[asterisk-users] how to set iaxmodem receiving speed
Hi all, I'm trying to lower my iaxmodem speed but still I haven't found any solution...I tried to add Class1RMQueryCmd: !24,48,72 to config.IAXtty but does not work...Hylafax says it it running at 9600 (sometimes at 14400) baud.. Any ideas? Thank you. Giorgio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk forward call
How about adding in a time check so that during certain hours Asterisk waits 8 seconds before answering, otherwise it answers right away. You could also setup a status variable within the AstDB to indicate immediate answer or delayed answer. Just some thoughts. Dale On 05/15/2012 05:40 PM, motty.cruz wrote: Thanks John, I was trying to find a way to work around the partial ring. Ignoring is what I'm doing now wait(8) seconds but when the number not being forward that means there is a delay of 8seconds before phones start ringing. Thanks, motty -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Tuesday, May 15, 2012 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk forward call Call forwarding ALWAYS sends a partial ring to the line. This is by design from ESS 1 days, and serves to remind the party that the line is forwarded. Asterisk certainly can be configured to ignore the first ring in any number of ways. John Novack Guy Gold wrote: On Tue,May 15 02:00:PM, motty.cruz wrote: Hello All, My Asterisk server is working fine except that at night I forward my number to another phone number, however my asterisk server still rings once before call is forward. My Local Phone provider is ATT and they said that there is not way around it, I'm always going to get a partical ring. Any suggestions how to stop the Asterisk from rining once before forward to another number? Hi Motty, I'm assuming that you're forwarding calls unconditionally, yes ? I haven't tested it for a while, but, I'm pretty sure that if your PBX is not told to ring a device before forwarding the call, it should not do so. I do recall having worked in a non-PBX office , and when we performed CFWD, the local phone would ring once and then get forwarded, but, that's because the local phone never took control of the call coming from the carrier. In your case , the PBX can take over the call, never produce a ring , and then dial the CFWD number. I guess a trace of this instance can be useful . Guy Gold -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The truth speaks for itself. I'm just the messenger. Lyta Alexander - Babylon 5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
I have iaxmodem version 1.2.0 installed on my system. I have set the following in the IAX configuration file, SIGHUP'd FaxGetty and submitted a single page outbound fax via Asterisk; Class1RMQueryCmd: !24,48,72 # enable this to disable V.17 receiving Class1TMQueryCmd: !24,48,72 # enable this to disable V.17 sending The resulting output from my T.38 Gateway reports the following; -- Connection Statistics Bit Rate :7200 ECM : No Pages : 1 -- Hungup 'IAX2/iaxmodem0-11055' I also tested with the maximum speed set to 4800, the image was received however the responses to EOP timed out, I don't know if the is to do with my Asterisk T.38 gateway or my VoIP providers T.38 gateway. The result was the fax was retried for the defined number of attempts. Cheers, Larry. On 16/05/2012 6:28 PM, gincantalupo wrote: Hi all, I'm trying to lower my iaxmodem speed but still I haven't found any solution...I tried to add Class1RMQueryCmd: !24,48,72 to config.IAXtty but does not work...Hylafax says it it running at 9600 (sometimes at 14400) baud.. Any ideas? Thank you. Giorgio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wrong SIP to SIP SIP Cause mapping
Hello,
I'm using asterisk v1.8 with a standard scenario, A Sip call from A to B
through asterisk :
A --SIP-- ASTERISK --SIP-- B
The asterisk extension is :
exten = _X.,1,Dial(SIP/B/${EXTEN},600)
exten = _X.,n,Hangup()
When B send a 404 back to the asterisk, the asterisk sends a 503 to A. It is
the same with 403 and some others erroc code.
I think it should send back to A the same error code.
I have done tests with some versions:
- 1.8.11.x : wrong sip cause mapping
- 1.8.13.0rc1 : wrong sip cause mapping
- 1.10.3 : wrong sip cause mapping
- 1.8.8.0 : works good
Do i do something wrong or should i open a bug ?
MOUTOT A.
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[asterisk-users] AsteriskNow: Does it support call details records?
Hi All; I did not install AsteriskNow, but I am thinking to install it. If I installed it, I can see the call details records for the extensions? I can request the CDR to be between specific dates and to specific extension? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk queue: announce in more than one language or announceoverride
Hi, I'd like a single queue to announce the caller's position, etc., in more than one language without user interaction. ie. announce position in English then in French then in Spanish Is this possible (without ivr)? Can anyone please give me a Queue cmd example with 'announceoverride'? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
Read the subject line more closely. Tested receiving too, I set the Send Receive speed of the receiving analogue modem to that below, the log file on the sending modem (iaxmodem) reported it capable of 9600. May 16 21:32:04.28: [ 2335]: REMOTE best rate 9600 bit/s May 16 21:32:04.28: [ 2335]: REMOTE max A3 page width (303 mm) May 16 21:32:04.28: [ 2335]: REMOTE max unlimited page length May 16 21:32:04.28: [ 2335]: REMOTE best vres R16 x 15.4 line/mm May 16 21:32:04.28: [ 2335]: REMOTE format support: MH, MR, MMR May 16 21:32:04.28: [ 2335]: REMOTE supports T.30 Annex A, 256-byte ECM May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline May 16 21:32:04.28: [ 2335]: USE 9600 bit/s Perhaps the issue is with Hylafax. Setting the Transmit Receive strings to !24,48,72,96 seems to yield the most reliability in transmission Cheers, Larry. On 16/05/2012 7:23 PM, Larry Moore wrote: I have iaxmodem version 1.2.0 installed on my system. I have set the following in the IAX configuration file, SIGHUP'd FaxGetty and submitted a single page outbound fax via Asterisk; Class1RMQueryCmd: !24,48,72 # enable this to disable V.17 receiving Class1TMQueryCmd: !24,48,72 # enable this to disable V.17 sending The resulting output from my T.38 Gateway reports the following; -- Connection Statistics Bit Rate :7200 ECM : No Pages : 1 -- Hungup 'IAX2/iaxmodem0-11055' I also tested with the maximum speed set to 4800, the image was received however the responses to EOP timed out, I don't know if the is to do with my Asterisk T.38 gateway or my VoIP providers T.38 gateway. The result was the fax was retried for the defined number of attempts. Cheers, Larry. On 16/05/2012 6:28 PM, gincantalupo wrote: Hi all, I'm trying to lower my iaxmodem speed but still I haven't found any solution...I tried to add Class1RMQueryCmd: !24,48,72 to config.IAXtty but does not work...Hylafax says it it running at 9600 (sometimes at 14400) baud.. Any ideas? Thank you. Giorgio -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk forward call
So Asterisk is playing no role during the times you have the calls forwarded, and you just don't want it to ring? Why not just make it go off-hook during those times? On Tue, May 15, 2012 at 3:42 PM, motty.cruz motty.c...@gmail.com wrote: ** Hello Carlos, I'm already using the provider to forward #72 throught my provider, but by default it rings once. at this moment i'm ignoring the first 8 seconds when someone call my number otherwise my phone ring twice. Thanks, -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Carlos Alvarez *Sent:* Tuesday, May 15, 2012 2:18 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk forward call On Tue, May 15, 2012 at 2:00 PM, motty.cruz motty.c...@gmail.com wrote: Any suggestions how to stop the Asterisk from rining once before forward to another number? Since analog lines can only signal using ANALOG methods, it will always have a partial ring. Otherwise there's no way for Asterisk to know it is ringing. Why don't you just forward it using the phone company's call forwarding service? -- Carlos Alvarez TelEvolve 602-889-3003 -- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 50% of time SendDTMF failed
I am having a problem with SendDTMF() - 50% of time it did not succeed. I suspect it is not sending clear DTMF tones to the IVR. For example: SendDTMF(w3w2ww1w4) Sometime digit 3 and 2 work, and failed to do digit 1. Sometime digit 3 work and failed to do number 2. Sometime all went through fine. dtmfmode=rfc2833 are set in the sip.conf file How do I debug to see what went wrong and how to fix? Asterisk 1.8.12.0 Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W - Located in UK) VOIP Provider in UK. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
Hi Larry, thank you for your answer. This is same test I did. After this I lowered again to 4800...result: iaxmodem receives at 9600 b/s.that's why I cannot solve the puzzle. I put that line (Class1RMQueryCmd: !24,48,72) in config.IAXtty and tried other configuration files too but I always get faxes at 9600. Thank you Giorgio On 05/16/2012 03:59 PM, Larry Moore wrote: Read the subject line more closely. Tested receiving too, I set the Send Receive speed of the receiving analogue modem to that below, the log file on the sending modem (iaxmodem) reported it capable of 9600. May 16 21:32:04.28: [ 2335]: REMOTE best rate 9600 bit/s May 16 21:32:04.28: [ 2335]: REMOTE max A3 page width (303 mm) May 16 21:32:04.28: [ 2335]: REMOTE max unlimited page length May 16 21:32:04.28: [ 2335]: REMOTE best vres R16 x 15.4 line/mm May 16 21:32:04.28: [ 2335]: REMOTE format support: MH, MR, MMR May 16 21:32:04.28: [ 2335]: REMOTE supports T.30 Annex A, 256-byte ECM May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline May 16 21:32:04.28: [ 2335]: USE 9600 bit/s Perhaps the issue is with Hylafax. Setting the Transmit Receive strings to !24,48,72,96 seems to yield the most reliability in transmission Cheers, Larry. On 16/05/2012 7:23 PM, Larry Moore wrote: I have iaxmodem version 1.2.0 installed on my system. I have set the following in the IAX configuration file, SIGHUP'd FaxGetty and submitted a single page outbound fax via Asterisk; Class1RMQueryCmd: !24,48,72 # enable this to disable V.17 receiving Class1TMQueryCmd: !24,48,72 # enable this to disable V.17 sending The resulting output from my T.38 Gateway reports the following; -- Connection Statistics Bit Rate :7200 ECM : No Pages : 1 -- Hungup 'IAX2/iaxmodem0-11055' I also tested with the maximum speed set to 4800, the image was received however the responses to EOP timed out, I don't know if the is to do with my Asterisk T.38 gateway or my VoIP providers T.38 gateway. The result was the fax was retried for the defined number of attempts. Cheers, Larry. On 16/05/2012 6:28 PM, gincantalupo wrote: Hi all, I'm trying to lower my iaxmodem speed but still I haven't found any solution...I tried to add Class1RMQueryCmd: !24,48,72 to config.IAXtty but does not work...Hylafax says it it running at 9600 (sometimes at 14400) baud.. Any ideas? Thank you. Giorgio -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
Hi Larry, I forgot to mention I tried to set ModemRate at 4800 as well but without success. Giorgio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
On 16-05-12 17:10, gincantalupo wrote: Hi Larry, thank you for your answer. This is same test I did. After this I lowered again to 4800...result: iaxmodem receives at 9600 b/s.that's why I cannot solve the puzzle. I put that line (Class1RMQueryCmd: !24,48,72) in config.IAXtty and tried other configuration files too but I always get faxes at 9600. Did you restart iaxmodem and hylafax after you made those changes? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk forward call
Here is the issue, I have Asterisk Server with Digium TDM400P card four ports four anolog lines come throught this server, but we forward all this lines to a voip number (to our main Asterisk Server), for various reason call ID is one of them. When we forward the number to our voip Asterisk Server we get two incoming calls one from our anolog lines because of the partial ring and one from our main Asterisk Server, that is the reason i want to elimitate that partial ring. At this moment when our Asterisk Server with anolog lines get a call it wait 8 seconds before forward that call on. Hope i was specific and clear. Thanks, _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Wednesday, May 16, 2012 7:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk forward call So Asterisk is playing no role during the times you have the calls forwarded, and you just don't want it to ring? Why not just make it go off-hook during those times? On Tue, May 15, 2012 at 3:42 PM, motty.cruz motty.c...@gmail.com wrote: Hello Carlos, I'm already using the provider to forward #72 throught my provider, but by default it rings once. at this moment i'm ignoring the first 8 seconds when someone call my number otherwise my phone ring twice. Thanks, _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Tuesday, May 15, 2012 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk forward call On Tue, May 15, 2012 at 2:00 PM, motty.cruz motty.c...@gmail.com wrote: Any suggestions how to stop the Asterisk from rining once before forward to another number? Since analog lines can only signal using ANALOG methods, it will always have a partial ring. Otherwise there's no way for Asterisk to know it is ringing. Why don't you just forward it using the phone company's call forwarding service? -- Carlos Alvarez TelEvolve 602-889-3003 _ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 _ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5002 - Release Date: 05/16/12 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk forward call
The Ring is Asterisk ringing local extensions to indicate an incoming call (DAHDI-1 gets call, rings SIP/1000 and SIP/2000 to let you know you have a call). When you do the *72 forwarding, change the SIP/1000 and SIP/2000 ring to ring local/1. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz Sent: Wednesday, May 16, 2012 10:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk forward call Here is the issue, I have Asterisk Server with Digium TDM400P card four ports four anolog lines come throught this server, but we forward all this lines to a voip number (to our main Asterisk Server), for various reason call ID is one of them. When we forward the number to our voip Asterisk Server we get two incoming calls one from our anolog lines because of the partial ring and one from our main Asterisk Server, that is the reason i want to elimitate that partial ring. At this moment when our Asterisk Server with anolog lines get a call it wait 8 seconds before forward that call on. Hope i was specific and clear. Thanks, _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Wednesday, May 16, 2012 7:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk forward call So Asterisk is playing no role during the times you have the calls forwarded, and you just don't want it to ring? Why not just make it go off-hook during those times? On Tue, May 15, 2012 at 3:42 PM, motty.cruz motty.c...@gmail.com wrote: Hello Carlos, I'm already using the provider to forward #72 throught my provider, but by default it rings once. at this moment i'm ignoring the first 8 seconds when someone call my number otherwise my phone ring twice. Thanks, _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Tuesday, May 15, 2012 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk forward call On Tue, May 15, 2012 at 2:00 PM, motty.cruz motty.c...@gmail.com wrote: Any suggestions how to stop the Asterisk from rining once before forward to another number? Since analog lines can only signal using ANALOG methods, it will always have a partial ring. Otherwise there's no way for Asterisk to know it is ringing. Why don't you just forward it using the phone company's call forwarding service? -- Carlos Alvarez TelEvolve 602-889-3003 _ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 _ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5002 - Release Date: 05/16/12 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
Hi Patrick, of course I did. Stopped hylafax, killed iaxmodem process, re-started iaxmodem then hylafax. Result: Time To Receive: 0:00:14 Signal Rate: 9600 bit/s Data Format: 2-D MMR Error Correct: Yes :( At the very beginning before starting every kind of test, inside my hylafax/config/iaxmodem I added: Class1RMQueryCmd: !24,48# V.17 fast-train recv doesn't work well Class1TMQueryCmd: !24,48# V.17 fast-train recv doesn't work well Giorgio On 05/16/2012 05:33 PM, Patrick Lists wrote: On 16-05-12 17:10, gincantalupo wrote: Hi Larry, thank you for your answer. This is same test I did. After this I lowered again to 4800...result: iaxmodem receives at 9600 b/s.that's why I cannot solve the puzzle. I put that line (Class1RMQueryCmd: !24,48,72) in config.IAXtty and tried other configuration files too but I always get faxes at 9600. Did you restart iaxmodem and hylafax after you made those changes? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Incoming fax cuts ADSL line
Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
- Original Message - Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used. I'm unsure of the exact technical reasons behind this other than 'the fax signals/frequencies interfere with the ADSL signalling/frequencies used on the circuit'. It sounds like you might want to separate your fax/ADSL lines. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used I too have seen this, and also with credit card processing machines in shops that 'dial' the merchant bank to process transactions (in effect a modem). It can sometimes (but not always) be resolved by running two microfilters in series on the 'voice' side of the line, i.e. line - microfilter - microfilter - fax. I've also (here in the UK) seen it resolved through the use of higher quality faceplate splitters rather than the often low-cost units supplied free with consumer ADSL modem/routers. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
On 5/16/2012 12:07 PM, Tim Nelson wrote: - Original Message - Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used. I'm unsure of the exact technical reasons behind this other than 'the fax signals/frequencies interfere with the ADSL signalling/frequencies used on the circuit'. It sounds like you might want to separate your fax/ADSL lines. --Tim You might also be able to limit the Fax machines maximum transmission rate so the modem's transmission spectrum doesn't inch up into where the ADSL service is. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.1.3 on SUSE 10 SP2
Hi List, I finally got a VM I can take up and down to look at this problem. To recap, I'm trying to setup 10.1.3 on SUSE 10 SP2. Here is the uname -a output Linux XXX 2.6.16.60-0.21-smp #1 SMP Tue May 6 12:41:02 UTC 2008 x86_64 x86_64 x86_64 GNU/Linux I was able to get Asterisk up and running by doing make -I make install -i Make -I output [LD] astdb2sqlite3.o db1-ast/libdb1.a - astdb2sqlite3 /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so: file not recognized: File format not recognized collect2: ld returned 1 exit status +- Asterisk Build Complete -+ + Asterisk has successfully been built, and + + can be installed by running: + + + +make install + I have to delete the astdb table from /var/lib/asterisk/astdb.sqlite3 before each instance of Asterisk, but otherwise it seems to function properly. I've googled until my eyes are sore and this is the only answer I've found - the db1-ast/libdb1.a seems to be incompatible with the linker. Any other suggestions so this Asterisk install can work normally? P.S. I also had to change main/db.c to remove the IF NOT EXISTS in line 125. This hack isn't needed in 11 SP1. Thanks in advance Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Sharp Sent: Wednesday, May 16, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Incoming fax cuts ADSL line On 5/16/2012 12:07 PM, Tim Nelson wrote: - Original Message - Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used. I'm unsure of the exact technical reasons behind this other than 'the fax signals/frequencies interfere with the ADSL signalling/frequencies used on the circuit'. It sounds like you might want to separate your fax/ADSL lines. --Tim You might also be able to limit the Fax machines maximum transmission rate so the modem's transmission spectrum doesn't inch up into where the ADSL service is. ADSL is transmitted at a relatively low frequency using phase modulated carriers to achieve the bandwidth. It could be about 32 different phase/level locations on 360 degree/level pie chart or vector scope. The actual frequencies of the carrier are moderately low, maybe 100 to 200 kcps. Voice is low density. Faxes and modems are high density and loud. They can splatter or have harmonics that can confuse the local DSL demodulator. As others have said, the best thing to try is the best filters you can get between the phone line and the DSL demod, and maybe two filters in series. If that doesn't work, put the fax on a different line than the DSL, which could cost you money. Paying for better filters or two of them is less expensive than separate lines. Or move the DSL to an alternate existing voice only line, since you probably don't want to change the fax number Contents of this message were dredged from foggy memory. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Generate $500 $2500 a month - Own Your Own Business http://parkovani-u-letiste-praha.cz/httpwagregerw2.php?aforcamp=329 Well, this is it, Capet. kevon wingate Wed, 16 May 2012 18:07:05 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
Hi, On 05/16/2012 09:59 PM, Larry Moore wrote: Read the subject line more closely. Tested receiving too, I set the Send Receive speed of the receiving analogue modem to that below, the log file on the sending modem (iaxmodem) reported it capable of 9600. May 16 21:32:04.28: [ 2335]: REMOTE best rate 9600 bit/s May 16 21:32:04.28: [ 2335]: REMOTE max A3 page width (303 mm) May 16 21:32:04.28: [ 2335]: REMOTE max unlimited page length May 16 21:32:04.28: [ 2335]: REMOTE best vres R16 x 15.4 line/mm May 16 21:32:04.28: [ 2335]: REMOTE format support: MH, MR, MMR May 16 21:32:04.28: [ 2335]: REMOTE supports T.30 Annex A, 256-byte ECM May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline May 16 21:32:04.28: [ 2335]: USE 9600 bit/s Perhaps the issue is with Hylafax. Setting the Transmit Receive strings to !24,48,72,96 seems to yield the most reliability in transmission If you have an ATA in the path that is often the case. Many of them badly mess up a FAX signal. Without such a distortion machine V.17 should be fine. Cheers, Larry. On 16/05/2012 7:23 PM, Larry Moore wrote: I have iaxmodem version 1.2.0 installed on my system. I have set the following in the IAX configuration file, SIGHUP'd FaxGetty and submitted a single page outbound fax via Asterisk; Class1RMQueryCmd: !24,48,72 # enable this to disable V.17 receiving Class1TMQueryCmd: !24,48,72 # enable this to disable V.17 sending The resulting output from my T.38 Gateway reports the following; -- Connection Statistics Bit Rate :7200 ECM : No Pages : 1 -- Hungup 'IAX2/iaxmodem0-11055' I also tested with the maximum speed set to 4800, the image was received however the responses to EOP timed out, I don't know if the is to do with my Asterisk T.38 gateway or my VoIP providers T.38 gateway. The result was the fax was retried for the defined number of attempts. Cheers, Larry. On 16/05/2012 6:28 PM, gincantalupo wrote: Hi all, I'm trying to lower my iaxmodem speed but still I haven't found any solution...I tried to add Class1RMQueryCmd: !24,48,72 to config.IAXtty but does not work...Hylafax says it it running at 9600 (sometimes at 14400) baud.. This is correct behaviour. The sending side has fine control over the modem modes it uses. The receiving side can only specify that V.27ter, or V.27ter+V.29 or V.27ter+V.29+V.17 are OK. So, if you allow the 7200bps mode of V.29 you are compelled to allows the 9600bps mode too. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Problem on direct FXO port
Hi, I´m with asterisk 1.6.2.20 DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2 SpanDSP: spandsp-0.0.6pre20.tgz FXO lines. Sending faxes works ok. but receiving gives me error: here is the debug: http://pastebin.com/qfFeXWQW any idea?? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.1.3 on SUSE 10 SP2 - Resolved
Hi List, The resolution for the problem was to modify utils/Makefile with this line _ASTCFLAGS+=-DSTANDALONE -fPIC Instead of _ASTCFLAGS+=-DSTANDALONE Still doesn't resolve the fact that sqlite3 version 3.2.8 doesn't like the create table if not exists astdb, but I can live with that. Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
On 17/05/2012 1:24 AM, Steve Underwood wrote: Hi, On 05/16/2012 09:59 PM, Larry Moore wrote: Read the subject line more closely. Tested receiving too, I set the Send Receive speed of the receiving analogue modem to that below, the log file on the sending modem (iaxmodem) reported it capable of 9600. May 16 21:32:04.28: [ 2335]: REMOTE best rate 9600 bit/s May 16 21:32:04.28: [ 2335]: REMOTE max A3 page width (303 mm) May 16 21:32:04.28: [ 2335]: REMOTE max unlimited page length May 16 21:32:04.28: [ 2335]: REMOTE best vres R16 x 15.4 line/mm May 16 21:32:04.28: [ 2335]: REMOTE format support: MH, MR, MMR May 16 21:32:04.28: [ 2335]: REMOTE supports T.30 Annex A, 256-byte ECM May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline May 16 21:32:04.28: [ 2335]: USE 9600 bit/s Perhaps the issue is with Hylafax. Setting the Transmit Receive strings to !24,48,72,96 seems to yield the most reliability in transmission If you have an ATA in the path that is often the case. Many of them badly mess up a FAX signal. Without such a distortion machine V.17 should be fine. The receiving analogue modem is directly connected to the PSTN network and was used to to determine if the reported issue could be reproduced on a non-iaxmodem. My outgoing connections were through an iaxmodem with T.38 gateway enabled and disabled, with most successful transmissions being when T.38 gateway was used. Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
On 17/05/2012 12:18 AM, James Sharp wrote: On 5/16/2012 12:07 PM, Tim Nelson wrote: - Original Message - Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used. I'm unsure of the exact technical reasons behind this other than 'the fax signals/frequencies interfere with the ADSL signalling/frequencies used on the circuit'. It sounds like you might want to separate your fax/ADSL lines. --Tim You might also be able to limit the Fax machines maximum transmission rate so the modem's transmission spectrum doesn't inch up into where the ADSL service is. I have clients with their ADSL2+ service attached to their fax lines with no problems observed. Perhaps the issue is the fax machines attenuators are not set correctly are are to _loud_ on the PSTN. In Australia Telstra advised the signal level received at the exchange should be between -15dB and -17dB. They have a Fax On Line Diagnostic System (FOLDS) which you can send a transmission to, a report is returned advising of the quality of your transmission including measured signal and noise levels. In old days the fax machine might have a wired jumper block to set the attenuation, more modern devices would be configured from the front panel, typically in a maintenance mode. Your good old dial-up modems with fax capabilities would have an S-Register or two to set the attenuation. Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R-Series with NON-DIGIUM card on servers
Thanks Kevin. Buying one for Spain right now ;) 2012/5/15 Kevin P. Fleming kpflem...@digium.com On 05/12/2012 12:07 PM, Danny Dias wrote: What about the Database and recording calls replication? as i could see, the RSeries does not take into account these data. The Digium R-series devices are electronic switches used for routing telephony circuits; they don't have any part in the actual failover process, data replication, or anything of the sort. All of those functions need to be handled via software on the server(s) involved. The R-series user's manual describes one way this can be done using Asterisk and open source tools commonly available on Linux distributions. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
