using IAX2. I wanted to play
with SIP.
I setup a sip entry (type=friend) in the PSTN gateway box and a sip
entry (type=user) in the second box in order to send calls using SIP
to the second box. This works fine. However, when I setup the second
box as type=friend in order for it to be able to send calls back to
the gateway box, then calls no longer work from gateway box to the
second box. The reported error is:
Dec 5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite:
Failed to authenticate on INVITE to '2125551212 sip:
[EMAIL PROTECTED];tag=as0698b1b9'
In the gateway box, my sip.conf looks like this:
[general]
allowguest=yes
autocreatepeer=no
[secondbox]
type=friend
host=10.0.0.2
secret=mysecret
In the second box, my sip.conf looks like this:
[general]
allowguest=yes
autocreatepeer=no
[secondbox]
type=user
host=10.0.0.1
secret=mysecret
Any ideas on how to correctly set this up?
Thanks,
Waldo
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Re: [Asterisk-Users] Connecting 2 Asterisk using SIP
username= did it. Thanks, Waldo On Dec 5, 2005, at 2:14 AM, Luki wrote: Any ideas on how to correctly set this up? Try adding authuser= and/or username= to the configuration. Do a SIP DEBUG and see what peer asterisk looks for when trying to authenticate the INVITE. It probably can't find the right peer; authuser on the initiating end should help in this case. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Restore logging functionality...
Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages' for asterisk. I recreated the files by 'touch' filename and I have gone into the Asterisk CLI and tried both 'logger restart' and 'logger rotate' but the logs still show nothing. I run 'logger show channels' and the output below shows up. I have recompiled Asterisk 1.2 and still the logs do not show up. I am getting data into the 'queue_log' and the 'events' logs however so I know logger is running. Any suggestions to fix this??? CLI output tomato*CLI logger show channels Channel Type StatusConfiguration --- --- tomato*CLI tomato*CLI Output from /var/log/asterisk directory [EMAIL PROTECTED] asterisk]# ls -la total 140 drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . drwxr-xr-x 11 root root 4096 Dec 4 04:03 .. drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-custom -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 17 06:41 event_log.1 -rw-r--r-- 1 root root 0 Nov 17 06:45 event_log.2 -rw-r--r-- 1 root root 0 Nov 18 06:38 event_log.3 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log *** Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] kernel lockup with Fedora Core 4.0 2.6.14-1.1637
I have an Asterisk system with Fedora Core 4.0, kernel 2.6.14-1.1637. It sometimes locks up with heavy load (e.g., lots of HDLC messages). This requires a hard reboot. I saw some other reports of hard lockups under load. I have disabled as much as possible in the BIOS and as much as possible in the modules (e.g., removing USB, turning off lots of not-needed services, etc.) Could this be a Fedora problem, zaptel problem, or other? This is reproducible on several systems. I am using ZAPTEL 1.0.9.2. My next test is to try the 1644 kernel update. ___ --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] Restore logging functionality...
Chuck Bunn wrote: drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log you can: delete your logfiles, * will re-create them I think or: change the owner to asterisk. (chown asterisk.asterisk /var/log/asterisk/ -R) cheers ___ --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] Getting started with Asterisk and Aastra 9133i
Let me simplify my problem. I have a single Aastra 9133i SIP phone and latest Asterisk from SVN source running on Fedora Core 4. The phone currently says No Service I would like to be able to dial 1234 from the phone and get Asterisk to play back an audio message or say some digits. I can't get this to work with either SayDigits or Playback. Please help. == sip.conf == [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [3006] type=friend username=3006 secret=mypassword host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all mailbox=3006 === extensions.conf === [tutorial] exten = 1234,1,Answer exten = 1234,2,SayDigits(123456789) ** TFTP directory ** = mymacaddress.cfg = sip line1 auth name: 3006 sip line1 password: mypassword sip line1 user name: 3006 sip line1 display name: myname sip line1 screen name: myname === aastra.cfg === dhcp: 1# DHCP enabled. sip silence suppression: 2 # 0 = off, 1 = on, 2 = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.99# IP of registrar. --- THIS IS THE IP of my Asterisk and tftp server sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.99# Enable time server and enter at ___ --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] Getting started with Asterisk and Aastra 9133i
One more thing. I upgraded the firmware of the 9133i to 1.3. ___ --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] Getting started with Asterisk and Aastra 9133i
On Mon, 2005-12-05 at 11:15 -0500, Robert La Ferla wrote: Let me simplify my problem. I have a single Aastra 9133i SIP phone and latest Asterisk from SVN source running on Fedora Core 4. The phone currently says No Service I would like to be able to dial 1234 from the phone and get Asterisk to play back an audio message or say some digits. I can't get this to work with either SayDigits or Playback. Please help. == sip.conf == [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [3006] type=friend username=3006 secret=mypassword host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all mailbox=3006 === extensions.conf === [tutorial] exten = 1234,1,Answer exten = 1234,2,SayDigits(123456789) ** TFTP directory ** = mymacaddress.cfg = sip line1 auth name: 3006 sip line1 password: mypassword sip line1 user name: 3006 sip line1 display name: myname sip line1 screen name: myname === aastra.cfg === dhcp: 1# DHCP enabled. sip silence suppression: 2 # 0 = off, 1 = on, 2 = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.99# IP of registrar. --- THIS IS THE IP of my Asterisk and tftp server sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.99# Enable time server and enter at I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server tftpserver ip; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with timedate format though). Cheers Pete ___ --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] DISA function
I tried to use DISA 1.2 with regular asterisk (not [EMAIL PROTECTED]), and had problems with it (losing the last digit or occasionally other digits), YMMV. On 12/4/05, Richard Smith [EMAIL PROTECTED] wrote: Hi all, I was wondering whether the DISA function on the latest asterisk 1.2 stable release actually works better than the other prior releases. Basically the [EMAIL PROTECTED] version 2.0 BETA 4 I'm using does not recognise the DTMF tones all the time and sometime when it does, it disconnects. Cheers, 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
Pete Barnwell wrote: I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server tftpserver ip; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with timedate format though). I read about the mac address case sensitivity so I used an all uppercase filename which works fine. The downloading of the firmware works fine too. I also have the ntp time/date working. I just can't get Asterisk to respond to the phone! Help! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
On Mon, 2005-12-05 at 11:27 -0500, Robert La Ferla wrote: Pete Barnwell wrote: I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server tftpserver ip; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with timedate format though). I read about the mac address case sensitivity so I used an all uppercase filename which works fine. The downloading of the firmware works fine too. I also have the ntp time/date working. I just can't get Asterisk to respond to the phone! Help! One thing is to do a factory reset to reinit everything, I did that with my 9112i after upgrading the firmware. -- Dave Cotton [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] Transfer/take call to/from other phone
hi, Quoting Chuck Bunn [EMAIL PROTECTED]: Push the '#' key followed by the extension for a blind transfer. absolut perfect, thanks :-) .Is there also a shortcut, to take a phone call from other phones to me? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --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] Looking for advice on cell carrier's default Un avaliable message
In our dialplan, we use centralized voicemail for SIP, IAX and cell phones. This means, if a caller calls a user's DID, it tries his SIP/IAX extension, then if he doesn't answer there, it tries his cell, then it goes to Comedian Mail. Everything works 100%, except when the user shuts his cell phone off. When that happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played. Asterisk detects this as the call being answered and completes the call. However, this is undesirable behavior. We want it to go to Comedian mail instead. Note that this is contrary to what the carrier said would happen. The carrier indicated to us that it would just ring and ring and ring forever, which is what we want. Now they are saying: too bad, this is the way it works, deal with it In order to have the desired behavior, there are three options: 1. Carrier makes it ring forever (not gonna happen) 2. I set the call forward/Unavailable on the cell to a DID that points to Comedian Mail and do some Caller ID stuff to make it go to the right mailbox. This isn't practical from a management standpoint, it would be troublesome and error prone to maintain 3. When the cell is off, the carrier's Unavailable message plays right away, within 2 seconds of the call being dialed. So, somehow magically modify the dialplan so that if a cell is answered within 2 seconds, go to Comedian Mail. Of these options, 3) would provide the optimum workaround, but I don't think it's possible to express this in an Asterisk dialplan. Anyone have any advice or dialplan magic on how to do 3) ? ? ___ --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] Looking for advice on cell carrier's default Un avaliable message
Look into the findme feature, this will require the person receiving the callto push a buttonhit 1 to accept this call before a callgets transfered to a cell phone (or home phone for that matter), if nobody hits 1 it continuesin the dialplan, this will prevent calls from being transfered to cell phone voicemail or the caller getting the unavailable message from the cell phone carrier. On 12/5/05, Colin Anderson [EMAIL PROTECTED] wrote: In our dialplan, we use centralized voicemail for SIP, IAX and cell phones.This means, if a caller calls a user's DID, it tries his SIP/IAX extension, then if he doesn't answer there, it tries his cell, then it goes to ComedianMail.Everything works 100%, except when the user shuts his cell phone off. Whenthat happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played.Asterisk detects this as the call being answered and completes the call.However, this is undesirable behavior. We want it to go to Comedian mail instead. Note that this is contrary to what the carrier said would happen.The carrier indicated to us that it would just ring and ring and ringforever, which is what we want. Now they are saying: too bad, this is the way it works, deal with itIn order to have the desired behavior, there are three options:1. Carrier makes it ring forever (not gonna happen)2. I set the call forward/Unavailable on the cell to a DID that points to Comedian Mail and do some Caller ID stuff to make it go to the rightmailbox. This isn't practical from a management standpoint, it would betroublesome and error prone to maintain3. When the cell is off, the carrier's Unavailable message plays right away, within 2 seconds of the call being dialed. So, somehow magically modify thedialplan so that if a cell is answered within 2 seconds, go to ComedianMail.Of these options, 3) would provide the optimum workaround, but I don't think it's possible to express this in an Asterisk dialplan.Anyone have any advice or dialplan magic on how to do 3) ? ?___--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] Getting started with Asterisk and Aastra 9133i
Dave Cotton wrote: One thing is to do a factory reset to reinit everything, I did that with my 9112i after upgrading the firmware. I just did that. Now Asterisk is giving me the follow error: (0.99 is my Asterisk server and 0.111 is the phone) Dec 5 12:04:10 NOTICE[14222]: chan_sip.c:10817 handle_request_register: Registration from 'No User sip:[EMAIL PROTECTED]:5060' failed for '192.168.0.111' - Username/auth name mismatch -- Registered SIP '3006' at 192.168.0.111 port 5060 expires 300 ___ --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 won't answer malformed caller id
Hello, Hopefully someone can advise me on the last problem I have in my config. Among my trunks I have an spa-3000 with the pstn connected to an ata-186 that I am trying to bring into asterisk. All works perfectly except apparently when I receive a malformed caller id from this outside service like below. There is no closing quote on this caller id and that's apparently the way it's passed in from the ata-186 to the spa-3000. Asterisk will just not answer this call apparently. Is there any mechanism for asterisk to deal with this? Dec 5 11:14:41 WARNING[8118] chan_sip.c: No closing quote found in 'WIRELESS CALLE sip:[EMAIL PROTECTED];tag=3957b3bfa5fe1a2o1' Dec 5 11:14:41 WARNING[8118] chan_sip.c: Huh? Not a SIP header (WIRELESS CALLE sip:[EMAIL PROTECTED];tag=3957b3bfa5fe1a2o1)? thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for advice on cell carrier's default Un avaliable message
Neat macro but not quite what Im looking for if I force call recipients to press 1 to accept a call they will scream bloody murder. Good idea though. -Original Message- From: Joe Pukepail [mailto:[EMAIL PROTECTED] Sent: Monday, December 05, 2005 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for advice on cell carrier's default Un avaliable message Look into the findme feature, this will require the person receiving the callto push a buttonhit 1 to accept this call before a callgets transfered to a cell phone (or home phone for that matter), if nobody hits 1 it continuesin the dialplan, this will prevent calls from being transfered to cell phone voicemail or the caller getting the unavailable message from the cell phone carrier. On 12/5/05, Colin Anderson [EMAIL PROTECTED] wrote: In our dialplan, we use centralized voicemail for SIP, IAX and cell phones. This means, if a caller calls a user's DID, it tries his SIP/IAX extension, then if he doesn't answer there, it tries his cell, then it goes to Comedian Mail. Everything works 100%, except when the user shuts his cell phone off. When that happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played. Asterisk detects this as the call being answered and completes the call. However, this is undesirable behavior. We want it to go to Comedian mail instead. Note that this is contrary to what the carrier said would happen. The carrier indicated to us that it would just ring and ring and ring forever, which is what we want. Now they are saying: too bad, this is the way it works, deal with it In order to have the desired behavior, there are three options: 1. Carrier makes it ring forever (not gonna happen) 2. I set the call forward/Unavailable on the cell to a DID that points to Comedian Mail and do some Caller ID stuff to make it go to the right mailbox. This isn't practical from a management standpoint, it would be troublesome and error prone to maintain 3. When the cell is off, the carrier's Unavailable message plays right away, within 2 seconds of the call being dialed. So, somehow magically modify the dialplan so that if a cell is answered within 2 seconds, go to Comedian Mail. Of these options, 3) would provide the optimum workaround, but I don't think it's possible to express this in an Asterisk dialplan. Anyone have any advice or dialplan magic on how to do 3) ? ? ___ --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] Transfer/take call to/from other phone
This is what I use. You pre-pend a '4' to the extension number (I used
that because that is how our old pbx worked). There is a number you can
use that will pickup any ringing extension but I forgot what that is. It
should be listed on the asterisk wiki for Pickup.
exten = _4XXX,1,Pickup(${EXTEN:1})
exten = _4XXX,1,Hangup
- James
Denny Schierz wrote:
hi,
Quoting Chuck Bunn [EMAIL PROTECTED]:
Push the '#' key followed by the extension for a blind transfer.
absolut perfect, thanks :-) .Is there also a shortcut, to take a phone
call from
other phones to me?
cu denny
This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] Include a variable from another file in configfiles
Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? -Message d'origine- De : Administrator TOOTAI [mailto:[EMAIL PROTECTED] Envoyé : lundi 5 décembre 2005 12:43 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Include a variable from another file in configfiles Amaury BOSSE a écrit : I would like to know if it is possible to include a variable in sip_nat.conf. I have a file with my network configuration and I want to parse it and to use LAN IP in sip_nat.conf. Is there a way to parse a file and include variables in a .conf file. Amaury In your sip.conf #include /path/to/the/file/you/want/to/include In this file Asterisk will find the command, eg localnet=your LAN IP -- Daniel ___ --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] Looking for advice on cell carrier's default Un avaliable message
On Monday 05 December 2005 12:09, Colin Anderson wrote: Everything works 100%, except when the user shuts his cell phone off. When that happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played. Asterisk detects this as the call being answered and completes the call. Turn off voicemail on his cell phone, give out his DID instead of his cell #. Send an SMS to his cellphone when new voicemail is left. As far as Dial()ing his cell goes, use 'r' (this is exactly what it's designed for) so that when the carrier is saying The person you're calling is out of the calling area or has his phone off all the caller hears is ringing. I just described how I have my own system working and it seems to work just fine. :-) -A. ___ --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] Transfer/take call to/from other phone
Hi, To pick up another persons phone that is ringing dial '*8' followed by their extension. To do an attended transfer dial '*2' followed by the extension... Hope that helps Denny Schierz wrote: hi, Quoting Chuck Bunn [EMAIL PROTECTED]: Push the '#' key followed by the extension for a blind transfer. absolut perfect, thanks :-) .Is there also a shortcut, to take a phone call from other phones to me? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --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
solved (Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i)
I solved it by registering the phone in the sip.conf. ___ --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] Restore logging functionality...
Hi, I deleted the files and ran 'logger restart' - no dice, 'logger rotate' - no dice, 'reload' - no dice, 'restart gracefully' - no dice. Logs are not recreated??? Any other ideas Thanks Marco Supino wrote: The user running asterisk doesnt have permission to write on the files, delete them , and asterisk will recreate them as user asterisk, or chown them, or change them to 777 best of all, delete them! Marco. Chuck Bunn wrote: Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages' for asterisk. I recreated the files by 'touch' filename and I have gone into the Asterisk CLI and tried both 'logger restart' and 'logger rotate' but the logs still show nothing. I run 'logger show channels' and the output below shows up. I have recompiled Asterisk 1.2 and still the logs do not show up. I am getting data into the 'queue_log' and the 'events' logs however so I know logger is running. Any suggestions to fix this??? CLI output tomato*CLI logger show channels Channel Type StatusConfiguration --- --- tomato*CLI tomato*CLI Output from /var/log/asterisk directory [EMAIL PROTECTED] asterisk]# ls -la total 140 drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . drwxr-xr-x 11 root root 4096 Dec 4 04:03 .. drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-custom -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 17 06:41 event_log.1 -rw-r--r-- 1 root root 0 Nov 17 06:45 event_log.2 -rw-r--r-- 1 root root 0 Nov 18 06:38 event_log.3 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log *** 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
[Asterisk-Users] Asterisk Queues Tutorial updated...
Hello, Just a note to say the Asterisk Queues Tutorial at http://www.orderlyq.com/asteriskqueues.html has been updated to take account of changes in the 1.2.0 release. Anybody who has used our tutorial to create their queues, or uses queues and is thinking of upgrading, will probably find this new version useful. Comments feedback welcome - though message me privately please to avoid bugging the list Many thanks, Matt King Managing Director, Orderly Software Ltd. http://www.orderlyq.com - the world's most advanced queue system. P.S. You can also check out our new statistics package, OrderlyStats, at http://www.orderlyq.com/statistics.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] Re: Asterisk 1.2 problems ([EMAIL PROTECTED])
We are using firmware version 6.3. Dont we need a service agreement to get the latest drivers? We let ours expire since we werent having any problems. Isnt it also true that once you upgrade the firmware there is no way to revert to an earlier version? This is worrisome because we have heard of bad versions and do not want to upgrade without having a back out plan. Thanks, Tim What version firmware are you running on your Cisco Phones? We are running Asterisk 1.2 with the 7.4 firmware. The latest is 7.5 but there are some strange things that happen with this firmware. If I were you I would try a different firmware on the phones. Hope this helps. Jeremiah Help! I've encountered some problems with Asterisk that Iâm unable to solve. We have been running Asterisk version 1.0.9 for many months using a few local network connected Cisco 7960 phones as SIP clients. All our phones are currently internal so there is no NAT involved. We were not having any problems until last week when some strange issues started to crop up. I started experiencing calls that I initially believed were being dropped, but discovered that only one side of the conversation had dropped. The other party could hear me but I couldn't hear them. This seems to happen more often on longer calls but is not consistent. I am also seeing issues where incoming or local extension calls that are hung up by the originator before being answered will continue to ring the SIP phone. At the time the errors occur, the Asterisk console displays a variety of ...retrans_pkt: Maximum retries exceeded on call.. messages. I scoured the forums for an answer, found many reference s to these errors, tried every suggested fix that I could find, but none have resolved these problems. After working on the problem for several days, I finally built a new box and installed Asterisk 1.2 on it. Using this new 1.2 box I no longer see the Maximum retries exceeded on call warnings on the console but still experience the strange behavior. Unfortunately, the errors occur randomly so I am unable to reproduce the error on demand. I turned on SIP debugging and set console logging to debug and captured an instance of the problem with the hang up not being recognized. The details are below: I dial in from my cell phone. My Cisco phone begins to ring. I then hang up my cell phone. Asterisk acknowledges the hang up, but the Cisco phone continues to ring. After a minute or so, or if I pickup the phone, Asterisk display the following message That's odd... Got a response on a call we donât know about. Cseq 102 Cmd SIP/2.0 I've included a copy of the console output when this occurs that shows both the SIP message and the Asterisk debug output. Let me know if any more information would be of use and thanks in advance! The Cisco phone is on IP 192.168.2.203 The Asterisk switch is on IP 192.168.2.30 -- SIP read from 192.168.2.203:50237: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 192.168.2.30:5060;branch=z9hG4bK3dd277f1;rport From: JOHN DOE sip:[EMAIL PROTECTED];tag=as78389007 To: sip:[EMAIL PROTECTED]:5060;tag=001380df7eee002b0c2db83c-5ecedbb5 Call-ID: [EMAIL PROTECTED] Date: Fri, 02 Dec 2005 17:04:49 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Content-Length: 0 Dec 2 09:04:37 VERBOSE[3842] logger.c: --- (10 headers 0 lines)Dec 2 09:04:37 VERBOSE[3842] logger.c: --- (10 headers 0 lines)--- Dec 2 09:04:37 DEBUG[3842] chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 102 Cmd SIP/2.0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 problems
Thanks! It looks like you were right. We placed the phones and PBX on a minimal, physically separate network and have had no problems. We were using a 3com unmanaged switch but have ordered an HP managed switch with VLANs and VoIP QoS capabilities. We couldnt find anything about Shadow ping, is this an app? Is it useful? Also, this issue sounds like a good argument against the use of soft phones since you would be unable to segregate voice and data, right? Thanks, Tim On Fri, 2005-12-02 at 14:22, [EMAIL PROTECTED] wrote: Help! I've encountered some problems with Asterisk that IÂm unable to solve. We have been running Asterisk version 1.0.9 for many months using a few local network connected Cisco 7960 phones as SIP clients. All our phones are currently internal so there is no NAT involved. We were not having any problems until last week when some strange issues started to crop up. I started experiencing calls that I initially believed were being dropped, but discovered that only one side of the conversation had dropped. The other party could hear me but I couldn't hear them. This seems to happen more often on longer calls but is not consistent. I am also seeing issues where incoming or local extension calls that are hung up by the originator before being answered will continue to ring the SIP phone. At the time the errors occur, the Asterisk console displays a variety of ...retrans_pkt: Maximum retries exceeded on call.. messages. I scoured the forums for an answer, found many refere nce s to these errors, tried every suggested fix that I could find, but none have resolved these problems. After working on the problem for several days, I finally built a new box and installed Asterisk 1.2 on it. Using this new 1.2 box I no longer see the Maximum retries exceeded on call warnings on the console but still experience the strange behavior. Unfortunately, the errors occur randomly so I am unable to reproduce the error on demand. I turned on SIP debugging and set console logging to debug and captured an instance of the problem with the hang up not being recognized. The details are below: I dial in from my cell phone. My Cisco phone begins to ring. I then hang up my cell phone. Asterisk acknowledges the hang up, but the Cisco phone continues to ring. After a minute or so, or if I pickup the phone, Asterisk display the following message That's odd... Got a response on a call we donÂt know about. Cseq 102 Cmd SIP/2.0 I've included a copy of the console output when this occurs that shows both the SIP message and the Asterisk debug output. Odds are you have local network congestion -- Dropped packets or delayed packets. Try moving your phone and asterisk server to an isolated network switch - no other traffic (certainly no computers) - then test. If the problems go away, then update your virus scanners and check your computers. Good Luck Jon Carnes ___ --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] Looking for advice on cell carrier's default Un avaliable message
Turn off voicemail on his cell phone, give out his DID instead of his cell #. Send an SMS to his cellphone when new voicemail is left. That's what we do now. Works fine. As far as Dial()ing his cell goes, use 'r' (this is exactly what it's designed for) so that when the carrier is saying The person you're calling is out of the calling area or has his phone off all the caller hears is ringing. That appears to work *perfectly* but I don't get it. With the 'r' option on, how can Asterisk determine that the user has answered the phone as opposed to the carrier? Is it a signal that the carrier is sending? Anyway, thanks. Works like a hot damn. ___ --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] Panasonic DBS DISA
Hopefully, someone here has dealt with a Panasonic DBS in this way. I have put an Asterisk server in front of our Panasonic DBS phone system. The goal is to phase out our DBS, but during the transition, I still need to have asterisk extensions access some features of our Panasonic. The two features in question are paging though the Panasonic DBS and pickup of parked calls. The T1 card in my Panasonic sees Asterisk as a CO, but is also configured to send 56XX and 57XX directly out the T1, so I can call from system to system transparently. Also, (I have not decided yet) I may keep the Panasonic indefinitely just for paging and for the analog extensions for fax, etc. I assume that I have two options: 1. Use DISA in the Panasonic DBS and have an *9001 (Panasonic code for pickup park pos. 1) extension in Asterisk to dial into the Panasonic, log into DISA and dial *9001 in the Panasonic. Then do similar for other park positions and paging. I am having trouble figuring out DISA in the Panasonic. 2. Configure an analog station port on asterisk and connect it directly to an analog extension on the Panasonic to send these Panasonic codes. The catch here is that I only have so many analog extensions on the Panasonic and may not have one available. Also, I have no more slots in my Asterisk to put in an analog card to do this with. Also, I think that the iaxy, etc. can only be used as analog CO ports. Factoring the issues with above, the DISA over T1 would seem the best if I could get it to work. Has anyone here dealt with DISA on a Panasonic DBS? -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --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] Looking for advice on cell carrier's default Un avaliable message
On Monday 05 December 2005 13:39, Colin Anderson wrote: That appears to work *perfectly* but I don't get it. With the 'r' option on, how can Asterisk determine that the user has answered the phone as opposed to the carrier? Is it a signal that the carrier is sending? Anyway, thanks. Works like a hot damn. With the carrier voicemail turned off (not subscribed to) the carrier does not answer the line to say this person is out of the service area or has their phone off -- it's the same trick (early audio) used with digital lines to inform the caller of a problem without charging them for the privilege. -A. ___ --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] video phones
Anyone using any H.263+ video phones and want to relay their experiences? -Jonathan ___ --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] Anyone know anything about the new Linksys One product - does it use Asterisk?
Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? 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] DISA function
I had a problem with DTMF with DISA.. I am using a Sipura SPA 3000 for the line. I set the FXO port impedance (on the PSTN line tab) to 900 as advised by others and it worked. Having said that, I'm sure you will be using some other FXO adapter.. Just thought I'd tell. - Original Message - From: Richard Smith To: asterisk-users@lists.digium.com Sent: Monday, December 05, 2005 01:44 Subject: [Asterisk-Users] DISA function Hi all, I was wondering whether the DISA function on the latest asterisk 1.2 stable release actually works better than the other prior releases. Basically the [EMAIL PROTECTED] version 2.0 BETA 4 I'm using does not recognise the DTMF tones all the time and sometime when it does, it disconnects. Cheers, 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk?
No it does not user Asterisk. It is a proprietary system based around the Call Manager products. Linksys sells the system to a service provider who then offers the service to end users. Basically, LinksysOne is a means by which service providers can offer a hosted PBX solution. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, December 05, 2005 11:28 AM To: Asterisk - Users Subject: [Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk? Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? 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] Anyone know anything about the new Linksys One product - does it use Asterisk?
You can find more information at http://www.linksysone.com/ Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Chuck Bunn wrote: Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? 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
[Asterisk-Users] Error when compiling asterisk
Any help on this pleaseHi, I am getting this error when compiling asterisk `ls *.c`: unrecognized optionh -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ...GNU long options: --debug --dump-po-strings --dump-strings --help --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexpShell options: -irsD or -c command (invocation only) -abefhkmnptuvxBCHP or -o optionmake: *** [.depend] Error 2 Any ideas of what the problem might be. Thank you Appe l audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez le ici ! ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Biz mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-biz Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez le ici ! ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez le ici ! ___ --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] transfers from Polycom 501 involving Sipura 300 and asterisk 1.2
When transferring a call that came in on the Sipura and picked up by a Polycom 501 (sip 1.52), then transferred to another polycom using the transfer button on the polycom (havn't tried with the blind transfer from the polycom phone), then as soon as the transfer is complete (after pressing transfer again on the polycom) then the caller on the Sipura side can hear the new polycom caller, but the polycom cannot hear the sipura caller. This is all on a flat network, no nat, no gateways, between any of the points. If I change canreinvite=no for the sipura then everyting works fine. I'm assuming this is a bug in 1.2, but before I jump to conclusions I would like to know if anyone else has seen this? I did not yet have a chance to capture the output, but will do so if needed. Thank You ___ --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] Preventing incoming calls from ringing SIP lines
Hi We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2 But we want to prevent either direct incoming calls or calls from other extensions from ringing if the user is in another incoming call (i.e incoming into the extension), making an outgoing call or even checking their voicemail. In 1.0 the SetGroup and CheckGroup commands could do this but you have to build it into all parts of the dial plan. In 1.2 these do not exist and the Set(Group type commands with GotoIf are supposed to be used. But I still have not seen anywhere a full example of this. There is the call-limit setting in SIP - beautiful, works at the SIP level so easier than the dial plan. BUT with this you cannot do attended or blind transfers - not sensible. This must be a very common requirement, certainly is judging from the posts but in hours of searching I have not see the sort of complete solution which looks feasible. Thanks and sorry if I've missed it. Alternatively I'd be happy to use single line SIP softphones but cannot find one which feels good. TIA Paul R ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys SPA-941 DTMF failure with asterisk v.1.2
Been working on testing asterisk 1.2 before upgrading our production
systems from 1.0.x and have found a few issues. The one I am
working on
now involves DTMF failure with the following setup:
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)* SIP- *Global Crossing* (PSTN)
g711 with RFC 2833 out of band DTMF is used throughout the entire setup
from the Linksys to Global Crossing. Asterisk servers are using
asterisk SVN 1.2 from Friday. asteriskA is used as a SIP
registrar server for SIP devices to connect and asteriskB is used as a
gateway to our SIP provider.
In order to test DTMF at each stage, I set up the following so asterisk could playback which digits I entered:
; Test DTMF
exten = 123,1,Read(NUMBER)
exten = 123,2,SayDigits(${NUMBER})
exten = 123,3,Goto(1)
Here are the tests I ran and the results
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)* SIP- *Global Crossing* (PSTN)
Test Failed - poor DTMF accuracy
I then trying reverting asteriskB to version 1.0.x of asterisk and surprisingly, DTMF worked fine:
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.0)* SIP- *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
I then tried using a Cisco 7960 in place of the Linksys SPA-941 and all worked fine there as well:
*Cisco 7960* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)* SIP- *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
One would think the issue is with the SIP provider (Global Crossing)
but what makes it odd is that DTMF fails only when using the
Linksys and only when using version 1.2 of asterisk. So for now I am ruling out Global Crossing.
Any thoughts?
PS: Bug 5780 states that it is related to g729, not g711 which is in use here.
___
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Re: [Asterisk-Users] h323 vs oh323
Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] Linksys SPA-841 Missing Calls
Subject: RE: [Asterisk-Users] Linksys SPA-841 Missing Calls Dave Morrow [EMAIL PROTECTED] wrote: I've narrowed it down to the phones dislike for my older 3COM switch. I noticed on the weekend that when these missed calls occur, if I ping the phone, the first few packets are dropped..almost like it's gone to sleep.. We have had some network issues with our SPA-841's as well. We ended up having to take the phone off our standard network. Even though it was a completely switched network, we believe sufficient ARP broadcasts packets were being sent to the phones to slow them down. Our symptom was choppy or robotic sound similar to what you'd expect with high packet loss, accompanied by extremely high decode latency numbers on the System page. Even that wasn't enough: we needed higher quality switches than the cheap ones we expected to be able to use, to avoid other sound quality issues which continued to crop up. This is good evidence as to why they didn't put 2 ethernet ports on the phone: it would only make things worse if you shared the port with a PC or workstation, I'd expect. Alan ___ --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] h323 vs oh323
I am still having a non-solved problem with Oh323/h323 and checking Digium homepage after a long time, it looks like they need some dimes now to support me in this case. I have 46(2 T1) PSTN channels receiving calls through H323 protocol. With oh323, after 40 channels in use, It crashes due to some bug related to the limit of file handles. Even playing with some high values in /proc/sys/fs/file-max, didn't solve. With chan_h323, I don't have this problem but, I have this one: localhost*CLI show channels Channel Location State Application(Data) Zap/20-1 [EMAIL PROTECTED]:1 Up Bridged Call(H323/ip$a.b.c.d) 1 active channel 5 active calls I have only one active channel but 5 active calls?! Asterisk version 1.2.0 with H323 and the same pwlib/H323 libs recommended by the README. Checking the logs, I have tons of these errors: Dec 6 00:36:17 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:18 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:19 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:20 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:21 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:22 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! And this one too: Dec 6 00:36:18 WARNING[31530] channel.c: Prodding channel 'H323/ip$202.83.196.25:32791/31907' failed How to solve this problem? Isamar On Mon, 5 Dec 2005, David Waugh wrote: Prely subjective, but I first installed h323 and it worked. Somewhere along the line something happened and it no longer worked. Recompiling it etc seemed to have no effect. I then tried oh323 and it worked first time and has stayed working. I probably did soemthing wrong, but oh323 seems to work for me. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Innocent Evil Sent: 05 December 2005 14:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] h323 vs oh323 Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] Re: Linksys SPA-941 DTMF failure with asterisk v.1.2
One other piece of information that I just stumbled on while doing a packet capture which may explain the whole thing:
The Cisco packet shows the RTP event as this:
RFC 2833 RTP Event
Event ID: DTMF Pound # (11)
End of Event: True
Reserved: False
Volume: 10
Event Duration: 1600
The Linksys packet shows the following:
RFC 2833 RTP Event
Event ID: DTMF Pound # (11)
End of Event: True
Reserved: False
Volume: 0
Event Duration: 1760
Notice the volume setting in the Linksys packet. Could this be
the issue? I have changed every DTMF-related setting in the
Linksys that I can think of with no change in behavior.
What still doesn't make sense to me is that why would this not work
with asterisk 1.2 yet still work when used with asterisk 1.0.x?
On 12/5/05, tracinet [EMAIL PROTECTED] wrote:
Been working on testing asterisk 1.2 before upgrading our production
systems from 1.0.x and have found a few issues. The one I am
working on
now involves DTMF failure with the following setup:
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)* SIP- *Global Crossing* (PSTN)
g711 with RFC 2833 out of band DTMF is used throughout the entire setup
from the Linksys to Global Crossing. Asterisk servers are using
asterisk SVN 1.2 from Friday. asteriskA is used as a SIP
registrar server for SIP devices to connect and asteriskB is used as a
gateway to our SIP provider.
In order to test DTMF at each stage, I set up the following so asterisk could playback which digits I entered:
; Test DTMF
exten = 123,1,Read(NUMBER)
exten = 123,2,SayDigits(${NUMBER})
exten = 123,3,Goto(1)
Here are the tests I ran and the results
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)* SIP- *Global Crossing* (PSTN)
Test Failed - poor DTMF accuracy
I then trying reverting asteriskB to version 1.0.x of asterisk and surprisingly, DTMF worked fine:
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.0)* SIP- *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
I then tried using a Cisco 7960 in place of the Linksys SPA-941 and all worked fine there as well:
*Cisco 7960* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)* SIP- *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
One would think the issue is with the SIP provider (Global Crossing)
but what makes it odd is that DTMF fails only when using the
Linksys and only when using version 1.2 of asterisk. So for now I am ruling out Global Crossing.
Any thoughts?
PS: Bug 5780 states that it is related to g729, not g711 which is in use here.
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Re: [Asterisk-Users] Asterisk + WiFi Phones
I'm curious if anything new has been determined on this phone? Is it SIP compatible with Asterisk and, say, Broadvoice? I'm a little wary that this may be vaporware. The phone doesn't seem to be listed by the FCC. But, I would preorder one if it's Asterisk and Broadvoice compatibile. Phil PS- Contact us form on the viopsupply site seems to be broken? Just spins for me. Cory Andrews wrote: The F3000 is also a clamshell, flip type phone. I should be receiving an eval unit shortly and will post my findings after we work it over in the lab. Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Luki wrote: UTStarCom has the F3000 coming in December, which will have according to their spec * WEP (64 and 128 bit )/WPA/MD5 Auth * Handover/Roaming between different AP and SSID So what else is different compared to the F1000? The 1000 also does WEP 64/128 and WPA with the newest firmware. Not sure about MD5 auth, but SIP nonce/MD5 response certainly is implemented. Roaming kind of works, but could be improved. In one place I made it from 4th floor - elevator - lobby while on the phone and without any noticeable dropouts (ulaw codec). But the building was covered with access points, on average NetStumbler saw 6 at the same time. So it works, but not always. Don't get me wrong, the phone does have issues and in my opinion is not production quality, meaning it will freak out unexpectedly and only a reboot helps, which hardly ever happens to any Sipura adapters or phones. Hopefully the new 3.6 firmware performs better. Luki ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI indications.
Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk unallocated number but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16 bchan=1-10 alaw=1-10 loadzone=pl defaultzone=pl My /etc/asterisk/zapata.conf: [channels] language=en context=from-pstn switchtype=euroisdn signalling=pri_cpe pridialplan=local usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=no cancallforward=no callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no priindication=outofband group = 1 channel = 1-10 Regards, Adam Rybak ___ --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] Include a variable from another file in configfiles
amaury BOSSE wrote: Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? Look into the System() command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Best Switch for VOIP Applications
I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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] h323 vs oh323
I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works right out of box and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] Best Switch for VOIP Applications
What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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] Best Switch for VOIP Applications
I have a 24 port that is doing well for us. I will check out the LinkSys. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, December 05, 2005 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Best Switch for VOIP Applications What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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] h323 vs oh323
So, we have h323, oh323 and ooh323 I knew about h323 and oh323 but didn't know about ooh323. What is URL of ooh323, I want to know more about them. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 6 Dec 2005 09:16:05 +1100 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] h323 vs oh323 I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works right out of box and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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___ --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] Best Switch for VOIP Applications
Wiley Siler wrote: What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Does the SRW2024 support port mirroring? I was shopping around, but couldn't find any Linksys switch that support port mirroring. I ended with the DLINK DES-1226G which retails for a lot less than the SRW2024 (over here we can get it for US$300) and has VLAN (port-based or 802.1q) and port mirroring. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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] Best Switch for VOIP Applications
On 12/5/05, calvis [EMAIL PROTECTED] wrote: I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. We use the Fastiron workgroup swiches and really like them. Very solid but a tad expensive. Chris ___ --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] Include a variable from another file in configfiles
JP Carballo wrote: amaury BOSSE wrote: Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? Look into the System() command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System Oops, I missed the get a variable part. Your best bet is to use AGI. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Best Switch for VOIP Applications
On 14:42, Mon 05 Dec 05, snacktime wrote: On 12/5/05, calvis [EMAIL PROTECTED] wrote: I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. We use the Fastiron workgroup swiches and really like them. Very solid but a tad expensive. little expensive but also good are the cisco's. They play very nice with the 79XX series. Add PoE to that and you can really see why I like setups like that. I have no experience with the gbit line of cisco's though. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Best Switch for VOIP Applications
Cisco owns Linksys so they have some good features now. 64 VLANs, 8 port trunking groups, console port, 802.1p CoS support Four Quality of Service egress queues per port let you prioritize traffic via 802.1p. http://www1.linksys.com/products/product.asp?grid=35scid=40prid=673 This can be found for close to $400. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Monday, December 05, 2005 3:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best Switch for VOIP Applications Wiley Siler wrote: What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Does the SRW2024 support port mirroring? I was shopping around, but couldn't find any Linksys switch that support port mirroring. I ended with the DLINK DES-1226G which retails for a lot less than the SRW2024 (over here we can get it for US$300) and has VLAN (port-based or 802.1q) and port mirroring. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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 ___ --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] Grandstream NTP
All my BT101's and GXP2000's are failing NTP update. My NTP server is on my local LAN (and I've tried external ones), DNS is OK (and I've used IP address instead of DNS name). tcpdump on NTP server shows valid request, AND a valid response, yet the phones still display 02-01-1900. I have tried latest (and BETA firmware). Does anyone have any ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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] Messages button on a Polycom 501
Need a little help. Just set up an [EMAIL PROTECTED] box with 5 Polycom 501 phones. Everything works great except the messages button which when pressed results in asterisk responding Person at extension 102 is on the phone. Please leave a message after the tone. I have searched the web and several of the the asterisk mailing list archive pages - but I haven't had any luck. Anyone have a suggestion? ___ --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] hierarchical VoIP system
And about the protocol used to create this hierarchical network? Should I use SIP (routing between SERs) or should I use IAX (routing between Asterisks)? About ENUM, Isnt the managing of the ENUM tree going to be very complicated and heavy when we reach the millions of users? Joao Jan Saell wrote: Hi there! We have kind of the same setup but are using a few number of SER boxes for the on net calls - using enum for the lookup would be a great idea so that you can get the numbers to do sip calls and move over slowly. And for the central routing voip server make the routing use SIP redirects as the central server then can handle a lot of calls as its only doing the routing decisions. Best regards jan --On Wednesday, November 30, 2005 05:45:21 PM + Joao Pereira [EMAIL PROTECTED] wrote: Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt ___ --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 ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ --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] Best Switch for VOIP Applications
Michiel van Baak wrote: On 14:42, Mon 05 Dec 05, snacktime wrote: On 12/5/05, calvis [EMAIL PROTECTED] wrote: I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. We use the Fastiron workgroup swiches and really like them. Very solid but a tad expensive. little expensive but also good are the cisco's. They play very nice with the 79XX series. Add PoE to that and you can really see why I like setups like that. I have no experience with the gbit line of cisco's though. We don't need GigE. We use Cat 5505 and 5509 switches. Dirt cheap from eBay. ___ --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
