Re: [asterisk-users] remote UPDATE command
Check the sip.conf.sample. 1.8 has several options related to the SIP UPDATE support. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arstan Sent: Monday, March 12, 2012 10:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] remote UPDATE command Hi guys, I am working on a setup where I have an Asterisk (1.6.2.7) with a SIP trunk to a provider's softswitch(IMS). Trunking works all ok, calling out and calling in. Except for the remote softswitch(IMS) will send UPDATE command(according to RFC 3311) and Asterisk does not accept and thus the call gets dropped. So, is there a work around for this matter? Or any of you had similar problem, I really appreciate any directions. The interesting part is that, they have an appliance based Asterisk that works perfectly fine, so I assume that it actually works ok. Just need to find out hows that possible.. Thanks! -- Regards, Arstan Jusupov -- _ -- 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] remote UPDATE command
Hi guys, I am working on a setup where I have an Asterisk (1.6.2.7) with a SIP trunk to a provider's softswitch(IMS). Trunking works all ok, calling out and calling in. Except for the remote softswitch(IMS) will send UPDATE command(according to RFC 3311) and Asterisk does not accept and thus the call gets dropped. So, is there a work around for this matter? Or any of you had similar problem, I really appreciate any directions. The interesting part is that, they have an appliance based Asterisk that works perfectly fine, so I assume that it actually works ok. Just need to find out hows that possible.. Thanks! -- Regards, Arstan Jusupov -- _ -- 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] Rate sheet "normalization"
On 03/12/2012 06:52 PM, Markus wrote: Now, if the "49" route of the first provider is cheaper, my system (a2billing) will still use the more expensive "4930" code because it is more specific. There is a great deal of wisdom in this approach that you may wish to consider carefully before abandoning. It is especially true in Europe, and really, anywhere outside the North American environment, which it doesn't sound like you're in to begin with. Calls to mobile operators, as well as certain premium number allocation prefixes, can be many times more expensive than calls to fixed-line operators. If 4945 were a mobile prefix that was three times more expensive than the overall +49 rate offered by the same provider, are you absolutely certain that you want to use the +49 rate? Even if you have some vendor that gives you a blended overall rate for +49, and you want to cherry-pick expensive routes off their blended rate plan, that still means you need to sort by longest prefix (descending) to know what you're comparing it against. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- 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] Rate sheet "normalization"
Hi, this question is not Asterisk specific, but since there are so many experts present on this list, maybe its OK to ask anyways. I'm having a hard time "normalizing" rate sheets from different providers. What I mean with this: the goal is to always get the cheapest rate for a given destination. What I would like to do is throw like 10 rate sheets from different providers together and as output get a single rate sheet with only the cheapest rates. However, some providers are listing a country, lets say Germany, as code "49" with a specific rate, and another provider will list each city individually, and each code separately, e.g. Berlin "4930", Hamburg "4940" etc., and probably different cities have different rates as well. Now, if the "49" route of the first provider is cheaper, my system (a2billing) will still use the more expensive "4930" code because it is more specific. I'm looking for some awesome, smart tool that will automatically "normalize" all these code differences and output a clean ratesheet with only the cheapest rates. Does such a thing exist? I wonder how everyone else is "normalizing" their different rate sheets. With a homebrewn script? 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] Capacity of single instance of Asterisk
On 03/12/2012 03:38 PM, Steve Edwards wrote: On Mon, 12 Mar 2012, Amit Patkar wrote: What will be impact on no of session when G729a is used? Assuming that transcoding is involved; if all the system is doing is passing through G.729A media streams, and recording them in unmixed G.729A format, there's no additional impact (the system might actually perform slightly better, as there is substantially less data being shuffled around). -- 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-users
Re: [asterisk-users] Capacity of single instance of Asterisk
On Mon, 12 Mar 2012, Amit Patkar wrote: I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many concurrent SIP sessions I can run from single instance of Asterisk on this server? I wish to use G711 codec with echo cancel. And all calls needs to be recorded. What kind of capacity are you looking to achieve? From my experience, Asterisk is not really much of a RAM hog. A couple GB is good for a couple hundred simultaneous calls. With 4 'Intel(R) Xeon(TM) CPU 3.40GHz' cores, I can handle a couple hundred simultaneous non-transcoding calls with no recording on Asterisk 1.2. With 24 cores and 16 GB on tap, you will probably find other resource limitations before either CPU or RAM are a limiting factor. Personally, I'm a 'don't put all your eggs in one basket' kind of guy. Assuming a simplistic linear relationship between my host and yours, what will you do when it crashes with 1600 calls in progress? What will you do when you need to install patches or upgrade or ... I like a couple of instances of OpenSIPS in front of several Asterisk instances, even if OpenSIPS is on the same boxes as Asterisk. What will be impact on no of session when G729a is used? Significant. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Very slow faxing with Fax For Asterisk
Che Switch to a Grandstream HT-701 or HT-502 and see if you get better results. The HT-286 has been discontinued for some time and the firmware has some major T.38 issues. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Chet W. Stevens" Sent: Monday, March 12, 2012 3:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Very slow faxing with Fax For Asterisk I have been testing Fax For Asterisk with an analog fax sitting behind a Grandstream Handytone 286. I have been sending a number of faxes but I find that they send extremely slow. In my most basic test of sending from the HT286 straight to a tiff image on Asterisk on the same subnet I am seeing as much as 200 seconds to send a single page. Debug in the CLI shows that we are negotiating down to T.38 4800. I have disabled echo cancellation and noise suppression on the HT286 but with no change. I have tested with different fax machines (HP 1040 Fax and Sharp UX-B800SE) but with the same results. My Asterisk and FFA module versions are: FAX For Asterisk Components:Applications: SVN-branch-1.8-digiumphones-r357808-/branches/1.8Digium FAX Driver: 1.8.4_1.3.0 (optimized for generic_64) Your suggestions are appreciated. Thank you. Chet Stevens cwstev...@interact.ccsd.net -- _ -- 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] Very slow faxing with Fax For Asterisk
I have been testing Fax For Asterisk with an analog fax sitting behind a Grandstream Handytone 286. I have been sending a number of faxes but I find that they send extremely slow. In my most basic test of sending from the HT286 straight to a tiff image on Asterisk on the same subnet I am seeing as much as 200 seconds to send a single page. Debug in the CLI shows that we are negotiating down to T.38 4800. I have disabled echo cancellation and noise suppression on the HT286 but with no change. I have tested with different fax machines (HP 1040 Fax and Sharp UX-B800SE) but with the same results. My Asterisk and FFA module versions are: FAX For Asterisk Components: Applications: SVN-branch-1.8-digiumphones-r357808-/branches/1.8 Digium FAX Driver: 1.8.4_1.3.0 (optimized for generic_64) Your suggestions are appreciated. Thank you. Chet Stevens cwstev...@interact.ccsd.net -- _ -- 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] Capacity of single instance of Asterisk
There are no such statistics. Your usage patterns are unique to you and depend on many factors. If you must look for the information then look in the mailing list archives or on voip-info.org. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Amit Patkar Sent: Monday, March 12, 2012 2:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Capacity of single instance of Asterisk Hi Can someome give tested and proven information on Asterisk capabilities? I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many concurrent SIP sessions I can run from single instance of Asterisk on this server? I wish to use G711 codec with echo cancel. And all calls needs to be recorded. What will be impact on no of session when G729a is used? Thanks & Regards, Amit Patkar -- _ -- 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] Capacity of single instance of Asterisk
Hi Can someome give tested and proven information on Asterisk capabilities? I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many concurrent SIP sessions I can run from single instance of Asterisk on this server? I wish to use G711 codec with echo cancel. And all calls needs to be recorded. What will be impact on no of session when G729a is used? Thanks & Regards, Amit Patkar -- _ -- 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] Problem with ReceiveFax
On 03/12/2012 11:10 AM, Larry Moore wrote: Check Start & End RTP Ports match values set in /etc/asterisk/udptl.conf for udptlstart & udptlend This is unnecessary; the two endpoints are free to use different port ranges if they wish, it won't make any difference. -- 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-users
Re: [asterisk-users] Problem with ReceiveFax
On 12/03/2012 10:53 PM, Ishfaq Malik wrote: Thanks for the input so far. I'm going to keep plugging away and if anyone has any insights, they will be gladly appreciated. Ish In SIP Account Configuration on Draytek; Set Voice Active Detect to Off In Phone Settings on the Draytek; Enable Symmetric RTP Check Start & End RTP Ports match values set in /etc/asterisk/udptl.conf for udptlstart & udptlend In /etc/asterisk/udptl.conf set; use_even_ports=yes -- _ -- 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] Problem with ReceiveFax
On Mon, 2012-03-12 at 14:53 +, Ishfaq Malik wrote: > On Mon, 2012-03-12 at 09:11 -0400, Bryant Zimmerman wrote: > > Looking at the information you have sent in this posting in certainly > > appears that the 'f' option has indeed helped however you have > > another > > matter to overcome. > > > > You may wish to set the following parameters in your peer > > configuration > > for 588. > > > > ignoresdpversion=yes > This one passed me by, thanks for the info > > directmedia=no > > > > I use Spandsp FAX successfully. > > > > I have also attached an analogue Fax Modem to the FXS port on an > > SPA8800 > > and an HT-502 and have been able to receive faxes on them when I last > > tested, the SPA8800 like the HT-502 are now in storage. > > > > Looking at the User Guide for the Vigor 2701 there is an option in > > the > > configuration to enable T.38 mode, did you enable it? > Foolishly I hadn't and feel very stupid about that! > > > > > > In my sip.conf I have the following; > > > > [general] > > . > > . > > . > > faxdetect=cng > > t38pt_udptl=yes,redundancy,maxdatagram=400 > tried this and ,fec, > > ;t38pt_usertpsource=yes > had this already set > > . > > . > > . > > [903] > > ; Cisco SPA8800 FXS Port 3 > > ; Grandstream HT502 FXS Port 1 > > ; Analogue FAX Modem attached > > type=friend > > defaultuser=903 > > secret=you_guessed_it > > call-limit=2 > > disallow=g722 > > transport=udp > > qualify=yes > > canreinvite=no > > directmedia=no > > host=dynamic > > context=FAX-T38 > > faxdetect=no > > > > Larry. > > > > > > > It's still not working but not throwing any errors at all. I've checked > that the fax machine itself works (actually hoping that it wouldn't and > that would explain my problem) by plugging it into a phone line and it > does. > Does anyone know if these settings are correct and if not, how to change them? CLI> fax show settings FAX For Asterisk Settings: ECM: Enabled Status Events: On Minimum Bit Rate: 2400 Maximum Bit Rate: 14400 Modem Modulations Allowed: V17,V27,V29 FAX Technology Modules: DIGIUM (Digium FAX Driver) Settings: Maximum T.38 Packet Delay: 800 T.38 Session Packet Capture: Off G.711 Session Audio Capture: Off Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] German voice recognition
To the best of my knowledge, your best options, not necessarily in order are: 1. Vestec ASR 2. Lumenvox ASR 3. google ASR (there was a good post in February about how to use this) 4. Sphynx ASR Options 1 and 2 are/were "recommended" by Digium. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Monday, March 12, 2012 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] German voice recognition Hi, I am looking (for the best) solution to recognize *german* words or simple phrases with a given number of words (eins, zwei drei etc. or hauptmenü, zurück etc.). Can somebody give me a good link? Can I find external service providers who can be accessed via ASR()? Best regards, -Thorsten- -- _ -- 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] German voice recognition
Hi, I am looking (for the best) solution to recognize *german* words or simple phrases with a given number of words (eins, zwei drei etc. or hauptmenü, zurück etc.). Can somebody give me a good link? Can I find external service providers who can be accessed via ASR()? Best regards, -Thorsten- -- _ -- 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] Problem with ReceiveFax
On Mon, 2012-03-12 at 09:11 -0400, Bryant Zimmerman wrote:
> Looking at the information you have sent in this posting in certainly
> appears that the 'f' option has indeed helped however you have
> another
> matter to overcome.
>
> You may wish to set the following parameters in your peer
> configuration
> for 588.
>
> ignoresdpversion=yes
This one passed me by, thanks for the info
> directmedia=no
>
> I use Spandsp FAX successfully.
>
> I have also attached an analogue Fax Modem to the FXS port on an
> SPA8800
> and an HT-502 and have been able to receive faxes on them when I last
> tested, the SPA8800 like the HT-502 are now in storage.
>
> Looking at the User Guide for the Vigor 2701 there is an option in
> the
> configuration to enable T.38 mode, did you enable it?
Foolishly I hadn't and feel very stupid about that!
>
>
> In my sip.conf I have the following;
>
> [general]
> .
> .
> .
> faxdetect=cng
> t38pt_udptl=yes,redundancy,maxdatagram=400
tried this and ,fec,
> ;t38pt_usertpsource=yes
had this already set
> .
> .
> .
> [903]
> ; Cisco SPA8800 FXS Port 3
> ; Grandstream HT502 FXS Port 1
> ; Analogue FAX Modem attached
> type=friend
> defaultuser=903
> secret=you_guessed_it
> call-limit=2
> disallow=g722
> transport=udp
> qualify=yes
> canreinvite=no
> directmedia=no
> host=dynamic
> context=FAX-T38
> faxdetect=no
>
> Larry.
>
>
>
It's still not working but not throwing any errors at all. I've checked
that the fax machine itself works (actually hoping that it wouldn't and
that would explain my problem) by plugging it into a phone line and it
does.
Now getting the following
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [200@local:1] Goto("SIP/588-000b", "fax-in,s,1")
-- Goto (fax-in,s,1)
-- Executing [s@fax-in:1] Answer("SIP/588-000b", "")
-- Executing [s@fax-in:2] Wait("SIP/588-000b", "3")
-- Executing [s@fax-in:3] Set("SIP/588-000b",
"FAXFILE=/tmp/fax-588-20120312-144800.tiff")
-- Executing [s@fax-in:4] ReceiveFAX("SIP/588-000b",
"/tmp/fax-588-20120312-144800.tiff,f")
-- Channel 'SIP/588-000b' receiving FAX
'/tmp/fax-588-20120312-144800.tiff'
-- Channel 'SIP/588-000b' FAX session '11' started
-- FAX handle 0: [ 000.000190 ], STAT_EVT_STRT_RX st: IDLE
rt: IDLENSRX
-- FAX handle 0: [ 000.000277 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY
rt: RRDYNHRY
-- FAX handle 0: [ 000.000308 ], P30EVN_RECEIVE_STARTED
-- FAX handle 0: [ 000.000341 ], STAT_INFO_CSI
-- FAX handle 0: [ 000.000377 ], STAT_INFO_DIS
-- FAX handle 0: [ 002.174255 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP
rt: WDSRNT21
-- FAX handle 0: [ 005.503113 ], STAT_EVT_T4_EXPst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 005.503170 ], STAT_INFO_CSI
-- FAX handle 0: [ 005.503202 ], STAT_INFO_DIS
-- FAX handle 0: [ 007.669035 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP
rt: WDSRNT21
-- FAX handle 0: [ 011.001891 ], STAT_EVT_T4_EXPst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 011.001943 ], STAT_INFO_CSI
-- FAX handle 0: [ 011.001974 ], STAT_INFO_DIS
-- FAX handle 0: [ 013.169811 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP
rt: WDSRNT21
-- FAX handle 0: [ 016.502670 ], STAT_EVT_T4_EXPst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 016.502723 ], STAT_INFO_CSI
-- FAX handle 0: [ 016.502753 ], STAT_INFO_DIS
-- FAX handle 0: [ 018.668594 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP
rt: WDSRNT21
-- FAX handle 0: [ 022.001447 ], STAT_EVT_T4_EXPst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 022.001496 ], STAT_INFO_CSI
-- FAX handle 0: [ 022.001527 ], STAT_INFO_DIS
-- FAX handle 0: [ 024.169368 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP
rt: WDSRNT21
-- FAX handle 0: [ 027.503224 ], STAT_EVT_T4_EXPst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 027.503282 ], STAT_INFO_CSI
-- FAX handle 0: [ 027.503313 ], STAT_INFO_DIS
-- FAX handle 0: [ 029.669146 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP
rt: WDSRNT21
-- FAX handle 0: [ 033.002003 ], STAT_EVT_T4_EXPst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 033.002054 ], STAT_INFO_CSI
-- FAX handle 0: [ 033.002084 ], STAT_INFO_DIS
-- FAX handle 0: [ 035.169922 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP
rt: WDSRNT21
-- FAX handle 0: [ 038.502780 ], STAT_EVT_T4_EXPst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 038.502835 ], STAT_INFO_CSI
-- FAX handle 0: [ 038.502866 ], STAT_INFO_DIS
-- FAX handle 0: [ 040.001719 ], STAT_EVT_T1_EXPst: WT_DIS_RSP
Re: [asterisk-users] DAHDISendCallreroutingFacility
for using this application it is enough to put this in chan_dahdi.conf ? ... context=incoming facilityenable=yes transfer=yes --- On Sat, 3/10/12, Karsten Wemheuer wrote: From: Karsten Wemheuer Subject: Re: [asterisk-users] DAHDISendCallreroutingFacility To: "Asterisk Users Mailing List - Non-Commercial Discussion" Date: Saturday, March 10, 2012, 12:05 PM Hi, Am Samstag, den 10.03.2012, 08:42 -0800 schrieb Mehdi Shirazi: > Hi > I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) > I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI > Already installed). > according to > https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 > Asterisk 1.8 include this application but I cannot see it with "core show > applications" > Do I need to install mISDN or other modules for using that ? > > Regards > M.Shirazi No You don't need mISDN or other modules. DAHDISendCallreroutingFacility is part of chan_dahdi. But as far as I know Asterisk 1.8.7 had problems with this application. Try using at least 1.8.8 (1.8.10.0 is currently the stable version of 1.8 release). HTH, Karsten -- _ -- 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] Multi-record SRV records
Asterisk's peer matching changes between releases. Your best bet is to try it and see. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asgaroth Sent: Monday, March 12, 2012 10:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multi-record SRV records Hi, On 12/03/12 13:48, Eric Wieling wrote: > Have you tried permit/deny on the peer? > No, I've not tried this, however, will those entries be checked if the inbound call is not matched against the peer that those settings matched with? I'll have to research this option a little more. -- _ -- 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] Multi-record SRV records
Hi, On 12/03/12 13:48, Eric Wieling wrote: Have you tried permit/deny on the peer? No, I've not tried this, however, will those entries be checked if the inbound call is not matched against the peer that those settings matched with? I'll have to research this option a little more. -- _ -- 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] Multi-record SRV records
Greetings back :) Greetings, That will depend on my SIP providers, I'm not sure if they swap their IP's indeed,and send calls down my way with alternating IP's, perhaps they're "smart" enough to only send calls down my way with the same IP that was bound with the registration request to begin with. (This is a shot in the dark, I might be saying nonsense). This is just it, if I do a SIP trace, on what is happening, the registrations are bouncing between the 2 addresses, and the inbound call comes in from the last address that my registration was against. What appears to be happening though is that my peer definition's DNS address does not get updated with the last address registered against. For example: [a] register against example.com's IP 10.10.0.1 [b] sip peer [example] has resolved the host=example.com to be 10.10.0.1 [c] call comes in from registered address 10.10.0.1, this works fine [d] i then re/refresh-register against 10.10.0.2 [e] sip peer [example] still has the initial resolution of host=example.com to be 10.10.0.1 [f] call comes in via 10.10.0.2 and this fails at my pbx. One of my providers, while troubleshooting an issue for inbound calls, offered me to use an IP rather then SRV to register - because "Asterisk is bad with using SRV's" , but the issues turned out to be diffrent. Can you recall what the issue may have been? I'm sorry I cannot shed much light on this one without monitoring my PBX's peers at this moment. No problem, thanks for the info/pointers thus far. -- _ -- 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] Multi-record SRV records
Have you tried permit/deny on the peer? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Guy Gold Sent: Monday, March 12, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multi-record SRV records On Mon,Mar 12 01:33:PM, Asgaroth wrote: > Are you saying that the above scenario is working for you for incoming > calls? Greetings, That will depend on my SIP providers, I'm not sure if they swap their IP's indeed,and send calls down my way with alternating IP's, perhaps they're "smart" enough to only send calls down my way with the same IP that was bound with the registration request to begin with. (This is a shot in the dark, I might be saying nonsense). One of my providers, while troubleshooting an issue for inbound calls, offered me to use an IP rather then SRV to register - because "Asterisk is bad with using SRV's" , but the issues turned out to be diffrent. I'm sorry I cannot shed much light on this one without monitoring my PBX's peers at this moment. 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 -- _ -- 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] Multi-record SRV records
On Mon,Mar 12 01:33:PM, Asgaroth wrote: > Are you saying that the above scenario is working for you for > incoming calls? Greetings, That will depend on my SIP providers, I'm not sure if they swap their IP's indeed,and send calls down my way with alternating IP's, perhaps they're "smart" enough to only send calls down my way with the same IP that was bound with the registration request to begin with. (This is a shot in the dark, I might be saying nonsense). One of my providers, while troubleshooting an issue for inbound calls, offered me to use an IP rather then SRV to register - because "Asterisk is bad with using SRV's" , but the issues turned out to be diffrent. I'm sorry I cannot shed much light on this one without monitoring my PBX's peers at this moment. 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
Re: [asterisk-users] Multi-record SRV records
Hi, I can report, at least from my side, on not having issue with SRV records, with Multiple SIP providers. Did not have issues at least with the 1.8 branch. Are you PBX's doing SRV lookups ? Yes, I do have the srv_lookup parameter set to "yes". What appears to be happening is that when the are 2+ definitions in the SRV record then asterisk will only use one of those IP addresses for the peer definition. So when a call comes in from the other IP address, then it does not match the peer. For example: [a] SRV domain "example.com" provides 2 addresses for it's proxies, 10.10.0.1 and 10.10.0.2 [b] My host= line for my peer definition is defined as "host=example.com" for peer [example] [c] When I start asterisk, or perform a sip reload, the "sip show peers" command only has one of 10.10.0.1 or 10.10.0.2 as the address defined for this peer. [d] When the peer has IP 10.10.0.1 as the peer address then incoming calls from provider address 10.10.0.2 fail and 10.10.0.1 are successful. [e] When the peer has IP 10.10.0.2 as the peer address then incoming calls from provider address 10.10.0.1 fail and 10.10.0.2 are successful. Registration and outbound calls are working as expected. Are you saying that the above scenario is working for you for incoming calls? -- _ -- 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] Problem with ReceiveFax
On 12/03/2012 5:27 PM, Ishfaq Malik wrote:
> On Fri, 2012-03-02 at 15:32 +, Ishfaq Malik wrote:
>
> I've tried this with the f option on receiveFax but it still isn't
> working. Any insight would be helpful as this is driving me a bit potty
>
> == Using UDPTL CoS mark 5
> == Using SIP RTP CoS mark 5
> -- Executing [200@local:1] Goto("SIP/588-", "fax-in,s,1")
> -- Goto (fax-in,s,1)
> -- Executing [s@fax-in:1] Answer("SIP/588-", "")
> -- Executing [s@fax-in:2] Wait("SIP/588-0000", "3")
> -- Executing [s@fax-in:3] Set("SIP/588-",
"FAXFILE=/tmp/fax-588-20120312-092231.tiff")
> -- Executing [s@fax-in:4] ReceiveFAX("SIP/588-",
"/tmp/fax-588-20120312-092231.tiff,f")
> -- Channel 'SIP/588-' receiving FAX
'/tmp/fax-588-20120312-092231.tiff'
> [2012-03-12 09:22:34] WARNING[4529]: res_fax.c:1508 receivefax_t38_init:
channel 'SIP/588-' refused to negotiate T.38
> -- Channel 'SIP/588-' FAX session '0' started
> [2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8740 process_sdp:
Unsupported SDP media type in offer: audio 0 RTP/AVP 8
> [2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8827 process_sdp: Failing
due to no acceptable offer found
> [2012-03-12 09:22:44] ERROR[4529]: res_fax.c:1344 generic_fax_exec:
channel 'SIP/588-' FAX session '0' failure, reason: 'fax session
timed-out' (TIMEOUT)
> -- Executing [s@fax-in:5] Hangup("SIP/588-", "")
> == Spawn extension (fax-in, s, 5) exited non-zero on 'SIP/588-'
> -- Executing [h@fax-in:1] NoOP("SIP/588-", "### FAXSTATUS:
FAILED")
> -- Executing [h@fax-in:2] NoOP("SIP/588-", "### FAXERROR:
TIMEOUT")
> -- Executing [h@fax-in:3] NoOP("SIP/588-", "### FAXMODE: ")
> -- Executing [h@fax-in:4] NoOP("SIP/588-0000", "### FAXPAGES: 0")
> -- Executing [h@fax-in:5] NoOP("SIP/588-", "### FAXBITRATE: ")
> -- Executing [h@fax-in:6] NoOP("SIP/588-", "### FAXRESOLUTION:
")
> -- Executing [h@fax-in:7] NoOP("SIP/588-", "### REMOTESTATIONID:
")
> -- Executing [h@fax-in:8] System("SIP/588-", "mail -s FaxToEmail
i...@-net.co.uk< /tmp/fax-588-20120312-092231.tiff")
> -- FAX handle 0: [ 040.001588 ], entering CLOSING state
> -- Channel 'SIP/588-' FAX session '0' is complete, result:
'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution:
'unknown', transfer rate: '2400', remoteSID: ''
>
> Thanks in Advance
>
Looking at the information you have sent in this posting in certainly
appears that the 'f' option has indeed helped however you have another
matter to overcome.
You may wish to set the following parameters in your peer configuration
for 588.
ignoresdpversion=yes
directmedia=no
I use Spandsp FAX successfully.
I have also attached an analogue Fax Modem to the FXS port on an SPA8800
and an HT-502 and have been able to receive faxes on them when I last
tested, the SPA8800 like the HT-502 are now in storage.
Looking at the User Guide for the Vigor 2701 there is an option in the
configuration to enable T.38 mode, did you enable it?
In my sip.conf I have the following;
[general]
.
.
.
faxdetect=cng
t38pt_udptl=yes,redundancy,maxdatagram=400
;t38pt_usertpsource=yes
.
.
.
[903]
; Cisco SPA8800 FXS Port 3
; Grandstream HT502 FXS Port 1
; Analogue FAX Modem attached
type=friend
defaultuser=903
secret=you_guessed_it
call-limit=2
disallow=g722
transport=udp
qualify=yes
canreinvite=no
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
Larry.
---
Try the "ReceiveFax" "F" option and see if it makes a difference we have
had great success with it.
Thanks
Bryant
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Re: [asterisk-users] Problem with ReceiveFax
On 03/12/2012 04:27 AM, Ishfaq Malik wrote:
I've tried this with the f option on receiveFax but it still isn't
working. Any insight would be helpful as this is driving me a bit potty
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [200@local:1] Goto("SIP/588-", "fax-in,s,1")
-- Goto (fax-in,s,1)
-- Executing [s@fax-in:1] Answer("SIP/588-", "")
-- Executing [s@fax-in:2] Wait("SIP/588-", "3")
-- Executing [s@fax-in:3] Set("SIP/588-",
"FAXFILE=/tmp/fax-588-20120312-092231.tiff")
-- Executing [s@fax-in:4] ReceiveFAX("SIP/588-",
"/tmp/fax-588-20120312-092231.tiff,f")
-- Channel 'SIP/588-' receiving FAX
'/tmp/fax-588-20120312-092231.tiff'
[2012-03-12 09:22:34] WARNING[4529]: res_fax.c:1508 receivefax_t38_init:
channel 'SIP/588-' refused to negotiate T.38
-- Channel 'SIP/588-' FAX session '0' started
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8740 process_sdp: Unsupported
SDP media type in offer: audio 0 RTP/AVP 8
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8827 process_sdp: Failing due
to no acceptable offer found
[2012-03-12 09:22:44] ERROR[4529]: res_fax.c:1344 generic_fax_exec: channel
'SIP/588-' FAX session '0' failure, reason: 'fax session timed-out'
(TIMEOUT)
When the T.38 re-INVITE was rejected by your SIP peer, they sent an SDP
offer with the audio stream set to port number zero ('0'). This means
the audio stream is not active, and thus cannot be used.
--
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
--
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Re: [asterisk-users] Problem with ReceiveFax
On 12/03/2012 5:27 PM, Ishfaq Malik wrote:
On Fri, 2012-03-02 at 15:32 +, Ishfaq Malik wrote:
I've tried this with the f option on receiveFax but it still isn't
working. Any insight would be helpful as this is driving me a bit potty
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [200@local:1] Goto("SIP/588-", "fax-in,s,1")
-- Goto (fax-in,s,1)
-- Executing [s@fax-in:1] Answer("SIP/588-", "")
-- Executing [s@fax-in:2] Wait("SIP/588-", "3")
-- Executing [s@fax-in:3] Set("SIP/588-",
"FAXFILE=/tmp/fax-588-20120312-092231.tiff")
-- Executing [s@fax-in:4] ReceiveFAX("SIP/588-",
"/tmp/fax-588-20120312-092231.tiff,f")
-- Channel 'SIP/588-' receiving FAX
'/tmp/fax-588-20120312-092231.tiff'
[2012-03-12 09:22:34] WARNING[4529]: res_fax.c:1508 receivefax_t38_init:
channel 'SIP/588-' refused to negotiate T.38
-- Channel 'SIP/588-' FAX session '0' started
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8740 process_sdp: Unsupported
SDP media type in offer: audio 0 RTP/AVP 8
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8827 process_sdp: Failing due
to no acceptable offer found
[2012-03-12 09:22:44] ERROR[4529]: res_fax.c:1344 generic_fax_exec: channel
'SIP/588-' FAX session '0' failure, reason: 'fax session timed-out'
(TIMEOUT)
-- Executing [s@fax-in:5] Hangup("SIP/588-", "")
== Spawn extension (fax-in, s, 5) exited non-zero on 'SIP/588-'
-- Executing [h@fax-in:1] NoOP("SIP/588-", "### FAXSTATUS:
FAILED")
-- Executing [h@fax-in:2] NoOP("SIP/588-", "###FAXERROR:
TIMEOUT")
-- Executing [h@fax-in:3] NoOP("SIP/588-", "### FAXMODE: ")
-- Executing [h@fax-in:4] NoOP("SIP/588-", "###FAXPAGES:
0")
-- Executing [h@fax-in:5] NoOP("SIP/588-0000", "### FAXBITRATE: ")
-- Executing [h@fax-in:6] NoOP("SIP/588-", "### FAXRESOLUTION: ")
-- Executing [h@fax-in:7] NoOP("SIP/588-", "### REMOTESTATIONID: ")
-- Executing [h@fax-in:8] System("SIP/588-", "mail -s FaxToEmail
i...@-net.co.uk< /tmp/fax-588-20120312-092231.tiff")
-- FAX handle 0: [ 040.001588 ], entering CLOSING state
-- Channel 'SIP/588-' FAX session '0' is complete, result:
'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown',
transfer rate: '2400', remoteSID: ''
Thanks in Advance
Looking at the information you have sent in this posting in certainly
appears that the 'f' option has indeed helped however you have another
matter to overcome.
You may wish to set the following parameters in your peer configuration
for 588.
ignoresdpversion=yes
directmedia=no
I use Spandsp FAX successfully.
I have also attached an analogue Fax Modem to the FXS port on an SPA8800
and an HT-502 and have been able to receive faxes on them when I last
tested, the SPA8800 like the HT-502 are now in storage.
Looking at the User Guide for the Vigor 2701 there is an option in the
configuration to enable T.38 mode, did you enable it?
In my sip.conf I have the following;
[general]
.
.
.
faxdetect=cng
t38pt_udptl=yes,redundancy,maxdatagram=400
;t38pt_usertpsource=yes
.
.
.
[903]
; Cisco SPA8800 FXS Port 3
; Grandstream HT502 FXS Port 1
; Analogue FAX Modem attached
type=friend
defaultuser=903
secret=you_guessed_it
call-limit=2
disallow=g722
transport=udp
qualify=yes
canreinvite=no
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
Larry.
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Re: [asterisk-users] Problem with ReceiveFax
Sess-Timers : Accept
> Sess-Refresh : uas
> Sess-Expires : 1800 secs
> Min-Sess : 90 secs
> RTP Engine : asterisk
> Parkinglot :
> Use Reason : No
> Encryption : No
>
>
> Here's the relevant sip settings
>
> T.38 support: Yes
> T.38 EC mode: FEC
> T.38 MaxDtgrm: -1
>
>
> here's the fax settings
>
> fax show settings
> FAX For Asterisk Settings:
> ECM: Enabled
> Status Events: On
> Minimum Bit Rate: 2400
> Maximum Bit Rate: 14400
> Modem Modulations Allowed: V17,V27,V29
>
>
> FAX Technology Modules:
>
> DIGIUM (Digium FAX Driver) Settings:
> Maximum T.38 Packet Delay: 800
> T.38 Session Packet Capture: Off
> G.711 Session Audio Capture: Off
>
> fax show stats
>
> FAX Statistics:
> ---
>
> Current Sessions : 0
> Reserved Sessions: 0
> Transmit Attempts: 0
> Receive Attempts : 9
> Completed FAXes : 0
> Failed FAXes : 9
>
> Digium G.711
> Licensed Channels: 1
> Max Concurrent : 0
> Success : 0
> Switched to T.38 : 0
> Canceled : 0
> No FAX : 0
> Partial : 0
> Negotiation Failed : 0
> Train Failure : 0
> Protocol Error : 0
> IO Partial : 0
> IO Fail : 0
>
> Digium T.38
> Licensed Channels: 1
> Max Concurrent : 0
> Success : 0
> Canceled : 0
> No FAX : 0
> Partial : 0
> Negotiation Failed : 0
> Train Failure: 0
> Protocol Error : 0
> IO Partial : 0
> IO Fail : 0
>
I've tried this with the f option on receiveFax but it still isn't
working. Any insight would be helpful as this is driving me a bit potty
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [200@local:1] Goto("SIP/588-", "fax-in,s,1")
-- Goto (fax-in,s,1)
-- Executing [s@fax-in:1] Answer("SIP/588-", "")
-- Executing [s@fax-in:2] Wait("SIP/588-", "3")
-- Executing [s@fax-in:3] Set("SIP/588-",
"FAXFILE=/tmp/fax-588-20120312-092231.tiff")
-- Executing [s@fax-in:4] ReceiveFAX("SIP/588-",
"/tmp/fax-588-20120312-092231.tiff,f")
-- Channel 'SIP/588-' receiving FAX
'/tmp/fax-588-20120312-092231.tiff'
[2012-03-12 09:22:34] WARNING[4529]: res_fax.c:1508 receivefax_t38_init:
channel 'SIP/588-' refused to negotiate T.38
-- Channel 'SIP/588-' FAX session '0' started
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8740 process_sdp: Unsupported
SDP media type in offer: audio 0 RTP/AVP 8
[2012-03-12 09:22:34] WARNING[3763]: chan_sip.c:8827 process_sdp: Failing due
to no acceptable offer found
[2012-03-12 09:22:44] ERROR[4529]: res_fax.c:1344 generic_fax_exec: channel
'SIP/588-' FAX session '0' failure, reason: 'fax session timed-out'
(TIMEOUT)
-- Executing [s@fax-in:5] Hangup("SIP/588-", "")
== Spawn extension (fax-in, s, 5) exited non-zero on 'SIP/588-'
-- Executing [h@fax-in:1] NoOP("SIP/588-", "### FAXSTATUS:
FAILED")
-- Executing [h@fax-in:2] NoOP("SIP/588-", "###FAXERROR:
TIMEOUT")
-- Executing [h@fax-in:3] NoOP("SIP/588-", "### FAXMODE: ")
-- Executing [h@fax-in:4] NoOP("SIP/588-", "###FAXPAGES: 0")
-- Executing [h@fax-in:5] NoOP("SIP/588-", "### FAXBITRATE: ")
-- Executing [h@fax-in:6] NoOP("SIP/588-", "### FAXRESOLUTION: ")
-- Executing [h@fax-in:7] NoOP("SIP/588-", "### REMOTESTATIONID: ")
-- Executing [h@fax-in:8] System("SIP/588-", "mail -s FaxToEmail
i...@-net.co.uk < /tmp/fax-588-20120312-092231.tiff")
-- FAX handle 0: [ 040.001588 ], entering CLOSING state
-- Channel 'SIP/588-' FAX session '0' is complete, result: 'FAILED'
(FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer
rate: '2400', remoteSID: ''
Thanks in Advance
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] asterisk 1.8.9.2 channel.c: Channel allocation failed
Yes, I can give a higher ulimit, but I want to know why there were so much fd's. As I found out yesterday, the reason of running out of available file descriptors was: Some Agents in the Callcenter made ChanSpy on several Calls, but they didn't stop spying with *-key, just hangup the phone, and keep starting new chanspy's. So Asterisk keeps the Channel open when user just hangs up the phone. My ChanSpy context looks like this: [spy] exten => *9,1,Answer exten => *9,2,ChanSpy(SIP) exten => *9,3,Hangup exten => h,1,Hangup In the CLI there were several ChanSpy listed, although nobody was spying. I tried to remove the channels with: hangup request SIP/channelname but nothing happend, I was not able to remove the Channels, only a restart did the trick. yours christian -- _ -- 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
