Re: [asterisk-users] cli command show codecs
Bruce, that's exactly the command I was looking for. Thx a ton. Sans On Thu, Sep 1, 2011 at 12:17 AM, Bruce B wrote: > "sip show channels" is the command you are looking for. > > > On Wed, Aug 31, 2011 at 2:45 PM, RSCL Mumbai wrote: > >> asterisk -rx "core show channels verbose" does not provide transcoding >> details. >> >> Unless I have missed something. >> >> Sans >> >> >> On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas wrote: >> >>> Core show channels verbose is probably your best bet. I think the answer >>> also depends on your * version. >>> >>> ** ** >>> >>> *From:* asterisk-users-boun...@lists.digium.com [mailto: >>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai >>> *Sent:* Wednesday, August 31, 2011 10:44 AM >>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >>> *Subject:* [asterisk-users] cli command show codecs >>> >>> ** ** >>> >>> Hi, >>> >>> Is there a CLI command which will tell me the codec used for active calls >>> and if transcoding is happening ? >>> >>> Thx >>> Sans >>> >>> -- >>> _ >>> -- 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 >> > > > -- > _ > -- 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] cli command show codecs
Hi, Does audio files have codec formats? I simply convert all my audios (MOH, accouncements) to .wav format, 16bit, 11kHz (I believe this is the best format for asterisk). I am new to this and may be incorrect. Going forward, (a) How can I check the codec format of my announcements, MOH ? (b) How can I record/convert announcements, MoH etc to a particular format ? I believe its a good idea to prevent transcoding and save CPU overheads. Thx Sans On Thu, Sep 1, 2011 at 11:39 AM, Bruce B wrote: > if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your > IVR announcement is not recorded in g729 and you see g729 on the channel > when you call into IVR then it's transcoding as well. > > > On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling wrote: > >> Assuming SIP "sip show channels" will show you which codec is used for >> each call leg. However it does not track transcoding. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai >> Sent: Wednesday, August 31, 2011 2:45 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] cli command show codecs >> >> asterisk -rx "core show channels verbose" does not provide transcoding >> details. >> >> Unless I have missed something. >> >> Sans >> >> >> >> On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas >> wrote: >> >> >>Core show channels verbose is probably your best bet. I think the >> answer also depends on your * version. >> >> >> >>From: asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai >>Sent: Wednesday, August 31, 2011 10:44 AM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: [asterisk-users] cli command show codecs >> >> >> >>Hi, >> >>Is there a CLI command which will tell me the codec used for active >> calls and if transcoding is happening ? >> >>Thx >>Sans >> >> >>-- >> >> _ >>-- 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 >> > > > -- > _ > -- 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] Faxes suddenly failing
You should try disabling ecm.. put the following in res_fax.conf ; Enable/disable T.30 ECM (error correction mode) by default. ; Default: Enabled ecm=no On Thu, Sep 1, 2011 at 1:27 AM, C F wrote: > I think you should change the subject line to: > Faxes suddenly worked for 2 weeks. > > On Wed, Aug 31, 2011 at 3:49 PM, Tim King > wrote: > > I realize that faxing is not great with voip but here is my confusion. I > > have been working on a web based fax system for 2 weeks. During this time > I > > have sent over 100 2 page faxes without any errors. Now today as things > are > > finally completed I can not seem to get any fax to go through unless it > is a > > 1 page cover only. Anyone able to tell the issue from this debug output? > > > >-- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started > > -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE > > rt: IDLENSRX > > -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: > WT_RX_HW_RDY > > rt: RRDYNHRY > > -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED > > -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI > > -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 000.091837 ], stack sent 5 frames (100 ms) of energy. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 000.160248 ], stack sent 3 frames (60 ms) of silence. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 000.960201 ], channel sent 48 frames (960 ms) of silence. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 000.979464 ], channel sent 1 frames (20 ms) of energy. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 003.157848 ], stack sent 150 frames (3000 ms) of energy. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 003.219814 ], stack sent 3 frames (60 ms) of silence. > > -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: > WT_DIS_RSP > > rt: WDSRNT21 > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 005.579811 ], stack sent 118 frames (2360 ms) of energy. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 006.481179 ], channel sent 275 frames (5500 ms) of silence. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 007.801045 ], channel sent 66 frames (1320 ms) of energy. > > -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP > > -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: > WT_DIS_RSP > > rt: NT4X > > -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: > WT_DIS_RSP > > rt: UNEXPECT > > -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: > WT_DIS_RSP > > rt: RXXXNFRX > > -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI > > -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 011.152812 ], stack sent 279 frames (5580 ms) of silence. > > -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: > WT_DIS_RSP > > rt: WDSRNT21 > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 013.471827 ], stack sent 116 frames (2320 ms) of energy. > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 014.260642 ], channel sent 323 frames (6460 ms) of silence. > > -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 016.460661 ], channel sent 110 frames (2200 ms) of energy. > > -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS > > -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: > WT_DIS_RSP > > rt: WDSRNDCS > > -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 > > -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH > > -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 > > -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 > > -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM > > -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: > WT_DIS_RSP > > rt: WDSRNSWE > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 016.540315 ], channel sent 4 frames (80 ms) of silence. > > -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: > RCV_ECM_TRN > > rt: UNEXPECT > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 019.700543 ], channel sent 158 frames (3160 ms) of energy. > > -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: > RCV_ECM_TRN > > rt: RTCFNERT > > -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 019.912812 ], stack sent 322 frames (6440 ms) of silence. > > -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE st: > RCV_ECM_STRT > > rt: RECMNT21 > >> Channel 'SIP/MyVoipProvider-0046' fax session '12', [ > > 021.278809 ], stack sent 68 f
Re: [asterisk-users] Faxes suddenly failing
That debug looks cool but I have no idea what it means. If you are using T.38, turn it off, and do audio fax, recorded with MixMonitor. When you can hear the audio of the fax hopefully you will be able to tell what's going on, and if you're lucky it's something specific to the particular kind of testing you are doing. I don't think it's an exaggeration to say there have been hundreds of posts over the last few years about broken T.38. Avoid it in favor of traditional audio faxing. Even if you can control both endpoints, there's just so much that can go wrong when faxing over voip. If you need this to be 'reliable faxing', you should seriously consider doing your faxes over copper. If you cannot afford that, it should be a top-tier voip provider on a dedicated line, where you will not be starving for bandwidth, and you should never compress the audio on those calls. On Wed, Aug 31, 2011 at 7:27 PM, C F wrote: > I think you should change the subject line to: > Faxes suddenly worked for 2 weeks. > > On Wed, Aug 31, 2011 at 3:49 PM, Tim King wrote: >> I realize that faxing is not great with voip but here is my confusion. I >> have been working on a web based fax system for 2 weeks. During this time I >> have sent over 100 2 page faxes without any errors. Now today as things are >> finally completed I can not seem to get any fax to go through unless it is a >> 1 page cover only. Anyone able to tell the issue from this debug output? >> >> -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started >> -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE >> rt: IDLENSRX >> -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY >> rt: RRDYNHRY >> -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED >> -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI >> -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.091837 ], stack sent 5 frames (100 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.160248 ], stack sent 3 frames (60 ms) of silence. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.960201 ], channel sent 48 frames (960 ms) of silence. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.979464 ], channel sent 1 frames (20 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 003.157848 ], stack sent 150 frames (3000 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 003.219814 ], stack sent 3 frames (60 ms) of silence. >> -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP >> rt: WDSRNT21 >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 005.579811 ], stack sent 118 frames (2360 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 006.481179 ], channel sent 275 frames (5500 ms) of silence. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 007.801045 ], channel sent 66 frames (1320 ms) of energy. >> -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP >> -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP >> rt: NT4X >> -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP >> rt: UNEXPECT >> -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_END st: WT_DIS_RSP >> rt: RXXXNFRX >> -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI >> -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 011.152812 ], stack sent 279 frames (5580 ms) of silence. >> -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP >> rt: WDSRNT21 >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 013.471827 ], stack sent 116 frames (2320 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 014.260642 ], channel sent 323 frames (6460 ms) of silence. >> -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 016.460661 ], channel sent 110 frames (2200 ms) of energy. >> -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS >> -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP >> rt: WDSRNDCS >> -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 >> -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH >> -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 >> -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 >> -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM >> -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECM st: WT_DIS_RSP >> rt: WDSRNSWE >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 016.540315 ], channel sent 4 frames (80 ms) of silence. >> -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN >> rt: UNE
[asterisk-users] Asterisk 1.8.3.3 T.38 Gateway
I have found numerous claims that 1.8 can do T.38 gateway with a patch, however I am yet to find the patch or any instructions on implementing it. Anyone have a link? -- _ -- 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 1.8.3.3 T.38 Gateway
On 1/09/2011 7:04 PM, Tim King wrote: I have found numerous claims that 1.8 can do T.38 gateway with a patch, however I am yet to find the patch or any instructions on implementing it. Anyone have a link? https://issues.asterisk.org/view.php?id=13405 -- _ -- 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] Simultaneous ring on Soft phone and Desk phone.
Hi, In my Office all our users have a Desk phone. Some of the users who are using laptops have a Soft phone too along with their Desk phone. Right now we are using two different extensions for their desk and soft phones. Is it possible to have simultaneous ring for both the extensions (ie. soft phone and desk phone). I tried using " Dial(SIP/desk & SIP/soft) " and it works fine when both the phones are online. But when the soft phone goes offline, none of the phones rings and it says that the user is unavailable. When I tried the above method with both the extensions on the same server, both the phones where ringing simultaneously. But in my case both the extensions are on two different servers and we use SIP trunk to dial between them. We have users located at 7 different Office locations, and each Office has its own PBX for desk phone. All the soft phone extensions are registering to another server. Has anyone setup a similar scenario before.?? Thanks, Najim. -- _ -- 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] Faxes suddenly failing
Hi Tim, On 09/01/2011 03:49 AM, Tim King wrote: I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During this time I have sent over 100 2 page faxes without any errors. Now today as things are finally completed I can not seem to get any fax to go through unless it is a 1 page cover only. Anyone able to tell the issue from this debug output? -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.091837 ], stack sent 5 frames (100 ms) of energy. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.160248 ], stack sent 3 frames (60 ms) of silence. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.960201 ], channel sent 48 frames (960 ms) of silence. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.979464 ], channel sent 1 frames (20 ms) of energy. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.157848 ], stack sent 150 frames (3000 ms) of energy. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.219814 ], stack sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 005.579811 ], stack sent 118 frames (2360 ms) of energy. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 006.481179 ], channel sent 275 frames (5500 ms) of silence. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 007.801045 ], channel sent 66 frames (1320 ms) of energy. -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP rt: NT4X -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt: UNEXPECT -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: WT_DIS_RSP rt: RXXXNFRX -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 011.152812 ], stack sent 279 frames (5580 ms) of silence. -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 013.471827 ], stack sent 116 frames (2320 ms) of energy. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 014.260642 ], channel sent 323 frames (6460 ms) of silence. -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.460661 ], channel sent 110 frames (2200 ms) of energy. -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: WT_DIS_RSP rt: WDSRNSWE > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.540315 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.700543 ], channel sent 158 frames (3160 ms) of energy. -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: RCV_ECM_TRN rt: RTCFNERT -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.912812 ], stack sent 322 frames (6440 ms) of silence. -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE st: RCV_ECM_STRT rt: RECMNT21 > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 021.278809 ], stack sent 68 frames (1360 ms) of energy. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 022.261160 ], channel sent 128 frames (2560 ms) of silence. -- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_STRT rt: RECMNSRI -- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C -- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START -- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.102000 ], channel sent 442 frames (8840 ms) of energy. > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.160415 ], channel sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 031.160196 ], STAT_EVT_RX_IMG_
Re: [asterisk-users] cli command show codecs
Maybe this will be better than my first answer – Audio files do indeed have codec formats. If you are in a particular codec (say G729), Playback/Background and MOH will search for files that match the codec format first, then transcode WAV/GSM/whatever you have to that format if it isn’t found. Ideally, you want to have a copy of each codec you can play for all sounds and MOH. Each of the “canned sounds” comes in each codec format (you pick the ones you want when you do make menuselect). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, September 01, 2011 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs Hi, Does audio files have codec formats? I simply convert all my audios (MOH, accouncements) to .wav format, 16bit, 11kHz (I believe this is the best format for asterisk). I am new to this and may be incorrect. Going forward, (a) How can I check the codec format of my announcements, MOH ? (b) How can I record/convert announcements, MoH etc to a particular format ? I believe its a good idea to prevent transcoding and save CPU overheads. Thx Sans On Thu, Sep 1, 2011 at 11:39 AM, Bruce B wrote: if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your IVR announcement is not recorded in g729 and you see g729 on the channel when you call into IVR then it's transcoding as well. On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling wrote: Assuming SIP "sip show channels" will show you which codec is used for each call leg. However it does not track transcoding. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs asterisk -rx "core show channels verbose" does not provide transcoding details. Unless I have missed something. Sans On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas wrote: Core show channels verbose is probably your best bet. I think the answer also depends on your * version. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Wednesday, August 31, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cli command show codecs Hi, Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ? Thx Sans -- _ -- 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 -- _ -- 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] cli command show codecs
Thx @Danny I am feeling a bit lost here... We are using G711-aLaw for all our calls (endpoints) and I would like to align everything to this codec. I have an MOH file -- a custom wav file. How do I check its codec format ? And if its not G711-aLaw, how do I convert it to G711-aLaw. Thank you. Sans On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas wrote: > Maybe this will be better than my first answer – Audio files do indeed have > codec formats. If you are in a particular codec (say G729), > Playback/Background and MOH will search for files that match the codec > format first, then transcode WAV/GSM/whatever you have to that format if it > isn’t found. Ideally, you want to have a copy of each codec you can play > for all sounds and MOH. Each of the “canned sounds” comes in each codec > format (you pick the ones you want when you do make menuselect). > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai > Sent: Thursday, September 01, 2011 5:35 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] cli command show codecs > > > > Hi, > > Does audio files have codec formats? I simply convert all my audios (MOH, > accouncements) to .wav format, 16bit, 11kHz (I believe this is the best > format for asterisk). > I am new to this and may be incorrect. > > Going forward, > (a) How can I check the codec format of my announcements, MOH ? > (b) How can I record/convert announcements, MoH etc to a particular format ? > > I believe its a good idea to prevent transcoding and save CPU overheads. > > Thx > Sans > > > On Thu, Sep 1, 2011 at 11:39 AM, Bruce B wrote: > > if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If your > IVR announcement is not recorded in g729 and you see g729 on the channel > when you call into IVR then it's transcoding as well. > > > > On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling wrote: > > Assuming SIP "sip show channels" will show you which codec is used for each > call leg. However it does not track transcoding. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai > > Sent: Wednesday, August 31, 2011 2:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] cli command show codecs > > asterisk -rx "core show channels verbose" does not provide transcoding > details. > > Unless I have missed something. > > Sans > > > > On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas wrote: > > > Core show channels verbose is probably your best bet. I think the > answer also depends on your * version. > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai > Sent: Wednesday, August 31, 2011 10:44 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] cli command show codecs > > > > Hi, > > Is there a CLI command which will tell me the codec used for active > calls and if transcoding is happening ? > > Thx > Sans > > > -- > _ > -- 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 > > > > -- > _ > -- 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 > -- _ -- Ba
Re: [asterisk-users] cli command show codecs
Asterisk has a built-in file convert help file convert Usage: file convert Convert from file_in to file_out. If an absolute path is not given, the default Asterisk sounds directory will be used. Example: file convert tt-weasels.gsm tt-weasels.ulaw -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, September 01, 2011 8:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs Thx @Danny I am feeling a bit lost here... We are using G711-aLaw for all our calls (endpoints) and I would like to align everything to this codec. I have an MOH file -- a custom wav file. How do I check its codec format ? And if its not G711-aLaw, how do I convert it to G711-aLaw. Thank you. Sans On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas wrote: > Maybe this will be better than my first answer – Audio files do indeed > have codec formats. If you are in a particular codec (say G729), > Playback/Background and MOH will search for files that match the codec > format first, then transcode WAV/GSM/whatever you have to that format > if it isn’t found. Ideally, you want to have a copy of each codec you > can play for all sounds and MOH. Each of the “canned sounds” comes in > each codec format (you pick the ones you want when you do make menuselect). > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL > Mumbai > Sent: Thursday, September 01, 2011 5:35 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] cli command show codecs > > > > Hi, > > Does audio files have codec formats? I simply convert all my audios > (MOH, > accouncements) to .wav format, 16bit, 11kHz (I believe this is the > best format for asterisk). > I am new to this and may be incorrect. > > Going forward, > (a) How can I check the codec format of my announcements, MOH ? > (b) How can I record/convert announcements, MoH etc to a particular format ? > > I believe its a good idea to prevent transcoding and save CPU overheads. > > Thx > Sans > > > On Thu, Sep 1, 2011 at 11:39 AM, Bruce B wrote: > > if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If > your IVR announcement is not recorded in g729 and you see g729 on the > channel when you call into IVR then it's transcoding as well. > > > > On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling wrote: > > Assuming SIP "sip show channels" will show you which codec is used for > each call leg. However it does not track transcoding. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL > Mumbai > > Sent: Wednesday, August 31, 2011 2:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] cli command show codecs > > asterisk -rx "core show channels verbose" does not provide transcoding > details. > > Unless I have missed something. > > Sans > > > > On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas wrote: > > >Core show channels verbose is probably your best bet. I think > the answer also depends on your * version. > > > >From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL > Mumbai >Sent: Wednesday, August 31, 2011 10:44 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [asterisk-users] cli command show codecs > > > >Hi, > >Is there a CLI command which will tell me the codec used for > active calls and if transcoding is happening ? > >Thx >Sans > > >-- > > _ >-- 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 > > > > -- > _ > -- 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 ma
Re: [asterisk-users] Asterisk 1.8.3.3 T.38 Gateway
On 11-09-01 07:04 AM, Tim King wrote: I have found numerous claims that 1.8 can do T.38 gateway with a patch, however I am yet to find the patch or any instructions on implementing it. Anyone have a link? Asterisk-10.0.0-beta1 is another option. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://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] cli command show codecs
Thanks again @Danny. File converter worked like a charm. asterisk -rx "file convert /var/lib/asterisk/mohmp3/wav_Track11.wav wav_Track11.alaw" I coped the new file from sounds/ folder to my desktop And I tried to upload the new .alaw file using FreePBX, I got the following error: Error Processing: "sox failed to convert file and original could not be copied as a fall back" for wav_Track111.alaw! This is not a fatal error, your Music on Hold may still work. Pls help with this last bit. Thx Sans On Thu, Sep 1, 2011 at 7:17 PM, Danny Nicholas wrote: > Asterisk has a built-in file convert > > help file convert > Usage: file convert > Convert from file_in to file_out. If an absolute path is not given, the > default Asterisk sounds directory will be used. > > Example: > file convert tt-weasels.gsm tt-weasels.ulaw > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai > Sent: Thursday, September 01, 2011 8:26 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] cli command show codecs > > Thx @Danny > > I am feeling a bit lost here... > > We are using G711-aLaw for all our calls (endpoints) and I would like to > align everything to this codec. > > I have an MOH file -- a custom wav file. How do I check its codec format ? > > And if its not G711-aLaw, how do I convert it to G711-aLaw. > > Thank you. > Sans > > > > > > On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas wrote: >> Maybe this will be better than my first answer – Audio files do indeed >> have codec formats. If you are in a particular codec (say G729), >> Playback/Background and MOH will search for files that match the codec >> format first, then transcode WAV/GSM/whatever you have to that format >> if it isn’t found. Ideally, you want to have a copy of each codec you >> can play for all sounds and MOH. Each of the “canned sounds” comes in >> each codec format (you pick the ones you want when you do make menuselect). >> >> >> >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >> Mumbai >> Sent: Thursday, September 01, 2011 5:35 AM >> >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] cli command show codecs >> >> >> >> Hi, >> >> Does audio files have codec formats? I simply convert all my audios >> (MOH, >> accouncements) to .wav format, 16bit, 11kHz (I believe this is the >> best format for asterisk). >> I am new to this and may be incorrect. >> >> Going forward, >> (a) How can I check the codec format of my announcements, MOH ? >> (b) How can I record/convert announcements, MoH etc to a particular format ? >> >> I believe its a good idea to prevent transcoding and save CPU overheads. >> >> Thx >> Sans >> >> >> On Thu, Sep 1, 2011 at 11:39 AM, Bruce B wrote: >> >> if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If >> your IVR announcement is not recorded in g729 and you see g729 on the >> channel when you call into IVR then it's transcoding as well. >> >> >> >> On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling wrote: >> >> Assuming SIP "sip show channels" will show you which codec is used for >> each call leg. However it does not track transcoding. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >> Mumbai >> >> Sent: Wednesday, August 31, 2011 2:45 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> >> Subject: Re: [asterisk-users] cli command show codecs >> >> asterisk -rx "core show channels verbose" does not provide transcoding >> details. >> >> Unless I have missed something. >> >> Sans >> >> >> >> On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas wrote: >> >> >> Core show channels verbose is probably your best bet. I think >> the answer also depends on your * version. >> >> >> >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >> Mumbai >> Sent: Wednesday, August 31, 2011 10:44 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] cli command show codecs >> >> >> >> Hi, >> >> Is there a CLI command which will tell me the codec used for >> active calls and if transcoding is happening ? >> >> Thx >> Sans >> >> >> -- >> >> _ >> -- 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 1.8.3.3 T.38 Gateway
- Original Message - > On 11-09-01 07:04 AM, Tim King wrote: > > I have found numerous claims that 1.8 can do T.38 gateway with a > > patch, > > however I am yet to find the patch or any instructions on > > implementing it. > > Anyone have a link? > > > Asterisk-10.0.0-beta1 is another option. > I've been testing the T.38 functionality in 10.0.0-beta1 with very successful results. --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] cli command show codecs
Surprisingly, despite the error message, the files is uploaded in "/var/lib/asterisk/mohmp3" with correct permissions and ownership. Its not showing in FreePBX MOH Screen. I guess its a FreePBX issue. Sans On Thu, Sep 1, 2011 at 7:56 PM, RSCL Mumbai wrote: > Thanks again @Danny. > > File converter worked like a charm. > asterisk -rx "file convert /var/lib/asterisk/mohmp3/wav_Track11.wav > wav_Track11.alaw" > > I coped the new file from sounds/ folder to my desktop > And I tried to upload the new .alaw file using FreePBX, > > I got the following error: > > Error Processing: "sox failed to convert file and original could not > be copied as a fall back" for wav_Track111.alaw! > This is not a fatal error, your Music on Hold may still work. > > > Pls help with this last bit. > > Thx > Sans > > > > > On Thu, Sep 1, 2011 at 7:17 PM, Danny Nicholas wrote: >> Asterisk has a built-in file convert >> >> help file convert >> Usage: file convert >> Convert from file_in to file_out. If an absolute path is not given, the >> default Asterisk sounds directory will be used. >> >> Example: >> file convert tt-weasels.gsm tt-weasels.ulaw >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai >> Sent: Thursday, September 01, 2011 8:26 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] cli command show codecs >> >> Thx @Danny >> >> I am feeling a bit lost here... >> >> We are using G711-aLaw for all our calls (endpoints) and I would like to >> align everything to this codec. >> >> I have an MOH file -- a custom wav file. How do I check its codec format ? >> >> And if its not G711-aLaw, how do I convert it to G711-aLaw. >> >> Thank you. >> Sans >> >> >> >> >> >> On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas wrote: >>> Maybe this will be better than my first answer – Audio files do indeed >>> have codec formats. If you are in a particular codec (say G729), >>> Playback/Background and MOH will search for files that match the codec >>> format first, then transcode WAV/GSM/whatever you have to that format >>> if it isn’t found. Ideally, you want to have a copy of each codec you >>> can play for all sounds and MOH. Each of the “canned sounds” comes in >>> each codec format (you pick the ones you want when you do make menuselect). >>> >>> >>> >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >>> Mumbai >>> Sent: Thursday, September 01, 2011 5:35 AM >>> >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] cli command show codecs >>> >>> >>> >>> Hi, >>> >>> Does audio files have codec formats? I simply convert all my audios >>> (MOH, >>> accouncements) to .wav format, 16bit, 11kHz (I believe this is the >>> best format for asterisk). >>> I am new to this and may be incorrect. >>> >>> Going forward, >>> (a) How can I check the codec format of my announcements, MOH ? >>> (b) How can I record/convert announcements, MoH etc to a particular format ? >>> >>> I believe its a good idea to prevent transcoding and save CPU overheads. >>> >>> Thx >>> Sans >>> >>> >>> On Thu, Sep 1, 2011 at 11:39 AM, Bruce B wrote: >>> >>> if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If >>> your IVR announcement is not recorded in g729 and you see g729 on the >>> channel when you call into IVR then it's transcoding as well. >>> >>> >>> >>> On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling wrote: >>> >>> Assuming SIP "sip show channels" will show you which codec is used for >>> each call leg. However it does not track transcoding. >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >>> Mumbai >>> >>> Sent: Wednesday, August 31, 2011 2:45 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> >>> Subject: Re: [asterisk-users] cli command show codecs >>> >>> asterisk -rx "core show channels verbose" does not provide transcoding >>> details. >>> >>> Unless I have missed something. >>> >>> Sans >>> >>> >>> >>> On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas wrote: >>> >>> >>> Core show channels verbose is probably your best bet. I think >>> the answer also depends on your * version. >>> >>> >>> >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >>> Mumbai >>> Sent: Wednesday, August 31, 2011 10:44 AM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: [asterisk-users] cli command show codecs >>> >>> >>> >>> Hi, >>> >>> Is there a CLI command which will tell me the codec used for >>> active calls and if transcoding is happening ? >>> >>> Thx >>> Sans >>> >>> >>> -- >>> >>> __
Re: [asterisk-users] cli command show codecs
On Thu, 1 Sep 2011, RSCL Mumbai wrote: I simply convert all my audios (MOH, accouncements) to .wav format, 16bit, 11kHz (I believe this is the best format for asterisk). 8KHz? (a) How can I check the codec format of my announcements, MOH ? The 'file' command will show you the format for some codecs: -t2::sedwards:/var/lib/asterisk/sounds$ file demo-congrats.* demo-congrats.gsm: data demo-congrats.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz (b) How can I record/convert announcements, MoH etc to a particular format ? sox can transcode files. In your quest to eliminate transcoding, the 'module show like codec' CLI command will show which codecs you have loaded. The 'use' column will show how many times a codec is in use. If it is non-zero for a codec you are not expecting, start digging through the 'show channel's. -- 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
[asterisk-users] Anyone using Asterisk on VirtualBox ?
Hi, Anyone using Asterisk on Virtualbox. I am using and facing CPU peaking issue. Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of now), 64bit CentOS 5.4. Only SIP and softphones. Max 10 simultaneous calls. Unable to ascertain if the problem is with Asterisk, Virtualbox, Configuration, or the whole system should not be the way it is. Anyone will to share their settings and help me. Thx Sanjay -- _ -- 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] cli command show codecs
On Thu, 1 Sep 2011, RSCL Mumbai wrote: Error Processing: "sox failed to convert file and original could not be copied as a fall back" for wav_Track111.alaw! Maybe FreePBX does not know how to construct a sox command line for files that do not have headers. -- 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] cli command show codecs
If I recall correctly, wav has headers and alaw does not. If this is so, that would be where the problem lies. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, September 01, 2011 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cli command show codecs On Thu, 1 Sep 2011, RSCL Mumbai wrote: > Error Processing: "sox failed to convert file and original could not > be copied as a fall back" for wav_Track111.alaw! Maybe FreePBX does not know how to construct a sox command line for files that do not have headers. -- 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 -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Thu, 1 Sep 2011, RSCL Mumbai wrote: Hi, Anyone using Asterisk on Virtualbox. I am using and facing CPU peaking issue. Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of now), 64bit CentOS 5.4. Only SIP and softphones. Max 10 simultaneous calls. Unable to ascertain if the problem is with Asterisk, Virtualbox, Configuration, or the whole system should not be the way it is. Anyone will to share their settings and help me. Thx Sanjay I tried and failed with VirtualBox too. Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance). I don't think VirtualBox is up to real-time stuff. We use LXC now, and it is fantastic. j -- _ -- 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] Faxes suddenly failing
kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Lee. -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere wrote: > > On Thu, 1 Sep 2011, RSCL Mumbai wrote: > > Hi, >> >> Anyone using Asterisk on Virtualbox. >> >> I am using and facing CPU peaking issue. >> >> Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores >> and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of >> now), 64bit CentOS 5.4. >> Only SIP and softphones. >> Max 10 simultaneous calls. >> >> Unable to ascertain if the problem is with Asterisk, Virtualbox, >> Configuration, or the whole system should not be the way it is. >> >> Anyone will to share their settings and help me. >> >> Thx >> Sanjay >> >> > I tried and failed with VirtualBox too. Timing seemed impossible to > maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and > nothing else going on (single instance). I don't think VirtualBox is up to > real-time stuff. > > We use LXC now, and it is fantastic. > > j > > Thx Jeff. Kindly share some more details on the kind of hardware you are using, LXC parameters and the kind of load the system can handle. I am sure this will help me and more like myself. Thx Sanjay -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Thu, Sep 1, 2011 at 9:25 PM, RSCL Mumbai wrote: > On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere wrote: > >> >> On Thu, 1 Sep 2011, RSCL Mumbai wrote: >> >> Hi, >>> >>> Anyone using Asterisk on Virtualbox. >>> >>> I am using and facing CPU peaking issue. >>> >>> Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores >>> and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of >>> now), 64bit CentOS 5.4. >>> Only SIP and softphones. >>> Max 10 simultaneous calls. >>> >>> Unable to ascertain if the problem is with Asterisk, Virtualbox, >>> Configuration, or the whole system should not be the way it is. >>> >>> Anyone will to share their settings and help me. >>> >>> Thx >>> Sanjay >>> >>> >> I tried and failed with VirtualBox too. Timing seemed impossible to >> maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and >> nothing else going on (single instance). I don't think VirtualBox is up to >> real-time stuff. >> >> We use LXC now, and it is fantastic. >> >> j >> >> > Thx Jeff. > > Kindly share some more details on the kind of hardware you are using, LXC > parameters and the kind of load the system can handle. > > I am sure this will help me and more like myself. > > Thx > Sanjay > My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay -- _ -- 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] Faxes suddenly failing
On 09/01/2011 11:50 PM, Lee Howard wrote: kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Because its the industry standard last resort of anyone who doesn't understand FAX and is using T.38. 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] problems with hylafax + iaxmodem + asterisk1.8.5
Hi!
from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5.
I have 2 computers in the lan, one is the Asterisk PBX and the other is the
server with hylafax and iaxmodem installed.
In Asterisk I set up an IAX trunk in this way:
___
iax.conf
[iaxmodem]
type=friend
context=outgoing-fax
disallow=all
allow=ulaw
username=iaxmodem
secret=password
qualify=yes
notransfer=yes
host=dynamic
requirecalltoken=no
callerid="Fax" <06456789>
t38pt_udptl=yes
___
In asterisk CLI when I write "IAX2 show peers" I read that the device is
reachable:
iaxmodem/iaxmod 10.0.1.202 (D) 255.255.255.255 4570 OK (3 ms)
In the end I put the configuration Hylafax and Iaxmodem.
I've created a context in Asterisk for incoming fax:
context IncomingFax {
_. => {
Dial(IAX2/iaxmodem);
};
h => {
&riaggancia();
}
};
the "call" comes, the modem answers but does not receive any faxes.
I give you also logs "/var/log/syslog" and "xferfaxlog"
Thanks for your patience.
**
* config ttyIAX in /etc/iaxmodem/ttyIAX
**
device /dev/ttyIAX
owner uucp:uucp
mode 660
port 4570
refresh 300
server 10.0.1.204 // this is asterisk 1.8.5
peername iaxmodem
secret password
cidname FAXServer
cidnumber 0123456789
codec slinear
**
* config.ttyIAX in /var/spool/hylafax/etc/config.ttyIAX
**
CountryCode: 39
AreaCode: 06
FAXNumber: +39.06.456789
LongDistancePrefix: 0
InternationalPrefix: 00
DialStringRules: etc/dialrules
ServerTracing: 0xFFF
SessionTracing: 0xFFF
RecvFileMode: 0600
LogFileMode: 0600
DeviceMode: 0600
RingsBeforeAnswer: 1
SpeakerVolume: off
GettyArgs: "-h %l dx_%s"
LocalIdentifier: "IAXmodem"
TagLineFont: etc/lutRS18.pcf
TagLineFormat: "Ricevuto da %%l|%c|Pagina %%P di %%T"
MaxRecvPages: 200
ModemType: Class1 # use this to supply a hint
Class1AdaptRecvCmd: AT+FAR=1
Class1TMConnectDelay: 400 # counteract quick CONNECT response
ModemResetCmds: AT+VCID=1 # enables CallID display
PagerTTYParity: none
CallIDPattern: "NMBR="
CallIDPattern: "NAME="
CallIDPattern: "ANID="
CallIDPattern: "NDID="
***
xferfaxlog
**
09/01/11 17:13 CALL 00013 ttyIAX "" fax "+39.06.456789" "" 0 0 0:00:01
0:00:01 "Ring detected without successful handshake" "" "" "::s" "" ""
***
/var/log/syslog
**
STATE CHANGE: RUNNING -> LISTENING
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [9:DATE=0901]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [9:TIME=1650]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [5:NAME=]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [15:NMBR=0461829011]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [11:ANID=]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [13:USER=iaxmodem]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [11:PASS=]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [11:CDID=]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [6:NDID=s]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [4:RING]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 "06654321"
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 ""
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 ""
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 "s"
Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING -> ANSWERING
Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without
successful handshake
Sep 1 16:50:12 FAXServer FaxGetty[6225]: <-- [5:ATH0\r]
Sep 1 16:50:12 FAXServer FaxGetty[6225]: --> [2:OK]
Sep 1 16:50:12 FAXServer FaxGetty[6225]: MODEM set DTR OFF
Sep 1 16:50:12 FAXServer FaxGetty[6225]: MODEM set baud rate: 0 baud (flow
control unchanged)
Sep 1 16:50:13 FAXServer FaxGetty[6225]: MODEM set DTR OFF
Sep 1 16:50:13 FAXServer FaxGetty[6225]: MODEM set baud rate: 0 baud (flow
control unchanged)
Sep 1 16:50:13 FAXServer FaxGetty[6225]: DELAY 75 ms
Sep 1 16:50:13 FAXServer FaxGetty[6225]: MODEM set DTR ON
Sep 1 16:50:13 FAXServer FaxGetty[6225]: DELAY 2600 ms
Sep 1 16:50:17 FAXServer FaxGetty[6225]: MODEM set baud rate: 19200 baud,
input flow XON/XOFF, output flow XON/XOFF
Sep 1 16:50:17 FAXServer FaxGetty[6225]: DELAY 10 ms
Sep 1 16:50:17 FAXServer FaxGetty[6225]: MODEM flush i/o
Sep 1 16:50:17 FAXServer FaxGetty[6225]: <-- [4:ATZ\r]
Sep 1 16:50:17 FAXServer FaxGetty[6225]: --> [2:OK]
Sep 1 16:50:17 FAXServer FaxGetty[6225]: DELAY 3000 ms
Sep 1 16:50:17 FAXServer HylaFAX[6247]: checkHostIdentity("localhost")
Sep 1 16:50:17 FAXServer
Re: [asterisk-users] Faxes suddenly failing
Steve Underwood wrote: On 09/01/2011 11:50 PM, Lee Howard wrote: kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Because its the industry standard last resort of anyone who doesn't understand FAX and is using T.38. Even HP recommends for their own fax machines it numerous times: http://h2.www2.hp.com/bizsupport/TechSupport/Document.jsp?lang=en&cc=us&taskId=110&prodSeriesId=378056&prodTypeId=18972&objectID=c00062808 http://h2.www2.hp.com/bizsupport/TechSupport/Document.jsp?objectID=buu02549&lang=en&cc=us&contentType=SupportFAQ&prodSeriesId=3366988&prodTypeId=15179 Yes, always a last-ditch effort, and if it actually succeeds in getting a legible document through then it means that either 1) the ECM protocol on either the sender or the receiver is gravely flawed, or 2) something that requires ECM (like V.34-Fax/SuperG3) ended up being disabled along with ECM and that the problem really had to do with that something and not with ECM. I've never seen a fax document that couldn't make it through with ECM enabled be able to come through legibly with ECM disabled otherwise. Thanks, Lee. -- _ -- 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 1.8.3.3 T.38 Gateway
Hi, > I've been testing the T.38 functionality in 10.0.0-beta1 > with very successful results. what about 1.8? Will the T38 enhancement also be included in the 1.8 version? Thanks a lot, -- Chau y hasta luego, Thorolf -- _ -- 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 1.8.3.3 T.38 Gateway
On 09/01/2011 11:21 AM, Thorolf Godawa wrote: Hi, I've been testing the T.38 functionality in 10.0.0-beta1 with very successful results. what about 1.8? Will the T38 enhancement also be included in the 1.8 version? T.38 gateway will not be included in the 1.8 releases. It was a feature developed after the first 1.8 release was made. -- 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] problems with hylafax + iaxmodem + asterisk1.8.5
Alessio wrote: I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. . Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [4:RING] Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 "06654321" Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 "" Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 "" Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 "s" Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING -> ANSWERING Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without successful handshake Sep 1 16:50:12 FAXServer FaxGetty[6225]: <-- [5:ATH0\r] Sep 1 16:50:12 FAXServer FaxGetty[6225]: --> [2:OK] It happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. However, let me say now that your setup that you describe strings the IAX2 channels out over your LAN which is no guarantee that there won't be jitter to cause you other problems. Normally iaxmodem (and probably therefore HylaFAX) should run on the same system as Asterisk. Thanks, Lee. -- _ -- 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] Unexpected behavior change from Asterisk 1.6.2.14 to Asterisk 1.8.5.0
In our office, we were running an Asterisk 1.6.2.14 machine with DAHDI 2.3.0.1, FreePBX 2.8.1 and an analog DAHDI card with 8 FXO ports. This machine had several DAHDI trunks defined in the FreePBX interface, each one containing a single DAHDI channel. It
also had a few outgoing routes defined in FreePBX, each of those grouping several of these DAHDI trunks. This setup worked correctly until the hard drive started failing. After backing up most of the data, we changed the hard drive and installed Asterisk
1.8.5.0 and DAHDI 2.4.1.2 with the same FreePBX 2.8.1. We then noticed that outgoing calls using the analog card were failing if the first tried channel was busy, instead of trying the next channel in the outgoing route. We traced this problem to a
situation described in a FreePBX ticket: http://www.freepbx.org/v2/ticket/5008 . The logs in the old hard drive showed the following whenever a channel was busy:
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:19]
Dial("SIP/514-07bb", "DAHDI/4/3904170,300,tTwW") in new stack
[2011-08-30 08:55:39] WARNING[2597] app_dial.c: Unable to create channel of
type 'DAHDI' (cause 0 - Unknown)
[2011-08-30 08:55:39] VERBOSE[2597] app_dial.c: == Everyone is busy/congested
at this time (1:0/0/1)
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:20]
NoOp("SIP/514-07bb", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL
and HANGUPCAUSE = 0") in new stack
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:21]
Goto("SIP/514-07bb", "s-CHANUNAVAIL,1") in new stack
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto
(macro-dialout-trunk,s-CHANUNAVAIL,1)
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1]
Set("SIP/514-07bb", "RC=0") in new stack
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2]
Goto("SIP/514-07bb", "0,1") in new stack
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,0,1)
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [0@macro-dialout-trunk:1]
Goto("SIP/514-07bb", "continue,1") in new stack
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto
(macro-dialout-trunk,continue,1)
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [continue@macro-dialout-trunk:1]
GotoIf("SIP/514-07bb", "1?noreport") in new stack
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto
(macro-dialout-trunk,continue,3)
[2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [continue@macro-dialout-trunk:3]
NoOp("SIP/514-07bb", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 -
failing through to other trunks") in new stack
In the old setup (with Asterisk 1.6.2.14), the error type reported by app_dial
was 0-Unknown and the dialing status was CHANUNAVAIL.
[Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:19]
Dial("SIP/213-00e7", "DAHDI/5/2201177,300,tTwW") in new stack
[Aug 31 12:10:13] WARNING[17513] app_dial.c: Unable to create channel of type
'DAHDI' (cause 17 - User busy)
[Aug 31 12:10:13] VERBOSE[17513] app_dial.c: == Everyone is busy/congested at
this time (1:1/0/0)
[Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:20]
NoOp("SIP/213-00e7", "Dial failed for some reason with DIALSTATUS = BUSY and
HANGUPCAUSE = 17") in new stack
[Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:21]
Goto("SIP/213-00e7", "s-BUSY,1") in new stack
[Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Goto
(macro-dialout-trunk,s-BUSY,1)
[Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s-BUSY@macro-dialout-trunk:1]
NoOp("SIP/213-00e7", "Dial failed due to trunk reporting BUSY - giving up")
in new stack
[Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s-BUSY@macro-dialout-trunk:2]
PlayTones("SIP/213-00e7", "busy") in new stack
In the new setup (with Asterisk 1.8.5.0), the error type reported by app_dial
is 17-User busy and the dialing status is BUSY.
The FreePBX context is programmed so that it considers BUSY, along with NOANSWER, INVALIDNMBR and CHANGED, as nonrecoverable errors that abort the dialout attempt, which seems reasonable. The problem is that the new setup is returing BUSY instead of
CHANUNAVAIL when the particular channel that was tried is in use by a different call. We worked around the issue by applying the recommendation suggested in the ticket (create DAHDI groups in chan_dahdi.conf and use these as trunks). However, I believe the
previous behavior was correct and the new behavior to be in error. The workaround suggested by the ticket will not work in a scenario where a DAHDI group has all of its channels busy with calls, and the administrator intends additional calls to be routed
through non-DAHDI trunks (such as SIP/IAX trunks or custom trunks).
Re: [asterisk-users] Unexpected behavior change from Asterisk 1.6.2.14 to Asterisk 1.8.5.0
> In our office, we were running an Asterisk 1.6.2.14 machine with DAHDI
> 2.3.0.1, FreePBX 2.8.1 and an analog DAHDI card with 8 FXO ports. This
> machine had several DAHDI trunks defined in the FreePBX interface,
> each one containing a single DAHDI channel. It
> also had a few outgoing routes defined in FreePBX, each of those
> grouping several of these DAHDI trunks. This setup worked correctly
> until the hard drive started failing. After backing up most of the
> data, we changed the hard drive and installed Asterisk
> 1.8.5.0 and DAHDI 2.4.1.2 with the same FreePBX 2.8.1. We then noticed
> that outgoing calls using the analog card were failing if the first
> tried channel was busy, instead of trying the next channel in the
> outgoing route. We traced this problem to a
> situation described in a FreePBX ticket:
> http://www.freepbx.org/v2/ticket/5008 . The logs in the old hard drive
> showed the following whenever a channel was busy:
>
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [s@macro-dialout-trunk:19] Dial("SIP/514-07bb",
> "DAHDI/4/3904170,300,tTwW") in new stack
> [2011-08-30 08:55:39] WARNING[2597] app_dial.c: Unable to create
> channel of type 'DAHDI' (cause 0 - Unknown)
> [2011-08-30 08:55:39] VERBOSE[2597] app_dial.c: == Everyone is
> busy/congested at this time (1:0/0/1)
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [s@macro-dialout-trunk:20] NoOp("SIP/514-07bb", "Dial failed for
> some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0") in new
> stack
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [s@macro-dialout-trunk:21] Goto("SIP/514-07bb", "s-CHANUNAVAIL,1")
> in new stack
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto
> (macro-dialout-trunk,s-CHANUNAVAIL,1)
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/514-07bb", "RC=0")
> in new stack
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/514-07bb", "0,1")
> in new stack
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto
> (macro-dialout-trunk,0,1)
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [0@macro-dialout-trunk:1] Goto("SIP/514-07bb", "continue,1") in
> new stack
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto
> (macro-dialout-trunk,continue,1)
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [continue@macro-dialout-trunk:1] GotoIf("SIP/514-07bb",
> "1?noreport") in new stack
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto
> (macro-dialout-trunk,continue,3)
> [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing
> [continue@macro-dialout-trunk:3] NoOp("SIP/514-07bb", "TRUNK Dial
> failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other
> trunks") in new stack
>
> In the old setup (with Asterisk 1.6.2.14), the error type reported by
> app_dial was 0-Unknown and the dialing status was CHANUNAVAIL.
>
> [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing
> [s@macro-dialout-trunk:19] Dial("SIP/213-00e7",
> "DAHDI/5/2201177,300,tTwW") in new stack
> [Aug 31 12:10:13] WARNING[17513] app_dial.c: Unable to create channel
> of type 'DAHDI' (cause 17 - User busy)
> [Aug 31 12:10:13] VERBOSE[17513] app_dial.c: == Everyone is
> busy/congested at this time (1:1/0/0)
> [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing
> [s@macro-dialout-trunk:20] NoOp("SIP/213-00e7", "Dial failed for
> some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17") in new stack
> [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing
> [s@macro-dialout-trunk:21] Goto("SIP/213-00e7", "s-BUSY,1") in new
> stack
> [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Goto
> (macro-dialout-trunk,s-BUSY,1)
> [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing
> [s-BUSY@macro-dialout-trunk:1] NoOp("SIP/213-00e7", "Dial failed
> due to trunk reporting BUSY - giving up") in new stack
> [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing
> [s-BUSY@macro-dialout-trunk:2] PlayTones("SIP/213-00e7", "busy")
> in new stack
>
> In the new setup (with Asterisk 1.8.5.0), the error type reported by
> app_dial is 17-User busy and the dialing status is BUSY.
>
> The FreePBX context is programmed so that it considers BUSY, along
> with NOANSWER, INVALIDNMBR and CHANGED, as nonrecoverable errors that
> abort the dialout attempt, which seems reasonable. The problem is that
> the new setup is returing BUSY instead of
> CHANUNAVAIL when the particular channel that was tried is in use by a
> different call. We worked around the issue by applying the
> recommendation suggested in the ticket (create DAHDI groups in
> chan_dahdi.conf and use these as trunks). However, I believe the
> previous behavior was correct and the new behavior to be in error. The
> workaround suggested by the ticket will not work in a scenario where a
> DAHDI group has all of its channels busy with calls, and the
> administrator intends additional calls t
Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?
On Thu, Sep 01, 2011 at 10:32:52AM -0400, Jeff LaCoursiere wrote: > I tried and failed with VirtualBox too. Timing seemed impossible to > maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and > nothing else going on (single instance). I don't think VirtualBox is up > to real-time stuff. What timing module do you use? I recall on several cases that the pthreads timing module worked better than the timerfd one. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On 11-09-01 03:30 PM, Tzafrir Cohen wrote: On Thu, Sep 01, 2011 at 10:32:52AM -0400, Jeff LaCoursiere wrote: I tried and failed with VirtualBox too. Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance). I don't think VirtualBox is up to real-time stuff. What timing module do you use? I recall on several cases that the pthreads timing module worked better than the timerfd one. 1.8.7.0-rc1 should have a few fixes for timerfd. It would be good to get some feedback from testers. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://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] Asterisk 1.8.3.3 T.38 Gateway
On 11-09-01 10:35 AM, Tim Nelson wrote: - Original Message - On 11-09-01 07:04 AM, Tim King wrote: I have found numerous claims that 1.8 can do T.38 gateway with a patch, however I am yet to find the patch or any instructions on implementing it. Anyone have a link? Asterisk-10.0.0-beta1 is another option. I've been testing the T.38 functionality in 10.0.0-beta1 with very successful results. Any information about the results you can post is good. I know we are interested in seeing the results. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://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
[asterisk-users] Phantom rings after FXO/FXS setup
I have recently setup Trixbox 2.6.1 on a machine and configured it with an FXO and FXS module. I can make and receive calls just fine so there is no problem with the configuration of how the ports are set. The problem I am having is when I miss a call. The phone will ring 15 minutes later and continue to ring exactly 15 minutes after that and 15 after that...etc. I cannot find anything online that tells me how to get it to quit this. Any help is greatly appreciated. Thanks. -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] Phantom rings after FXO/FXS setup
Voicemail indication on the FXS port? I you have voicemail configured the ring is indicating that the extension has a message waiting. On Thu, 2011-09-01 at 14:51 -0500, Chris Ramirez wrote: > I have recently setup Trixbox 2.6.1 on a machine and configured it > with an FXO and FXS module. I can make and receive calls just fine so > there is no problem with the configuration of how the ports are set. > The problem I am having is when I miss a call. The phone will ring 15 > minutes later and continue to ring exactly 15 minutes after that and > 15 after that...etc. I cannot find anything online that tells me how > to get it to quit this. Any help is greatly appreciated. Thanks. > -- > Chris Ramirez > TELE-ONE COMMUNICATIONS, INC. > crami...@tele-onecom.com > 903-531-0777 > -- > _ > -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Phantom rings after FXO/FXS setup
I have already checked that. The voice mail was disabled when it first occurred. So I set the voice mail up and it still happens but with no new messages. On 9/1/2011 3:33 PM, Carlos Chavez wrote: Voicemail indication on the FXS port? I you have voicemail configured the ring is indicating that the extension has a message waiting. On Thu, 2011-09-01 at 14:51 -0500, Chris Ramirez wrote: I have recently setup Trixbox 2.6.1 on a machine and configured it with an FXO and FXS module. I can make and receive calls just fine so there is no problem with the configuration of how the ports are set. The problem I am having is when I miss a call. The phone will ring 15 minutes later and continue to ring exactly 15 minutes after that and 15 after that...etc. I cannot find anything online that tells me how to get it to quit this. Any help is greatly appreciated. Thanks. -- Chris Ramirez TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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 -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] USB or Ethernet based FXO device ?
On Wed, Aug 31, 2011 at 11:55:37AM -0400, Andrew Latham wrote: > On Wed, Aug 31, 2011 at 11:49 AM, Carlos Chavez > wrote: > > only thing to keep in mind is to always connect the units in a > > predetermined order to the USB ports so you do not mess up your > > configuration. > I am sure that Tzafrir can pipe in here. There is an method of > setting the ID of each astribank to keep them in order. Ask Xorcom > for more info. http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_xpp_order_explicitly_order_astribanks -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Distributed device state / presence info??
Hi all, Last couple of days i've arguing with my colleges about presence-info. >From the asterisk-bible and the wiki's i learned that it is possible to let asterisk do some of the presense-info by means of the jabber.conf file and a seperate xmpp-server. On the other hand, most soft-phones are capable of "doing something" with presence, allthough most of them use SIMPLE-protocol, instead of XMPP. So if when should one use the presence info from asterisk and when use the presence info from the softphones. It looks to me like doing the same job twice. What i assume (please correct me if i am wrong) is that when a client registers/deregisters, asterisk will update the presence info towards the XMPP-server. Correct? But otoh, what people would like to see is who is "on line". And not only on the asterisk-server that they are connected to, but also from other possible asterisk servers. And furthermore, each registered user might want to set their presencse-status to either free/busy/away/what-ever. So if the changing/reading is to be done on a softphone, what is the point of having asterisk doing someting with the device-status??? While writing, i've got a distinct feeling i'm understanding less by the minute ;( Anyway, what i'm building is a central server and a number of asterisk-boxes that act as proxy/six-iax-converter. All of the registered users should be able to see the presence of all the users on either proxy. Hans -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote: > > > My main interest of being on Virtual platform is portability / Backup. > In case of any h/w issues, or crashes, simply copy the VM on to > another box and you are up in minutes. > > > Sanjay > -- Doing that right now, although in my case i use XEN. Besides being hw independant, it is easier to play with a different version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able to switch back in minutes. hw -- _ -- 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
