Re: [asterisk-users] 1.6.1.10 Music On Hold
Hi, I think it can be related to https://issues.asterisk.org/view.php?id=16268 Best regards, Santi 2009/11/24 Örn Arnarson o...@arnarson.net Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson o...@arnarson.net Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange beep when using VoiceMailMain application
Hi Martin,
Thanks for the answer!
2009/9/7 Martin asteriskl...@callthem.info
that's probably for ADSI phones ... chan_local confuses the VoiceMailMain
app
and you hear it ...
I'm experiencing this with different SIP phones and softphones.
Why do you need to call it via chan_local ? Can't
you do Macro or just
call VoiceMailMain directly ?
That's a good question. The reason is that we were experiencing problems
with some DECT phones using the g729 codec and accessing the voicemail. The
phones stopped playing media when they stopped receiving RTP packets for a
few seconds, and usually this would happen between the locutions of the
VoiceMailMain application. So the solution we thought of was to use the Page
application in order to play some background audio at the same time as the
Voicemail. Something like this:
exten = _X.,1,Page(Local/${ext...@voicemail-page
Local/backgro...@voicemail-page,dq)
[voicemail-page]
exten = _X.,1,VoiceMailMain(${ext...@mydomain.com exten...@mydomain.com)
exten = background,1,MusicOnHold()
This has worked pretty well except for this weird beep at the beginning of
the call. While figuring out what might be the problem I observed this
happened if I tried to call VoiceMailMain via chan local.
What do you think?
Best regards,
Santi
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[asterisk-users] Strange beep when using VoiceMailMain application
Hello,
I'm experiencing a weird problem when using the VoiceMailMain application.
If I use the application after dialing a Local channel, there's strange beep
just after asterisk answers the call and before the first locution. The
extensions.conf I'm using is:
Ruido extraño al llamar a la aplicación VoiceMailMain
[default]
exten = _X.,1,Dial(Local/${ext...@test)
[test]
exten = _X.,1,VoiceMailMain(${ext...@mydomain.com exten...@mydomain.com)
On the other hand, if i use the application directly, this beep doesn't
appear.
The version I'm using is 1.4.26.1.
Does anybody know what might cause this?
Thanks. Best regards,
Santi
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Re: [asterisk-users] Versions of Asterisk 1.6
Hi David, Is T.38 Fax supported on both? I can tell you that I've been having problems with various version of Cisco IOS and T.38 on asterisk. I had a stable configuration fax-wise, but I had to upgrade the IOS because of a Cisco bug, and my T.38 has never been the same since. It's hard to blame asterisk for that problem. In fact, if you read through the T.38 bugs in Cisco IOS release notes it makes asterisk T.38 look solid by comparison. If downgrading didn't make my router freeze I'd downgrade the IOS. We are also having problems of interoperability between asterisk and CISCO. What version of the IOS was working for you? Thanks, Santi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no sdp or contact replacement using externip
I'm not sure if your problem is addressed by this: https://issues.asterisk.org/view.php?id=14546 . If that's the case it was solved in version 1.4.25 Best regards, Santi 2009/6/16 Ricardo Martins rpopp...@gmail.com Yes Gordon. I'm using nat=yes and I don't have an ALG enabled router/firewall. I used the sip debug output on the asterisk(s) and could see the sdp headers as they were gererated by asterisk, with the wrong (internal) address on it. Asterisk is sending the audio to the correct way, the public IP of client side NAT. But the client is sending it to the wrong address, the private IP of asterisk side NAT. Rgrs, Ricardo. Gordon Henderson escreveu: On Tue, 16 Jun 2009, Ricardo Martins wrote: Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. I have dozens of boxes doing it this way. All just work. Have you nat=yes in there too? Also you did port-forward from the router to the box as well, didn't you? Often the router will have a broke SIP ALG which will get in the way too. Turn it off if you can. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems receiving some faxes in T.38
Hi David, That's very similar to a setup I made. And I was troubleshooting similar problems. Let me ask you a question: Are you quite confident that the inbound faxes that fail are going to succeed on an ordinary fax machine? At least I'm sure of a couple of calling numbers that I know are real faxes that work. There are others, I suspect, are not really good faxes. In my case I was able to crank through my logs, and trace that the failing calls were people who were calling a fax line by mistake, or wardialers, or clients with lousy fax configurations where those faxes also fail to our 'real' fax machines. When we stopped counting the 'never going to work anyway' faxes in our fax success calculations we had nearly perfect success rates. And here's my debugging tip. Pick a number that always fails, change the Cisco dialpeer to send those as ordinary audio fax passthrough, no t.38, use asterisk with monitor to record them, and watch whether they ever succeed. I'm willing to be my two cents that they don't. Thanks for the tip, I'll try this in order to figure out which are real and which are not. Best regards, Santiago Gimeno ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems receiving some faxes in T.38
Hello,
We have been working with the ReceiveFax application for some weeks now in
order to receive faxes in T.38 and it works fairly well, but there are some
faxes that for some reason we are not able to receive correctly.
The asterisk version we are using is 1.6.0.6 with spandsp-0.0.5pre4 and the
asterisk machine is behind a CISCO mediaGW to be able to communicate with
the PSTN.
The SIP call flows are different between the faxes we receive correctly and
the ones that fail.
In the case of successfully received faxes, after establishing the audio
session between de CISCO and Asterisk, CISCO sends a re-INVITE with the T.38
SDP. The T.38 setup succeeds.
CISCOAsterisk
| |
| |
| |
|INVITE (SDP alaw)
|-|
|200 OK (SDP alaw)
|-|
|ACK |
|-|
|Re-INVITE (SDP T.38)
|-|
|200 OK (SDP T.38)
|-|
|ACK |
|-|
| |
|..|
|T.38 |
|..|
|[t.38]no signal
|-|
|[t.38]no signal
|-|
|[t.38]CED |
|-|
|[t.38]V21-preamble
|-|
| |
| |
On the other hand, with some faxes, the re-INVITE is sent by Asterisk and it
looks that there is something wrong in the T.38 setup that makes the fax
reception fail after the permitted retries. The FAXERROR variable is set to:
Disconnected after permitted retries.
What I can see from the traces is that it gets to a point that asterisk is
sending T.38 data to the CISCO but the CISCO doesn't answer.
CISCOAsterisk
| |
| |
| |
|INVITE (SDP alaw)
|-|
|200 OK (SDP alaw)
|-|
|ACK |
|-|
|Re-INVITE (SDP T.38)
|-|
|200 OK (SDP T.38)
|-|
|ACK |
|-|
| |
|..|
|T.38 |
|..|
|[t.38]no signal
|-|
|[t.38]no signal
|-|
|[t.38]CED |
|-|
|[t.38]no signal
|-|
|[t.38]V21-preamble
|-|
|[t.38]hdlc|
|-|
|[t.38]no signal
|-|
|[t.38]V21-preamble
|-|
|[t.38]hdlc|
|-|
|[t.38]no signal
|-|
|[t.38]V21-preamble
|-|
|[t.38]hdlc|
|-|
|[t.38]no signal
|-|
|[t.38]V21-preamble
|-|
|[t.38]hdlc|
|-|
|[t.38]no signal
|-|
|[t.38]V21-preamble
|-|
|[t.38]DCN |
|-|
|BYE |
|-|
|200 OK|
|-|
Any idea of what might be happening?
Thanks in advance. Best regards,
Santiago Gimeno
The relevant information in the asterisk configuration files is:
extensions.conf
[fax-in]
exten = 9,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,n,Answer()
exten = 9,n,Wait(3)
exten = 9,n,ReceiveFax(${INCOMING_FAXFILE})
sip.conf
[general]
canreinvite=no
t38pt_udptl=yes
disallow=all
allow=alaw
context=fax-in
The CISCO peer configuration:
dial-peer voice 6 voip
destination-pattern 88T
session protocol sipv2
session target ipv4:10.100.0.51
session transport udp
dtmf-relay rtp-nte
codec g711alaw
fax-relay ecm disable
fax nsf 00
fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback none
no vad
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Re: [asterisk-users] Problems receiving some faxes in T.38
in span_message: FLOW T.38 Tx10: (0) data v21/hdlc-data + 1 byte(s) May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx10: IFP c0 01 80 00 00 01 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx11: (0) data v21/hdlc-data + 1 byte(s) May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx11: IFP c0 01 80 00 00 89 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx12: (0) data v21/hdlc-data + 1 byte(s) May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx12: IFP c0 01 80 00 00 01 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx13: (0) data v21/hdlc-data + 1 byte(s) May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx13: IFP c0 01 80 00 00 01 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx14: (0) data v21/hdlc-data + 1 byte(s) May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx14: IFP c0 01 80 00 00 01 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx15: (0) data v21/hdlc-data + 1 byte(s) May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx15: IFP c0 01 80 00 00 18 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx16: (0) data v21/hdlc-fcs-OK-sig-end + 0 byte(s) May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx16: IFP c0 01 40 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38 Tx17: indicator no-signal May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38T Set rx type 4 May 20 16:40:04 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38T Set tx type 0 May 20 16:40:08 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 May 20 16:40:08 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX May 20 16:40:08 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38T Set rx type 0 May 20 16:40:08 server asterisk[2456]: DEBUG[25323]: app_fax.c:129 in span_message: FLOW T.38T Set tx type 4 Why turn off ECM? Turned it on. Best regards, Santiago Gimeno ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error in ReceiveFax with T.38 -- Asterisk 1.6.0.7-rc2
Hi David,
Thanks for the answer!
By using the h extension now I'm able to check that the Faxes are sent
successfully.
Best regards,
Santi
On Fri, Mar 27, 2009 at 4:42 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Mar 24, 2009 at 1:57 PM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Hello,
The NoOp output was not displayed at all. I'm assuming because of the
failure in the ReceiveFax application. In fact, the verbose output
Try changing
[fax-in]
exten = 9,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,n,Answer()
exten = 9,n,Wait(3)
exten = 9,n,ReceiveFax(${INCOMING_FAXFILE})
exten = 9,n,NoOp(FAXSTATUS: ${FAXSTATUS}, FAXERROR: ${FAXERROR},
FAXMODE: ${FAXMODE}, REMOTESTATIONID: ${REMOTESTATIONID}, FAXPAGES:
${FAXPAGES}, FAXBITRATE: ${FAXBITRATE}, FAXRESOLUTION:
${FAXRESOLUTION})
to
[fax-in]
exten = 9,s,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,s,n,Answer()
exten = 9,s,n,Wait(3)
exten = 9,s,n,ReceiveFax(${INCOMING_FAXFILE})
exten = 9,h,1,NoOp(FAXSTATUS: ${FAXSTATUS}, FAXERROR: ${FAXERROR},
FAXMODE: ${FAXMODE}, REMOTESTATIONID: ${REMOTESTATIONID}, FAXPAGES:
${FAXPAGES}, FAXBITRATE: ${FAXBITRATE}, FAXRESOLUTION:
${FAXRESOLUTION})
exten = 9,h,HangUp
You are correct that when receivefax completes you are now in hangup context.
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[asterisk-users] Error in ReceiveFax with T.38 -- Asterisk 1.6.0.7-rc2
Hello, In my scenario, the asterisk machine is installed behind a CISCO mediaGW in order to be able communicate with the PSTN. Asterisk is configured to use T.38 to send and receive faxes. I'm trying to receive a fax from a fax machine located in the PSTN. Apparently everything goes well: the fax machine says the transmission was successfully completed, and the fax file is successfully stored by asterisk. The problem is that I receive this error: WARNING[12229]: app_fax.c:650 in transmit: Transmission error and the ReceiveFax function ends abruptly. The log file with debug set to 4 is this: asterisk[12127]: DEBUG[12229]: app_fax.c:166 in phase_e_handler: Fax phase E handler. result=0 asterisk[12127]: DEBUG[12229]: app_fax.c:202 in phase_e_handler: Fax transmitted successfully. asterisk[12127]: DEBUG[12229]: app_fax.c:203 in phase_e_handler: Remote station ID: 0034913121867 asterisk[12127]: DEBUG[12229]: app_fax.c:204 in phase_e_handler: Pages transferred: 1 asterisk[12127]: DEBUG[12229]: app_fax.c:205 in phase_e_handler: Image resolution: 8031 x 3850 asterisk[12127]: DEBUG[12229]: app_fax.c:206 in phase_e_handler: Transfer Rate: 9600 asterisk[12127]: DEBUG[12229]: app_fax.c:130 in span_message: FLOW T.30 Changing from state 2 to 32 asterisk[12127]: DEBUG[12229]: app_fax.c:130 in span_message: FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED asterisk[12127]: DEBUG[12229]: app_fax.c:130 in span_message: FLOW FAX Set rx type 8 asterisk[12127]: DEBUG[12229]: app_fax.c:130 in span_message: FLOW FAX FAX exchange complete asterisk[12127]: DEBUG[12229]: app_fax.c:130 in span_message: FLOW FAX Set tx type 8 asterisk[12127]: DEBUG[12229]: app_fax.c:130 in span_message: FLOW FAX FAX exchange complete asterisk[12127]: DEBUG[12229]: channel.c:3470 in set_format: Set channel SIP/192.168.0.253-081b9c18 to write format alaw asterisk[12127]: DEBUG[12229]: channel.c:3470 in set_format: Set channel SIP/192.168.0.253-081b9c18 to read format alaw asterisk[12127]: WARNING[12229]: app_fax.c:650 in transmit: Transmission error Any idea of what might be happening? Thank you in advance, Santi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error in ReceiveFax with T.38 -- Asterisk 1.6.0.7-rc2
Sorry about that, I forgot to post them:
-extension.conf:
[fax-in]
exten = 9,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,n,Answer()
exten = 9,n,Wait(3)
exten = 9,n,ReceiveFax(${INCOMING_FAXFILE})
exten = 9,n,NoOp(FAXSTATUS: ${FAXSTATUS}, FAXERROR: ${FAXERROR},
FAXMODE: ${FAXMODE}, REMOTESTATIONID: ${REMOTESTATIONID}, FAXPAGES:
${FAXPAGES}, FAXBITRATE: ${FAXBITRATE}, FAXRESOLUTION:
${FAXRESOLUTION})
-sip.conf:
[general]
bindport=5060
bindaddr=192.168.222.160
domain=192.168.222.160
type=friend
canreinvite=no
t38pt_udptl=yes
disallow=all
allow=alaw
context=fax-in
the cisco peer configuration:
dial-peer voice 2 voip
destination-pattern 9T
codec g711alaw
session protocol sipv2
session target dns:barik-pstn.com
session transport udp
dtmf-relay rtp-nte
fax nsf 00
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
Best regards,
Santi
On Tue, Mar 24, 2009 at 2:47 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Mar 24, 2009 at 9:21 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
WARNING[12229]: app_fax.c:650 in transmit: Transmission error
and the ReceiveFax function ends abruptly.
That doesn't really help, other than that it seems your arrangement
defaulted to voice rather than using T.38
How about pasting in your dialplan and your Cisco config for this dialpeer?
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Re: [asterisk-users] Error in ReceiveFax with T.38 -- Asterisk 1.6.0.7-rc2
in __ast_pbx_run: Spawn extension (demo,9,4) exited non-zero on 'SIP/192.168.0.253-b7a96b70' asterisk[12127]: VERBOSE[15248]: == Spawn extension (demo, 9, 4) exited non-zero on 'SIP/192.168.0.253-b7a96b70' asterisk[12127]: DEBUG[15248]: channel.c:1560 in ast_softhangup_nolock: Soft-Hanging up channel 'SIP/192.168.0.253-b7a96b70' asterisk[12127]: DEBUG[15248]: channel.c:1653 in ast_hangup: Hanging up channel 'SIP/192.168.0.253-b7a96b70' asterisk[12127]: DEBUG[15248]: chan_sip.c:4914 in sip_hangup: Hangup call SIP/192.168.0.253-b7a96b70, SIP callid 77c43b95-17d211de-8d93ab80-f420c...@192.168.0.253 asterisk[12127]: DEBUG[15248]: devicestate.c:450 in ast_devstate_changed_literal: Notification of state change to be queued on device/channel SIP/192.168.0.253 Thanks. Best regards, Santi On Tue, Mar 24, 2009 at 6:25 PM, David Backeberg dbackeb...@gmail.com wrote: On Tue, Mar 24, 2009 at 11:33 AM, Santiago Gimeno santiago.gim...@gmail.com wrote: Sorry about that, I forgot to post them: That all looks pretty good. So in your original post, you clipped it off before you got all the useful no-op output at the end. I'm also assuming your file was empty? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callerid charset problems
Hi, I'm having problems when the callerid of a user defined in the sip.conf contains special characters such as: ñ, á, é, í, ó , etc. The strange thing is that these characters are displayed correctly in the dialplan by using the sip show peer command, but if this user makes a call, these characters are not displayed correcly in the SIP message. Any ideas of what might be happening? Thank you in advance. Regards, Santi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Silence suppression problem with DECT phones and g729 codec
Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send RTP with silence, but stop sending them and I think the problems is that the DECT phones are having problems with that. To check that this was the problem I have implemented a simple dialplan exten = *91,1,Set(CHANNEL(language)=es) exten = *91,n,Answer() exten = *91,n,Wait(4) exten = *91,n,Playback(vm-tmpexists) exten = *91,n,Wait(4) exten = *91,n,Playback(vm-tomakecall) exten = *91,n,Wait(4) exten = *91,n,Playback(vm-goodbye) exten = *91,n,Hangup ...and I have verified that if there is a pause between the playbacks the problem occurs, otherwise the audio is played correctly by the DECT phones I know it looks like a problem with the phones but, is there a way to configure asterisk so it sends RTP during silent periods? Thanks. Best regards, Santi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silence suppression problem with DECT phones and g729 codec
Yes, I had already tried that and it didn't work. Asterisk doesn't send any RTP. Regards, Santi On Fri, Mar 13, 2009 at 11:06 AM, Steve Howes st...@geekinter.net wrote: On 13 Mar 2009, at 09:51, Santiago Gimeno wrote: I know it looks like a problem with the phones but, is there a way to configure asterisk so it sends RTP during silent periods? Asterisk.conf transmit_silence_during_record = yes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
Thanks for the responses. I have solved the problem by using a different tiff generator. I used the gs command: # gs -q -sDEVICE=tiffg3 -dSAFER -dNOPAUSE -sOutputFile=test.tif test.pdf Best regards, Santi On Thu, Mar 12, 2009 at 3:30 PM, David Backeberg dbackeb...@gmail.comwrote: On Wed, Mar 11, 2009 at 7:32 AM, Santiago Gimeno santiago.gim...@gmail.com wrote: I finally solved the issue by changing the resolution and the width of the TIFF file to one that is accepted by the fax standard. In my case I changed to a resolution of 96x96 and a width of 1728. Now I am able to send faxes, but something weird is happening, the fax received in the fax-machine has the black and white colours inverted. Any ideas why this could be happening? The way I got my tiff file for testing was to use ReceiveFax to make a tiff from an inbound fax. I then used that tiff outbound for testing outbound faxing. Something you might want to consider doing? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
I finally solved the issue by changing the resolution and the width of the TIFF file to one that is accepted by the fax standard. In my case I changed to a resolution of 96x96 and a width of 1728. Now I am able to send faxes, but something weird is happening, the fax received in the fax-machine has the black and white colours inverted. Any ideas why this could be happening? Best regards, Santi On Tue, Mar 10, 2009 at 6:53 PM, Santiago Gimeno santiago.gim...@gmail.comwrote: Thanks for the tip. Sadly, it didn't work. I keep getting the same error: [Mar 10 18:49:48] WARNING[18855]: app_fax.c:176 phase_e_handler: Error transmitting fax. result=11: Far end cannot receive at the resolution of the image. regards, Santi On Tue, Mar 10, 2009 at 6:36 PM, Matthew Fredrickson cres...@digium.comwrote: Santiago Gimeno wrote: Hello, Thanks everybody for the answers. Could be. Would you post the Cisco config relevant to this? dial-peer voice 5 voip description ** ** preference 1 destination-pattern 1… voice-class codec 1 session protocol sipv2 session target ipv4:1.1.1.1 session transport udp dtmf-relay rtp-nte fax-relay ecm disable I think, that at least if you're using T.38, you may want to try enabling ECM. ECM can cause significant problems in a high-packet loss, non-T.38 environment, but I would think that in a T.38 environment, if you can keep ECM enabled, that would be a good thing. Matthew Fredrickson Digium, Inc. fax nsf 00 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw no vad And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. I have removed the T38CALL variable and it looks better but it still doesn't work. Now asterisk sends an initial INVITE with audio media in the SDP. The CISCO accepts this call after contacting the fax-machine. Then the CISCO sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE. But finally the fax transmission fails and the asterisk verbose trace is: *CLI -- Attempting call on SIP/080913216...@outbound-calls for 22...@fax-out:1 (Retry 1) == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Channel SIP/outbound-calls-0822aae8 was answered. == Starting SIP/outbound-calls-0822aae8 at fax-out,2,1 failed so falling back to exten 's' -- Executing [...@fax-out:1] Set(SIP/outbound-calls-0822aae8, FAXFILE=/root/santi/fax/prueba.tif) in new stack -- Executing [...@fax-out:2] SIPDtmfMode(SIP/outbound-calls-0822aae8, inband) in new stack -- Executing [...@fax-out:3] SendFAX(SIP/outbound-calls-0822aae8, /root/santi/fax/prueba.tif) in new stack [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error transmitting fax. result=11: Far end cannot receive at the resolution of the image. [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission error == Spawn extension (fax-out, s, 3) exited non-zero on 'SIP/outbound-calls-0822aae8' Any ideas? Thanks. Best regards, Santi On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: - Santiago Gimeno santiago.gim...@gmail.com mailto:santiago.gim...@gmail.com wrote: **The call-file I'm using is: Channel: SIP/08099...@outbound- calls MaxRetries: 3 WaitTime: 30 Set: LOCALSTATIONID=2 Set: LOCALHEADERINFO=T38 fax Set: T38CALL=1 Set: T38TXDETECT=yes CallerID: 2 Context: fax-out Extension: 2 priority:1 And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com www.asterisk.org http://www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
[asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
Hello,
I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly
with a CISCO mediaGW in order to send faxes to the PSTN using T.38.
When Asterisk sends the initial INVITE containing the T.38 media offer in
the SDP, the CISCO answers with a 488 Not Acceptable Media.
Apparently, it looks like a configuration problem in the CISCO, but I have
tested the CISCO with the Zoiper client and it successfully sends faxes. The
only difference I have noticed between the Asterisk and Zoiper is that
whereas the Asterisk sends the T.38 SDP information in the initial INVITE,
Zoiper establishes a voice call first and then re-negotiates(with a
re-INVITE) the session in order to send the T.38 media.
Is it possible to make Asterisk work like this? or is this a problem in the
configuration of the CISCO? Any ideas?
Thanks in advance.
Regards,
Santi
**The call-file I'm using is:
Channel: SIP/08099...@outbound-calls
MaxRetries: 3
WaitTime: 30
Set: LOCALSTATIONID=2
Set: LOCALHEADERINFO=T38 fax
Set: T38CALL=1
Set: T38TXDETECT=yes
CallerID: 2
Context: fax-out
Extension: 2
priority:1
My sip.conf file is:
sip.conf
[general]
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=192.168.222.160; IP address to bind to (0.0.0.0 binds to
all)
domain=192.168.222.160 ; Add IP address as local domain
t38pt_udptl=yes
[outbound-calls]
type=friend
context=openser
allow=all
;dtmfmode=info
host=10.100.222.201
insecure=very
canreinvite=no
pedantic=no
call-limit=10
The extensions.conf file
[fax-out]
exten =s,1,Set(FAXFILE=/root/santi/fax/prueba.tif)
exten =s,n,SipDTMFMode(inband)
exten =s,n,SendFax(${FAXFILE})
exten =s,n,Hangup
The SIP trace is:
INVITE sip:080...@10.100.222.201 sip%3a080...@10.100.222.201SIP/2.0
Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK306b777c;rport
Max-Forwards: 70
From: 2 sip:22...@192.168.222.160 sip%3a22...@192.168.222.160
;tag=as43e12927
To: sip:080...@10.100.222.201 sip%3a080...@10.100.222.201
Contact: sip:22...@192.168.222.160 sip%3a22...@192.168.222.160
Call-ID: 4f9fb8387458a3c6205e2c4467e48...@192.168.222.160
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Date: Tue, 10 Mar 2009 11:29:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 525135648 525135648 IN IP4 192.168.222.160
s=Asterisk PBX 1.6.0.6
c=IN IP4 192.168.222.160
t=0 0
m=image 4222 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC
#
U +0.015757 10.100.222.201:5060 - 192.168.222.160:5060
SIP/2.0 488 Not Acceptable Media
Reason: Q.850;cause=65
Date: Tue, 10 Mar 2009 11:29:18 GMT
From: 2 sip:22...@192.168.222.160 sip%3a22...@192.168.222.160
;tag=as43e12927
Allow-Events: telephone-event
Content-Length: 0
To: sip:080...@10.100.222.201 sip%3a080...@10.100.222.201
;tag=417D2718-582
Call-ID: 4f9fb8387458a3c6205e2c4467e48...@192.168.222.160
Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK306b777c;rport
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
#
U +0.000164 192.168.222.160:5060 - 10.100.222.201:5060
ACK sip:080...@10.100.222.201 sip%3a080...@10.100.222.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK306b777c;rport
Max-Forwards: 70
From: 2 sip:22...@192.168.222.160 sip%3a22...@192.168.222.160
;tag=as43e12927
To: sip:080...@10.100.222.201 sip%3a080...@10.100.222.201
;tag=417D2718-582
Contact: sip:22...@192.168.222.160 sip%3a22...@192.168.222.160
Call-ID: 4f9fb8387458a3c6205e2c4467e48...@192.168.222.160
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.6
Content-Length: 0
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Re: [asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
Hello, Thanks everybody for the answers. Could be. Would you post the Cisco config relevant to this? dial-peer voice 5 voip description ** ** preference 1 destination-pattern 1… voice-class codec 1 session protocol sipv2 session target ipv4:1.1.1.1 session transport udp dtmf-relay rtp-nte fax-relay ecm disable fax nsf 00 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw no vad And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. I have removed the T38CALL variable and it looks better but it still doesn't work. Now asterisk sends an initial INVITE with audio media in the SDP. The CISCO accepts this call after contacting the fax-machine. Then the CISCO sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE. But finally the fax transmission fails and the asterisk verbose trace is: *CLI -- Attempting call on SIP/080913216...@outbound-calls for 22...@fax-out:1 (Retry 1) == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Channel SIP/outbound-calls-0822aae8 was answered. == Starting SIP/outbound-calls-0822aae8 at fax-out,2,1 failed so falling back to exten 's' -- Executing [...@fax-out:1] Set(SIP/outbound-calls-0822aae8, FAXFILE=/root/santi/fax/prueba.tif) in new stack -- Executing [...@fax-out:2] SIPDtmfMode(SIP/outbound-calls-0822aae8, inband) in new stack -- Executing [...@fax-out:3] SendFAX(SIP/outbound-calls-0822aae8, /root/santi/fax/prueba.tif) in new stack [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error transmitting fax. result=11: Far end cannot receive at the resolution of the image. [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission error == Spawn extension (fax-out, s, 3) exited non-zero on 'SIP/outbound-calls-0822aae8' Any ideas? Thanks. Best regards, Santi On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp jc...@digium.com wrote: - Santiago Gimeno santiago.gim...@gmail.com wrote: **The call-file I'm using is: Channel: SIP/08099...@outbound- calls MaxRetries: 3 WaitTime: 30 Set: LOCALSTATIONID=2 Set: LOCALHEADERINFO=T38 fax Set: T38CALL=1 Set: T38TXDETECT=yes CallerID: 2 Context: fax-out Extension: 2 priority:1 And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
Thanks for the tip. Sadly, it didn't work. I keep getting the same error: [Mar 10 18:49:48] WARNING[18855]: app_fax.c:176 phase_e_handler: Error transmitting fax. result=11: Far end cannot receive at the resolution of the image. regards, Santi On Tue, Mar 10, 2009 at 6:36 PM, Matthew Fredrickson cres...@digium.comwrote: Santiago Gimeno wrote: Hello, Thanks everybody for the answers. Could be. Would you post the Cisco config relevant to this? dial-peer voice 5 voip description ** ** preference 1 destination-pattern 1… voice-class codec 1 session protocol sipv2 session target ipv4:1.1.1.1 session transport udp dtmf-relay rtp-nte fax-relay ecm disable I think, that at least if you're using T.38, you may want to try enabling ECM. ECM can cause significant problems in a high-packet loss, non-T.38 environment, but I would think that in a T.38 environment, if you can keep ECM enabled, that would be a good thing. Matthew Fredrickson Digium, Inc. fax nsf 00 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw no vad And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. I have removed the T38CALL variable and it looks better but it still doesn't work. Now asterisk sends an initial INVITE with audio media in the SDP. The CISCO accepts this call after contacting the fax-machine. Then the CISCO sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE. But finally the fax transmission fails and the asterisk verbose trace is: *CLI -- Attempting call on SIP/080913216...@outbound-calls for 22...@fax-out:1 (Retry 1) == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Channel SIP/outbound-calls-0822aae8 was answered. == Starting SIP/outbound-calls-0822aae8 at fax-out,2,1 failed so falling back to exten 's' -- Executing [...@fax-out:1] Set(SIP/outbound-calls-0822aae8, FAXFILE=/root/santi/fax/prueba.tif) in new stack -- Executing [...@fax-out:2] SIPDtmfMode(SIP/outbound-calls-0822aae8, inband) in new stack -- Executing [...@fax-out:3] SendFAX(SIP/outbound-calls-0822aae8, /root/santi/fax/prueba.tif) in new stack [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error transmitting fax. result=11: Far end cannot receive at the resolution of the image. [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission error == Spawn extension (fax-out, s, 3) exited non-zero on 'SIP/outbound-calls-0822aae8' Any ideas? Thanks. Best regards, Santi On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: - Santiago Gimeno santiago.gim...@gmail.com mailto:santiago.gim...@gmail.com wrote: **The call-file I'm using is: Channel: SIP/08099...@outbound- calls MaxRetries: 3 WaitTime: 30 Set: LOCALSTATIONID=2 Set: LOCALHEADERINFO=T38 fax Set: T38CALL=1 Set: T38TXDETECT=yes CallerID: 2 Context: fax-out Extension: 2 priority:1 And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com www.asterisk.org http://www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP dialog matching problem? (1.4.23.1)
On Fri, Mar 6, 2009 at 3:03 PM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Santiago Gimeno schrieb: Hello, Thanks for the reply. Yes, I'm using pedantic=yes. I will report this asap. One more thing that I have observed and might be also related to this issue. The scenario is the same as the one I described in the previous mail, but in this case, the SIP Phone that receives the 302 generates a new INVITE to the new address with exactly the same dialog information as the initial INVITE: call-id, from-tag and to-tag. This is wrong. This is definitely a new dialog, thus dialog-ids should change. Further, the request must not have a totag. Yes, you're right in that there must not be to-tag. In fact the INVITE doesn't have a to-tag. (I think this is legal as stated in the RFC 3261-8.1.3.4: /It is RECOMMENDED that the UAC reuse the same To, From, and Call-ID used in the original redirected I guess they mean the From URI and the To URI, but not the tags. I'm not sure about that. If you look at RFC 3665-3.6, there is a simple example of a call redirection in which the 2nd INVITE has the same Call-ID and from-tag but the CSeq numbers are different. The question is: should Asterisk reject the 2nd INVITE for having the same dialog id (call-id and from-tag) as the 1st INVITE even though that dialog is alreade in the TERMINATED state? best regards, Santi klaus request, but the UAC MAY also choose to update the Call-ID header field value for new requests, for example./). Asterisk answers to this INVITE with a 503 Unavailable because it matches with the previous dialog. I'm not sure if this is how Asterisk should behave, or it should allow the call to progress as the previous dialog is already in the TERMINATED state. What do you think? Best regards, Santi 2009/3/4 Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at Hi! Actually I would consider this as a bug, thus you should report it at bugs.digium.com http://bugs.digium.com. Are you using pedantic=yes (sip.conf)? If not, it would be interesting if the pedantic mode has the same problem. regards klaus Santiago Gimeno schrieb: Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER AsteriskPhoneB PhoneC | | | | | | | | | | | | | | | |INVITE B | | | | |-| | | | | |INVITE B | | | | |-| | | | | |INVITE B | | | | |-| | | | |486 Busy Here | | | | |-| | | | |ACK | | | | |-| | | |486 Busy Here | | | | |-| | | | |ACK | | | | |-| | | |302 MOVED (to C) | | | |-| | | | |ACK | | | | |-| | | | |INVITE C | | | | |-| | | | | |INVITE C | | | | |-| | | | |503 Unavailable | | | |-| | | | |ACK | | | | |-| | | |503 Unavailable | | | |-| | | | |ACK
Re: [asterisk-users] SIP dialog matching problem? (1.4.23.1)
Hello, Thanks for the reply. Yes, I'm using pedantic=yes. I will report this asap. One more thing that I have observed and might be also related to this issue. The scenario is the same as the one I described in the previous mail, but in this case, the SIP Phone that receives the 302 generates a new INVITE to the new address with exactly the same dialog information as the initial INVITE: call-id, from-tag and to-tag. (I think this is legal as stated in the RFC 3261-8.1.3.4: *It is RECOMMENDED that the UAC reuse the same To, From, and Call-ID used in the original redirected request, but the UAC MAY also choose to update the Call-ID header field value for new requests, for example.*). Asterisk answers to this INVITE with a 503 Unavailable because it matches with the previous dialog. I'm not sure if this is how Asterisk should behave, or it should allow the call to progress as the previous dialog is already in the TERMINATED state. What do you think? Best regards, Santi 2009/3/4 Klaus Darilion klaus.mailingli...@pernau.at Hi! Actually I would consider this as a bug, thus you should report it at bugs.digium.com. Are you using pedantic=yes (sip.conf)? If not, it would be interesting if the pedantic mode has the same problem. regards klaus Santiago Gimeno schrieb: Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER AsteriskPhoneB PhoneC | | | | | | | | | | | | | | | |INVITE B | | | | |-| | | | | |INVITE B | | | | |-| | | | | |INVITE B | | | | |-| | | | |486 Busy Here | | | | |-| | | | |ACK | | | | |-| | | |486 Busy Here | | | | |-| | | | |ACK | | | | |-| | | |302 MOVED (to C) | | | |-| | | | |ACK | | | | |-| | | | |INVITE C | | | | |-| | | | | |INVITE C | | | | |-| | | | |503 Unavailable | | | |-| | | | |ACK | | | | |-| | | |503 Unavailable | | | |-| | | | |ACK | | | | |-| | | | | | | | | | | | | | 1.- Phone A calls Phone B behind Asterisk. 2.- Phone B rejects call by sending a '486 Busy Here' response. 3.- When OpenSER receives the 486 it sends a '302 Moved Temporarily' to Phone A to redirect the call to Phone C. 4.- Phone A perfoms the redirection and sends a new INVITE to Phone C (that is also behind Asterisk) with same call-id BUT DIFFERENT from-tag, CSeq. 5.- Asterisk, for some reason, considers the new INVITE to belong to the previous call and then rejects the call with a '503 Unavailable'. But it cannot be considered to belong to the same dialog because the tags are different, although the call-id is the same. We have used pedantic checking. Could it be considered as a bug? Looking at the code of chan_sip.c (version 1.4.23.1), we have observed that in function 'find_call' line 4667, asterisk is considering the call as FOUND because of this test: !ast_test_flag(p-flags[1
[asterisk-users] SIP dialog matching problem? (1.4.23.1)
Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER AsteriskPhoneB PhoneC | | | | | | | | | | | | | | | |INVITE B | | | | |-| | | | | |INVITE B | | | | |-| | | | | |INVITE B | | | | |-| | | | |486 Busy Here | | | | |-| | | | |ACK | | | | |-| | | |486 Busy Here | | | | |-| | | | |ACK | | | | |-| | | |302 MOVED (to C) | | | |-| | | | |ACK | | | | |-| | | | |INVITE C | | | | |-| | | | | |INVITE C | | | | |-| | | | |503 Unavailable | | | |-| | | | |ACK | | | | |-| | | |503 Unavailable | | | |-| | | | |ACK | | | | |-| | | | | | | | | | | | | | 1.- Phone A calls Phone B behind Asterisk. 2.- Phone B rejects call by sending a '486 Busy Here' response. 3.- When OpenSER receives the 486 it sends a '302 Moved Temporarily' to Phone A to redirect the call to Phone C. 4.- Phone A perfoms the redirection and sends a new INVITE to Phone C (that is also behind Asterisk) with same call-id BUT DIFFERENT from-tag, CSeq. 5.- Asterisk, for some reason, considers the new INVITE to belong to the previous call and then rejects the call with a '503 Unavailable'. But it cannot be considered to belong to the same dialog because the tags are different, although the call-id is the same. We have used pedantic checking. Could it be considered as a bug? Looking at the code of chan_sip.c (version 1.4.23.1), we have observed that in function 'find_call' line 4667, asterisk is considering the call as FOUND because of this test: !ast_test_flag(p-flags[1], SIP_PAGE2_DIALOG_ESTABLISHED). Commenting out this comparison, the call proceeds correctly. Sure, there is some reason for this checking and we would like to know which is and in what does it affect. How could we fix it? The following is the asterisk console output when the call does not proceed: [Mar 2 12:15:24] DEBUG[9989]: chan_sip.c:15813 handle_request: Received INVITE (5) - Command in SIP INVITE [Mar 2 12:15:24] NOTICE[9989]: chan_sip.c:14724 handle_request_invite: Unable to create/find SIP channel for this INVITE [Mar 2 12:15:24] DEBUG[9989]: chan_sip.c:4653 find_call: = Looking for Call ID: 9463d153-64f11de-8602e9bf-a87f5...@172.16.103.15 (Checking From) --From tag 182B3580-E9 --To-tag as62e21069 Any feedback would be appreciated. Thank you in advance, Santi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Asterisk and Java
Hello there. I have a problem that I can't solve. I am developing an application with Java and Asterisk. In addition, I am using Windows Vista, AsteriskWin32 PBX, asterisk-java-0.3.jar and XLite. I startup the DefaultAgiServer without problems and I have a java application running for the extension 1300(extensions.conf). When I use X-lite and make a call to extension 1300 the application is ok and I can listen to the messages that I put on the java code. Next, I tried to use the function getData to print the pressed keys from the softphone. I can listen to the sound that I set for the function but the answer for the pressed keys is always -1. I can't figure out the answer to this problem. Please help me to solve this issue. Greetings Santiago ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk and Java
Thanks for your answer Martin. The problem was the library. I updated the library to v1.0 Thanks for all With kind regards Santiago Panchi 2008/9/30 Martin Smith [EMAIL PROTECTED] -1 means Asterisk thinks the command failed. I've seen that if you hangup on the script, thought it might also happen if the file you specified doesn't exist. I encourage you to get the latest 1.0 snapshot from http://asterisk-java.org as we had one parsing bug due to spacing in the response upon a timeout with no digits pressed. I'd also encourage you to check out the Asterisk-Java mailing list via http://asterisk-java.org/development/mail-lists.html. Cheers, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Santiago Panchi *Sent:* Tuesday, September 30, 2008 10:14 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Question about Asterisk and Java Hello there. I have a problem that I can't solve. I am developing an application with Java and Asterisk. In addition, I am using Windows Vista, AsteriskWin32 PBX, asterisk-java-0.3.jar and XLite. I startup the DefaultAgiServer without problems and I have a java application running for the extension 1300(extensions.conf). When I use X-lite and make a call to extension 1300 the application is ok and I can listen to the messages that I put on the java code. Next, I tried to use the function getData to print the pressed keys from the softphone. I can listen to the sound that I set for the function but the answer for the pressed keys is always -1. I can't figure out the answer to this problem. Please help me to solve this issue. Greetings Santiago ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager Originate Action and Cancel
I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialling ZAP channel from analogue
El dom, 25-02-2007 a las 20:12 +, --[ UxBoD ]-- escribió:
[internal]
include = outbound-local
include = uri
...
you're dialing 912345678, which has 9 digits
[outbound-local]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
this extension has 8 digits, so they don't match, try adding another X.
[uri]
exten = _[a-z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _[A-Z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _X.,1,Macro(uridial,[EMAIL PROTECTED])
asterisk is using the uridial macro cause your number match _X.
[macro-uridial]
exten = s,1,NoOp(Calling remote SIP peer ${ARG1})
exten = s,n,Dial(SIP/${ARG1},120,tr)
exten = s,n,Congestion()
--
Santiago Ruano Rincón
http://www.avatar.com.co
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[asterisk-users] Asterisk 1.2.14 - Chanspy, sound issues.
I upgraded my Asterisk system to version 1.2.14 to check if the sound quality issues I was having with Chanspy in 1.2.7 remained. I'm still getting them, and I'm honestly out of ideas except from RTFS. The called party sounds normally fine, but it's impossible to hear the caller. Sometimes, when the called party is talking, the caller can also be heard. The conversation sounds broken, to the point is almost useless. We don't have any other quality problems beside this. Sound is quite good when making a call or accessing other asterisk services. My setup is as follows: All calls are performed inside a LAN (NOT fully switched...), using SIP and g711. I use SJPhone v1.60 at agents and AT-530 VoIP Phones for the spies. * Intel(R) Pentium(R) 4 CPU 3.00GHz, 1GB RAM, Broadcom Corporation NetXtreme BCM5705_2 Gigabit Ethernet. * Linux foo.bar.com 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux * Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1w built by bachbuilder @ octopus.physik.fu-berlin.de on a i686 running Linux on 2006-12-19 00:11:55 UTC. Is someone else getting this kind of behaviour? Is Chanspy used normally under this conditions on other installations? Any ideas? saludos, -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy begin:vcard fn:Santiago Aguiar n:Aguiar;Santiago org:;Desarrollo adr:;;Palmar 2548;Montevideo;Montevideo;11600;Uruguay email;internet:[EMAIL PROTECTED] title:NetLabs tel;work:+598 2 7077687 tel;fax:+598 2 7094866 tel;home:+598 2 7075079 tel;cell:+598 99 579739 x-mozilla-html:TRUE url:http://www.netlabs.com.uy/ version:2.1 end:vcard ___ --Bandwidth 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] Chanspy severe sound problems
Hi everyone! I'm using Asterisk 1.2.7.1 on a CentOS 4 server with 5 - 9 agents and I'm having some issues with the Chanspy application. All the agents are on SIP channels with g711 and all the communications are inside a LAN. When I'm spying a SIP channel, the audio from one of the ends (normally the caller) sounds *extremely* (unusable) choppy, as if it was losing some frames. Sometimes the called party is heard almost perfectly, but there are ALWAYS sound quality issues. The agents do not report any problem, and the audio recorded with the Monitor applications sounds reasonably fine. I'm able to reproduce the problem with any amount of load and it happened also while doing tests with my computer as an Asterisk server. Additional Information: * Asterisk 1.2.7.1 built by test @ ast3 on a i686 running Linux on 2006-04-24 10:52:49 UTC * Linux foo.bar.com 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux * Intel(R) Pentium(R) 4 CPU 3.00GHz, 1GB RAM. did anyone encountered the same situation? Google only reported one similar problem without a solution (http://bugs.digium.com/print_bug_page.php?bug_id=7340) any ideas are welcome! thanks a lot! saludos, -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy begin:vcard fn:Santiago Aguiar n:Aguiar;Santiago org:;Desarrollo adr:;;Palmar 2548;Montevideo;Montevideo;11600;Uruguay email;internet:[EMAIL PROTECTED] title:NetLabs tel;work:+598 2 7077687 tel;fax:+598 2 7094866 tel;home:+598 2 7075079 tel;cell:+598 99 579739 x-mozilla-html:TRUE url:http://www.netlabs.com.uy/ version:2.1 end:vcard ___ --Bandwidth 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] Queue Manager
Hi, I'm looking a queue manager compatible with queues.conf. It should allow me to change agents from one queue to another and change it's priority without poblem. Also it must be web based :). Does anyone know any program? Thank you in advance! Santiago del Castillo ___ --Bandwidth 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] DeStar 0.2.2 released!
Hi, someone has made me realize that a more detailed description is needed for those who don't know about DeStar, so: DeStar is a Web-based management and configuration tool for the Asterisk PBX. DeStar's main features include: * Hosted PBX and virtual PBX features, which allow you to have several PBXs on a single machine. * Extensions can be managed for SIP, IAX, Zap, and more. * Auto-attendants are supported. * Trunks can be managed for SIP, IAX, Zap, ZapPRI, and more. * Dialout patterns (i.e. local, national, mobile-phones) can be used. * Asternic Flash Operator Panel is integrated. * Many application applets are included for voice mail, meeting rooms, and more. * It is extensible through a pluguin-based architecture. Best regards, Santiago Ruano Rincón http://destar.berlios.de signature.asc Description: Esta parte del mensaje está firmada digitalmente ___ --Bandwidth 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] DeStar 0.2.2 released!
Hello, I'm glad to announce that DeStar 0.2.2 version has been released. This release contains a large number of bugfixes and new features, see CHANGELOG.txt for the full list. You can find it in the usual place: http://developer.berlios.de/project/showfiles.php?group_id=2112 Thanks for using DeStar, Santiago Ruano Rincón http://destar.berlios.de signature.asc Description: Esta parte del mensaje está firmada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch
Hi, someone has tried this combo? I have a SRW224P switch and i tried to make the phone to work with PoE on this switch but it isn't work. I read about this and i found that this phone needs an 'special cable' in order to work with PoE. It's that true? Isn't there any way to make it work with a normal cable? :( Thanks! Santiago ___ --Bandwidth 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] Queue RoundRobin
Hi, I'm setting up a new asterisk for an ecommerce company with cust sup dept. The problem I'm having is with Roundrobin (and rrmemory also): Let's suppose that I have 2 agents logged in into a queue. When a client calls, and both agents are available. It rings the first one, but it doesn't answer the phone. The timeout takes effect and it should start ringing the second agent. But it doesn't. It keeps ringing the first one until it answers the phone Here's my queue.conf: [general] [QueueEN] announce = ann-english strategy = rrmemory timeout = 5 retry = 1 wrapuptime=0 maxlen = 0 announce-frequency = 20 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 [QueueES] strategy = rrmemory timeout = 5 retry = 5 wrapuptime=0 maxlen = 0 announce = ann-spanish announce-frequency = 10 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 The timeout is set too low so the test is faster. Cheers, Santiago ___ --Bandwidth 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] DeStar 0.1 released!
Hi everyone, We are glad to announce to all the Asterisk community the first release of DeStar[1], a web based interface to manage the Asterisk PBX. DeStar provides high-level abstraction above the Asterisk configuration, making it real easy to quickly setup a basic PBX, but simultaneously allowing great flexibility for those out there intending to manage medium-complexity Asterisk based telephony systems. DeStar main features include: * Extensions management: SIP, IAX, Zap, and more. * Auto-attendants support. * Trunks management: SIP, IAX, Zap, ZapPRI, and more. * Use of dialout patterns (i.e. local, national, mobile-phones, toll-free numbers, etc). * Asternic Flash Operator Panel [2] integration. * Call Detail Records search and graphical reports. * Many application applets incluided: Voice Mail, Meeting Room, and more. DeStar is written in Python and uses Quixote[3], Sqlite[4] and Pychart[5]. You can download it from [1] or get it for the Debian GNU/Linux testing and unstable distributions via apt. A good starting point would be the Project Home Page[1] or the Project Wiki[6]. You may suscribe to the destar users list entering [7], where ALL questions are welcome. For developers, the list suscription can be made in [8], ALL questions are welcome too. Or if you prefer, you may find us at the DeStar IRC channel, where we'll be willing to answer your questions, discuss technical aspects of DeStar or just attend your complains about it ;-): Server: irc.freenode.net Port: 6667 Channel: #destar There's still a lot of work to do, so we encourage all of you who may be seeking for alternatives to configure the Asterisk PBX to join us. Testers, Programmers, Documentators, Translators, Graphic Designers, Usability Analyzers and users in general are needed. Thanks to all those who helped us to reach this point. Now enjoy DeStar! Best regards, The DeStar Development Team. --- List of links [1] http://destar.berlios.de/ [2] http://www.asternic.org/ [3] http://www.mems-exchange.org/software/quixote/ [4] http://www.sqlite.org/ [5] http://home.gna.org/pychart/ [6] http://openfacts.berlios.de/index-en.phtml?title=DeStar [7] http://lists.berlios.de/mailman/listinfo/destar-user [8] http://lists.berlios.de/mailman/listinfo/destar-dev -- Santiago Ruano Rincón Avatar Ltda. Parquesoft Popayán Huella digital llave GPG: 3821 4FB5 774A 611D 31E4 B268 414B 8423 6FEC CDE0 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT100 and BETA 1.0.7.11
Yes I did with no problems... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Monday, September 05, 2005 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] BT100 and BETA 1.0.7.11 Hi, Did anybody successfully updated Grandstream BT100 with BETA 1.0.7.11 ? For me so far no success. Bartosz ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.16/83 - Release Date: 8/26/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callerid...
Hi, asterisk Users, sorry for my bad English im really newbie with this excellent pbx. But I ve a problem with callerid num when I recive a call from PSTN. PSTN- SipGateWay(Welltech3504)- Asterisk- BT100 How can I configure my asterisk to receive the callerid from callers and not the callerid from the extension of the SipGAteway Extension of Gateway (sip.conf) [115] type=friend ; either friend (peer+user), peer or user context=sip user=115 host=dynamic canreinvite=no nat=no ; there is not NAT between phone and Asterisk disallow=all ; need to disallow=all before we can use allow= allow=ulaw ; Note: In user sections the order of codecs allow=alaw ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT100 and BETA 1.0.7.11
Sorry , I only did the upgrade firmware version without erros! Software Version: Program-- 1.0.7.11 Bootloader-- 1.0.7.1 HTML-- 1.0.7.11 VOC-- 1.0.1.0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Monday, September 05, 2005 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BT100 and BETA 1.0.7.11 I am missing some files my grandstream phone wants to download: bootloader.bin. I cannot find that file in release 1.0.7.11. Any ideas ? Bartosz - Original Message - From: Santiago Vega [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, August 25, 2005 4:24 PM Subject: RE: [Asterisk-Users] BT100 and BETA 1.0.7.11 Yes I did with no problems... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Monday, September 05, 2005 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] BT100 and BETA 1.0.7.11 Hi, Did anybody successfully updated Grandstream BT100 with BETA 1.0.7.11 ? For me so far no success. Bartosz ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.16/83 - Release Date: 8/26/2005 ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.16/83 - Release Date: 8/26/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel
Hi, I just got two E100P cards and one TDM400P four FXS. zaptel.conf: span=1,0,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 fxsks=63-66 zapata.conf switchtype=euroisdn ; Span 1 group=1 signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 signalling=pri_cpe channel = 32-46 channel = 48-62 But when i # /sbin/ztcfg ZT_SPANCONFIG failed on span 2: No such device or address (6) the dmesg output is: Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States / North America) What is going on? Thanks in advance, Santiago ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-addons compilation error
Folks I am getting the following error as of today after updating both asterisk and asterisk-addons. These are both under /usr/src. Any ideas? dora-debian:/usr/local/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: error: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: error: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: error: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: error: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 dora-debian:/usr/local/src/asterisk-addons# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF and SIP
hi! I'm having the same problem, I'm connecting through a Planet VIP-450 ITG, and when I send a DTMF code I get a: WARNING: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? I tried using different dtmf settings in sip.conf, but the message is still there. I don't have problems using a softphone... any ideas??? saludos! santiago. Lee Norvall wrote: Hi Just tried that, and still the same with the same error! The spec for the phones includes rfc2833, so I don't think that is it. Rgds -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Justin Carlson Sent: 02 June 2004 19:23 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DTMF and SIP have you tried commenting out the dtmf lines in your sip.conf we had similar problems with our snom 200's and after commenting out the dtmf lines in sip.conf asterisk reload they worked great :-) On Wed, 2004-06-02 at 11:36, Lee Norvall wrote: Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Questions
hi everyone! Two days ago we installed asterisk in our labs to do some testing and try the product with a couple of ITGs. Overall, we really loved it! We found it easy to configure and manage, and with good debugging options. There are a couple of questions I would like to ask: a) We had some authentication issues trying to register a Planet ITG with asterisk. Apparently, asterisk ignored the username attribute on the sip.conf entry: [10] type=friend username=foo secret=foosec host=dynamic context=sip-call The ITG was connecting as 'sip:10@ITG-IP' and its md5 was calculated using the specified user 'foo'. However, asterisk was using '10' to calculate the md5, and therefore authentication failed. We don't know if we found a bug or we are doing something wrong ;) (the code in question is in channels/chan_sip.c:3812, were it looks it sends peer-name instead of peer-username, on v0.9.0). b) Is it possible to make asterisk play a file in a codec supported by the client?? We tried to play tt_monkeys, but we got an error when passing from GSM to g723, which is ok, but the client supported g711 also, and I suppose it could be used by asterisk. We added allow=g711 to sip.conf and it worked (however, we had an error if we used allow=all since it tried sending in gsm, which wasn't supported by the ITG). c) We are getting some NOTICE: sched.c:218 sched_settime: Request to schedule in the past?!?! on the CLI, we don't know yet its cause or what it means. thanks a lot for the support! saludos! santiago. netlabs Palmar 2548 Montevideo, Uruguay +(598 2) 707-7687
RE: [Asterisk-Users] SIP and error talking to voicemail
- Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 09, 2004 11:10 AM Subject: RE: [Asterisk-Users] SIP and error talking to voicemail Original Message Subject: RE: [Asterisk-Users] SIP and error talking to voicemail From: Dave Cotton [EMAIL PROTECTED] Date: Fri, January 09, 2004 1:03 am To: Asterisk List [EMAIL PROTECTED] On Fri, 2004-01-09 at 06:37, [EMAIL PROTECTED] wrote: How come every time I try connecting to their TFTP server I get permission denied? Something I'm doing wrong? tftp connect 130.94.123.253 tftp get bootload.bin Error code 2: Do not have permission to use this TFTP server I put the tftp address into my Grandstream and powered down/up et voila! Somewhere else to download 1.0.4.30 and 1.0.4.17 (just as a backup -- what I have now)? the http address has 1.0.4.18, 1.0.4.26 and 1.0.4.30 in zip form. -- Dave Cotton [EMAIL PROTECTED] Late night. I've been to http://www.grandstream.com/TEMP/FIRMWARE/ I just would like to find 1.0.4.17 so I know I'm not introducing any new bugs if I have to go back. I meant to say if you know somewhere else to get 1.0.4.38. I also tried just downloading it from my grandstream but it didn't seem to even want to try it -- probably the same problem. I still get permission denied when I try to TFTP manually also. hmm... If anyone has either of them, I'd appreciate a copy! Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP and error talking to voicemail
Late night. I've been to http://www.grandstream.com/TEMP/FIRMWARE/ I just would like to find 1.0.4.17 so I know I'm not introducing any new bugs if I have to go back. I meant to say if you know somewhere else to get 1.0.4.38. I also tried just downloading it from my grandstream but it didn't seem to even want to try it -- probably the same problem. I still get permission denied when I try to TFTP manually also. hmm... If anyone has either of them, I'd appreciate a copy! We have a copy of 1.0.4.39. If you want, you can get at: http://www.supercomputo.com/b13p4.39.zip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS What to do?
hi fred, i don't know if this question has been already answered... i haven't tested it whit asterisk YET, (i have to) check the following links: http://luxik.cdi.cz/~devik/qos http://www.ibiblio.org/pub/Linux/docs/HOWTO/other-formats/html_single/ADSL-Bandwidth-Management-HOWTO.html and tell me if you have found a solution -- santiago josé ruano rincón administración servidores y servicios de internet red de datos universidad del cauca http://www.unicauca.edu.co/~santiago/llaves/santiago_pub.asc hay 10 tipos de personas, las que entienden binario y las que no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RAS
hi everybody is it posible to configure a RAS with a digium card in a linux box? thanks -- santiago jos ruano rincn administracin servidores y servicios de internet red de datos universidad del cauca http://www.unicauca.edu.co/~santiago/llaves/santiago_pub.asc hay 10 tipos de personas, las que entienden binario y las que no signature.asc Description: Esta parte del mensaje =?ISO-8859-1?Q?est=E1?= firmadadigitalmente
[Asterisk-Users] sound problem
hi list, when I run asterisk, appears the following: WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested 8000 Hz, got 8178 Hz -- sound may be choppy WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable but I can use oss with xmms what i have to do? thanks, -- santiago jos ruano rincn administracin servidores y servicios de internet red de datos universidad del cauca -BEGIN PGP MESSAGE- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org owGbwMvMwCQoeb96kq+XwjnGNbZJrGmZRbmJtpJvLwQn5pVkJqbnK3jlF79UCCpN zMtXCMrMS/6cxxWal1mWWlScmZKYopCSmqPgnFianMjF1WHPzMoA0gozUJDpLSfD /IB9Gre8ZHfZ+/BvkX4osko5wPfUQ4b5ST8VT3hciP3inJl578O17DedDS9fAgA= =5oc0 -END PGP MESSAGE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] usrobotics modem and pstn
hi, i have a external usrobotics modem, i want to use it with asterisk to interact with the pstn, what i have to do? thanks, -- santiago jos ruano rincn administracin servidores y servicios de internet red de datos universidad del cauca -BEGIN PGP MESSAGE- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org owGbwMvMwCQoeb96kq+XwjnGNbZJrGmZRbmJtpJvLwQn5pVkJqbnK3jlF79UCCpN zMtXCMrMS/6cxxWal1mWWlScmZKYopCSmqPgnFianMjF1WHPzMoA0gozUJDpLSfD /IB9Gre8ZHfZ+/BvkX4osko5wPfUQ4b5ST8VT3hciP3inJl578O17DedDS9fAgA= =5oc0 -END PGP MESSAGE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
