[asterisk-users] GSM-Card for Asterisk / recommendation needed
Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a vgsm I from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it should be cheap ;-) Best regards -Thorsten- -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 SIP realtime and NAT
Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : Set MD5Secret: Not set Remote Secret: Not set Context : pack-local Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 501@local VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 5 Max forwards : 0 Dynamic : Yes Callerid : MaxCallBR: 384 kbps Expire : 3326 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: Yes Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : x.x.x.x:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: PACK501 SIP Options : (none) Codecs : 0x10c (ulaw|alaw|g729) Codec Order : (g729:20,alaw:20,ulaw:20) Auto-Framing : No 100 on REG : Yes Status : OK (17 ms) Useragent: snom870/8.4.20 Reg. Contact : sip:501@x.x.x.x:52753;line=hu9fedy3 Qualify Freq : 12 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No But in the DB I have clearly set nat to yes select name,nat from sip where name ='501'; +--+-+ | name | nat | +--+-+ | 501 | yes | +--+-+ In all previous versions of asterisk we have used with realtime we would see a line in the sip show peer looking like: Nat : Always Has the table definition changed in asterisk 1.8.3? Is there a bug stopping this value being picked up? Can someone even point me to the correct source files so I can attempt to try and work out the correct 1.8 sip table definition from there as I can't find one anywhere at all? Thanks in advance 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] Using voice modem as poor man's FXO in Asterisk 1.8
I totally agreed with Leif Madsen that viable options are available and time and effort spent on winmodem should be carefully considered. My system also works with an ATA as PSTN gateway and VOIP SIP provider for DID and inbound/outbound service. It will save time much more time and effort while keep up the productivity. CK On Wed, Mar 2, 2011 at 8:53 AM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: On 11-02-27 09:12 PM, Stuart Longland wrote: I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN phone line here. There are several very good answers in this thread, and I suggest reading them. However, if hardware costs are the issue, then my recommendation is always to look at a SIP connection from an ITSP as your connection to the PSTN. The costs are nearly trivial (at least in Canada here you can have a DID for inbound calls for something around $5 a month, with termination costs in the range of 1c/min -- in other commonwealth countries I presume the costs are similar?). My bill rarely rises above $20 a month, and I use my phone a lot. (Business, personal, and 3 DID numbers are included in that cost.) I highly suggest you spend your time and money elsewhere, rather than chasing the dragon that seems to be winmodem FXO connectivity. If you absolutely must have hardware, then I suggest you start with used ATA (analog telephony adapters) that can be found on eBay, kijiji, craigslist, or any other assorted websites. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 SIP realtime and NAT
Hi Scratch that The value name has changed from Nat to Force Rport Back to the drawing board On Wed, 2011-03-02 at 08:57 +, Ishfaq Malik wrote: Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : Set MD5Secret: Not set Remote Secret: Not set Context : pack-local Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : 501@local VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 5 Max forwards : 0 Dynamic : Yes Callerid : MaxCallBR: 384 kbps Expire : 3326 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: Yes Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : x.x.x.x:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: PACK501 SIP Options : (none) Codecs : 0x10c (ulaw|alaw|g729) Codec Order : (g729:20,alaw:20,ulaw:20) Auto-Framing : No 100 on REG : Yes Status : OK (17 ms) Useragent: snom870/8.4.20 Reg. Contact : sip:501@x.x.x.x:52753;line=hu9fedy3 Qualify Freq : 12 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No But in the DB I have clearly set nat to yes select name,nat from sip where name ='501'; +--+-+ | name | nat | +--+-+ | 501 | yes | +--+-+ In all previous versions of asterisk we have used with realtime we would see a line in the sip show peer looking like: Nat : Always Has the table definition changed in asterisk 1.8.3? Is there a bug stopping this value being picked up? Can someone even point me to the correct source files so I can attempt to try and work out the correct 1.8 sip table definition from there as I can't find one anywhere at all? Thanks in advance 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
[asterisk-users] [1.4] Call progress for Zaptel 1.4.3.1?
Hi
With an FXO module + Zaptel, I'd like to know if there are ways to
know when the remote party has answered the phone, whether calling
through a callfile or by sending DTMF's.
I read about {CHANNEL(state), ChanIsAvail(), and ${DIALSTATUS}: Are
those reliable ways to know when the channel is available for dialing
out and the call has been answered?
www.voip-info.org/wiki/view/Asterisk+func+channel
www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail
www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
==
[callback]
exten = start,1,NoOp(DialStatus is ${DIALSTATUS})
;how to pause until party has answered?
;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing
Offhook, Pre-ring, Unknown
exten = start,n,While([${CHANNEL(state) != OffHook])
exten = start,n,NoOp(Channel still ringing)
exten = start,n,EndWhile()
exten = start,n,Answer(500)
exten = start,n,Playback(please-wait)
exten = start,n,Flash()
exten = start,n,SendDTMF(${GSM},500)
;how to pause until party has answered?
exten = start,n,Wait(5)
==
Thank you.
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[asterisk-users] Registering Cisco 7942G IP phone with Asterisk!.
Hi, We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device) IP address (with current firmware on it) to register it with Asterisk. Do we need to upgrade the SIP firmware to any latest versions? If yes, to which version we should be updating it? It would be of great help if you advice us on what are files we need to have in the tftpboot directory apart from the firmware (like SIP Default.conf, SIPMAC.cnf.xml, CTLSEPMAC.tlv, etc) for the upgradation. Regards, Srinivas -- _ -- 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] wav files are not playing asterisk
Your error is in front of you. format_wav.c:148 check_header: Not in mono 2 [Feb 28 22:27:07] WARNING[2736]: file.c:386 fn_wrapper: Unable to open format wav Your wav file is not in proper format. Must be mono, and at 8khz, 16bit You can resample by using this command: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql more info here: (found by using a cool search engine called google and typing asterisk wav format http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asteri sk From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Wednesday, March 02, 2011 12:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] wav files are not playing asterisk Hi I am using Asterisks as client. By console dial I can make calls. When do dial s from console it wil play demo files that I can here from headphone connected to asterisk running system(Android OS).If I play gsm file noise is coming,but asterisk is not playing wav files,below is the error I am getting. Where I can see the channels is encodes as GSM,and how to change to wav.? *CLI dial s -- Executing [s@default:1] Wait(ALSA/hw:0,0, 1) in new stack The 'dial' command is deprecated and will be removed in a future release. Please use 'console dial' instead. *CLI -- Executing [s@default:2] Answer(ALSA/hw:0,0, ) in new stack Console call has been answered -- Executing [s@default:3] Set(ALSA/hw:0,0, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing [s@default:4] Set(ALSA/hw:0,0, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [s@default:5] BackGround(ALSA/hw:0,0, demo-congrats) in new stack [Feb 28 22:27:06] WARNING[2736]: file.c:665 ast_openstream_full: File demo-congrats does not exist in any format [Feb 28 22:27:07] WARNING[2736]: file.c:995 ast_streamfile: Unable to open demo-congrats (format 0x40 (slin)): No such file or directory [Feb 28 22:27:07] WARNING[2736]: pbx.c:5830 pbx_builtin_background: ast_streamfile failed on ALSA/hw:0,0 for demo-congrats -- Executing [s@default:6] BackGround(ALSA/hw:0,0, demo-instruct) in new stack [Feb 28 22:27:07] WARNING[2736]: format_wav.c:148 check_header: Not in mono 2 [Feb 28 22:27:07] WARNING[2736]: file.c:386 fn_wrapper: Unable to open format wav [Feb 28 22:27:07] WARNING[2736]: file.c:995 ast_streamfile: Unable to open demo-instruct (format 0x40 (slin)): No such file or directory [Feb 28 22:27:07] WARNING[2736]: pbx.c:5830 pbx_builtin_background: ast_streamfile failed on ALSA/hw:0,0 for demo-instruct -- Executing [s@default:7] WaitExten(ALSA/hw:0,0, ) in new stack -- Timeout on ALSA/hw:0,0, going to 't' -- Executing [t@default:1] Goto(ALSA/hw:0,0, #|1) in new stack -- Goto (default,#,1) -- Executing [#@default:1] Playback(ALSA/hw:0,0, demo-thanks) in new stack -- ALSA/hw:0,0 Playing 'demo-thanks' (language 'en') -- Executing [#@default:2] Hangup(ALSA/hw:0,0, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'ALSA/hw:0,0' Thanks Nikhil On 03/01/2011 07:55 PM, Steve Edwards wrote: On Tue, 1 Mar 2011, Nikhil wrote: I try to play a wav file in asterisk ,but its accepting only gsm files.Do u know where I need to change to make it works. Asterisk chooses a file encoding based on the channel encoding. If your channel is encoded as GSM, Asterisk will not look for a .wav of the same name if a .gsm is available. If the .gsm is not available, Asterisk will use the .wav with the additional 'overhead' of transcoding the data to GSM. Without any console log, these are just guesses: 1) Don't specify the file type in your dialplan. Asterisk chooses a file type for you based on channel encoding and formatting modules loaded. Is format_wav.so loaded? 2) Your WAV file is not encoded correctly. The 'file' command should show something like 'RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz' 3) You have permission issues. From a shell, as the user your instance of Asterisk runs as, can you access the file? If this doesn't help, please repost including the relevant dialplan context (as displayed by 'dialplan show relevant-context-name) and a snippet of the console log of a call playing the WAV file. -- _ -- 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] [1.4] Call progress for Zaptel 1.4.3.1?
On Wed, Mar 02, 2011 at 10:54:14AM +0100, Gilles wrote: Hi With an FXO module + Zaptel, I'd like to know if there are ways to know when the remote party has answered the phone, Any chance thi information is available through polarity reversal? In thise case: answeronpolarityswitch = yes hanguponpolarityswitch = yes -- 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] GSM-Card for Asterisk / recommendation needed
Try using openvox gsm cards. http://www.openvox.cn/store/g400p-p-63.html?cPath=25zenid=1fb3262c83d14d02b40fb6f577c7ebb7 its cheaper as well... On Wed, Mar 2, 2011 at 2:13 PM, Thorsten Göllner t...@ovm-group.com wrote: Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a vgsm I from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it should be cheap ;-) Best regards -Thorsten- -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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 -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.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] Asterisk 1.6 and windows RTC
Hello folks, for a customer of us we are trying to make asterisk and windows RTC library work together, but without success. We *need* to use gsm codec, so in the peer section we have disallow=all allow=gsm the sip signaling is ok, and when sniffing we got this session description: INVITE from windows RTC: v=0. o=- 0 0 IN IP4 172.31.9.130. s=session. c=IN IP4 172.31.9.130. b=CT:1000. t=0 0. m=audio 4632 RTP/AVP 97 111 112 6 0 8 4 5 3 101. k=base64:ftJemQZ2pTDV5gzzqxG6ps5Ol5qiOt2qbP9L9Or5JQg. a=rtpmap:97 red/8000. a=rtpmap:111 SIREN/16000. a=fmtp:111 bitrate=16000. a=rtpmap:112 G7221/16000. a=fmtp:112 bitrate=24000. a=rtpmap:6 DVI4/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=encryption:optional. a=direction:active. OK from asterisk 1.6 PBX: v=0. o=PBX 1705093286 1705093286 IN IP4 172.31.9.251. s=PBX. c=IN IP4 172.31.9.251. t=0 0. m=audio 14962 RTP/AVP 3 101. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. so, the rtp session should be GSM. But the audio does not work. In asterisk logs I see 'Got Siren7 offer at 24000 bps but only 32000 bps supported'. any hint? anyone with the same issue? unfortunately GSM is mandatory for us (we could not use alaw/ulaw, that seems working). thanks so much stefano -- _ -- 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] Hardware recommendation needed
Hi, We are planning to set up a prototype IVR system in Urdu language using Asterisk. For speech recognition, we will be using our own engine built using Sphinx, and for text to speech synthesis (for run time generation of responses based on user queries), we have a system for Urdu built in C++ that can be used as an API. My question is, can the Linksys SPA400 telephony gateway be used with Asterisk to develop the IVR system described? And if not, what other options should we explore? We have looked into the following options: 1. The Linksys SPA400 telephony gateway: we have used this previously with Trixbox to collect speech data over a telephone line, but we are not sure if it would support an IVR system such as the one described. 2. Digium telephony cards: we may have to rule these out because of cost issues if we have other options available. Also, most of these seem to be internal cards, and we would prefer to use an external device due to some equipment related limitations. 3. Dialogic cards: these were also ruled out due to cost issues. 4. We have also looked at Asterisk documentation and it seems that an IVR system setup should be possible with any of these devices, but could find no recommendations for IVR applications in particular. Any suggestions will be much appreciated. -- Thanks regards, Huda Sarfraz www.cle.org.pk -- _ -- 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] Failover Routing
It seems like it is a v1.8 only function at present (unless a backport is
released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}
Asterisk 1.8 also comes with a 'use_q850_reason' configuration option for
generating and parsing, if available:
-
That will give you what you want if you consider upgrading to v1.8.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 01 March 2011 16:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
Try this - it says it is for 1.8 but might work in 1.6
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepika Nijhawan
Sent: Tuesday, March 01, 2011 10:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
SIP_HEADER() gives you only access to headers of the initial INVITE request
(and not, for example, the final BYE message) How will I check sip response
with this like 404 or 503?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 01 March 2011 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
-Original Message-
From: Bob Beers [mailto:bob.be...@gmail.com]
Sent: 01 March 2011 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Deepika Nijhawan
Subject: Re: [asterisk-users] Failover Routing
On Tue, Mar 1, 2011 at 8:09 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com
wrote:
Ya, below is my routing:
Exten = 1234,1,Dial(SIP/abc)
Exten = 1234,n,Dial(SIP/xyz)
If 1234 is unallocated on SIP/abc it returns 1 in ${HANGUPCAUSE}
variable. For this I don't want it to try SIP/xyz. But overall, if we
get SIP 4xx reason then call should hangup like it
sends
back 404 not found for this case and if we get SIP 5xx response then
should
try SIP/xyz.
Is there any way to check sip responses and do failover routing based
on that?
Have you looked at SIP_HEADER() dialplan function?
https://wiki.asterisk.org/wiki/display/AST/Function_SIP_HEADER
Maybe you can parse Reason header in 4xx or 5xx response?
HTH,
-Bob
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepika Nijhawan
Sent: Tuesday, March 01, 2011 9:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
It says it for asterisk1.8. I am using asterisk1.6, can we use this function in
this version. Is it possible for you to give example on how to use?
I just went into my 1.4.37 console and find that SIP_HEADER is there in Core
show functions so it should be useable in 1.6.
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Re: [asterisk-users] [OT] Yealink IP Phones
It's all I use now. I was luckily enough to be involved with quite a bit of the beta testing in the UK - and, although there are a couple of 'nice-to-haves' missing, they are excellent handsets. Polycom sound quality at Grandstream prices ;) I particularly like the 'use your own screen logo' option. A gimmick maybe - but a nice one! Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: 25 February 2011 17:04 To: asterisk-users@lists.digium.com Subject: [asterisk-users] [OT] Yealink IP Phones Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear from you. -- Thanks, Phil If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] [1.4] Comparing value of string with spaces?
Hello
I haven't found an example on how to compare the value of a string
variable with spaces in it, and the While loop below never exits:
== extensions.conf
exten = start,n,Set(MYVAR=Dummy value)
exten = start,n,NoOp(${MYVAR})
;BAD TOO
;exten = start,n,While(!$[${MYVAR} : Some string])
exten = start,n,While($[${MYVAR} != Some string])
exten = start,n,Set(MYVAR=Some string)
exten = start,n,EndWhile()
==
Thank you.
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Re: [asterisk-users] [1.4] Comparing value of string with spaces?
Changing
exten = start,n,While($[${MYVAR} != Some string])
to
exten = start,n,While($[${MYVAR} != Some string])
does the trick for me.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: 02 March 2011 13:25
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [1.4] Comparing value of string with spaces?
Hello
I haven't found an example on how to compare the value of a
string variable with spaces in it, and the While loop below never exits:
== extensions.conf
exten = start,n,Set(MYVAR=Dummy value)
exten = start,n,NoOp(${MYVAR})
;BAD TOO
;exten = start,n,While(!$[${MYVAR} : Some string])
exten = start,n,While($[${MYVAR} != Some string])
exten = start,n,Set(MYVAR=Some string)
exten = start,n,EndWhile()
==
Thank you.
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you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8
On Wed, Mar 02, 2011 at 10:05:35AM +1000, Stuart Longland wrote: There's also regulatory requirements: here in Australia since I'm plugging into the PSTN, it needs to carry the ACMA's regulatory compliance mark. So buying something from overseas isn't an option. It's less of an option for me as I do not possess a credit card, and so many companies out there seem to think we're born with them. Thus ideally, I'm looking for where I can source one in my local area. Take a look at something like a Linksys SPA3102. Some people told me they managed to set this up as a SIP/PSTN gateway. -- Daniel Tryba -- _ -- 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] [1.4] Comparing value of string with spaces?
On Wed, 2 Mar 2011 13:37:57 -, Andrew Thomas
a...@datavox.co.uk wrote:
exten = start,n,While($[${MYVAR} != Some string])
Thanks Andrew.
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Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8
Hi, On Wed, Mar 02, 2011 at 05:20:07PM +0800, asterisk asterisk wrote: I totally agreed with Leif Madsen that viable options are available and time and effort spent on winmodem should be carefully considered. Indeed, but I never suggested anywhere using a winmodem. The modem I mentioned is a hardware modem connected via RS232, and there's the possibility that when our ADSL link goes down, I use it to bring up a backup 56kbps dial-up line. My system also works with an ATA as PSTN gateway and VOIP SIP provider for DID and inbound/outbound service. It will save time much more time and effort while keep up the productivity. Yep, all granted and I may consider a SIP service at some point, particularly for a business number. -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere. -- _ -- 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] [1.4] Call progress for Zaptel 1.4.3.1?
On Wed, 2 Mar 2011 14:06:12 +0200, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Any chance thi information is available through polarity reversal? In thise case: answeronpolarityswitch = yes hanguponpolarityswitch = yes Thanks for the tip. Google returned a bunch of discussions about issues such as not detecting that the remote party has hung up. Does someone know if it's OK to use all those for a TDM with an FXO module, or should only some be used together? == zapata.conf busydetect=yes busycount=4 answeronpolarityswitch=yes hanguponpolarityswitch=yes polarityonanswerdelay=1 polarityevents=yes callprogress=yes == BTW, am I correct in understanding that there are three ways for a telco to signal a hangup: - polarity reversal - disconnect clear (break in the electric current on the phone line when the other person hangs up) - BUSY signal ? I need to detect that the call is RINGING (which might not be answered at all), the remote end has answered and has hung up. Thank you. -- _ -- 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] Missing audio
I have a FreePBX system with PRI trunks that's doing a number of things very nicely, but frustrating me in one area. I am using a Grandstream GXW-4008 in an off-premises location to provide POTS service on four ports (this device worked fine in an early application using a hardware VPN to the Asterisk server). The Grandstream has a public static IP port, as does the Asterisk server. Extensions 1021, 1022, 1023, and 1024 register just fine. A ring group, 1020, distributes calls to these extensions and they handle incoming calls in hunt as I'd expect. Calls on the first port are consistently fine. Calls on the other ports are fine for a day or more, then they lose audio or have one-way audio. One mystery for me is that the first port always continues to work. I've assumed that this is some sort of UDP port problem, but I've Googled and studied stuff on-line and haven't figured out what I should be doing to fix it. I'd really appreciate some help. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -- _ -- 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] Doubt about cdr on asterisk
I have the following situation
I'm using Action Originate to originate a call for a costumer. Originate goes
to a context that call the dial application. Before the application (Dial using
the G option) to be invoked i'm setting the variable cellphone like this:
[firstcontext]
exten = s,1,Set(CDR(cellphone)=${CELLPHONE})
exten =
s,n,Dial(IAX2/user:pass@otherasterisk/${CELLPHONE},30,G(secondcontext^s^1))
[secondcontext]
exten = s,1,Hangup()
exten = s,n,Playback(something)
exten = s,n,NoOp(CDR(cellphone)
exten = s,n,Hangup()
When the costumer answer the call, caller party goes to secondcontex on
extension 1 and the called party goes to secondcontex on extension 2. On
firstcontext (before the Dial) i can see the value of variable cellphone, but
on my secondcontext (after Dial) the variable CDR(cellphone) is blank.
Is there something that i can do to pass the value after Dial application?
Luiz Gustavo Chiaretto
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Re: [asterisk-users] Doubt about cdr on asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luiz Gustavo
Chiaretto
Sent: Wednesday, March 02, 2011 8:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Doubt about cdr on asterisk
I have the following situation
I'm using Action Originate to originate a call for a costumer. Originate
goes to a context that call the dial application. Before the application
(Dial using the G option) to be invoked i'm setting the variable cellphone
like this:
[firstcontext]
exten = s,1,Set(CDR(cellphone)=${CELLPHONE})
exten =
s,n,Dial(IAX2/user:pass@otherasterisk/${CELLPHONE},30,G(secondcontext^s^1))
[secondcontext]
exten = s,1,Hangup()
exten = s,n,Playback(something)
exten = s,n,NoOp(CDR(cellphone)
exten = s,n,Hangup()
When the costumer answer the call, caller party goes to secondcontex on
extension 1 and the called party goes to secondcontex on extension 2. On
firstcontext (before the Dial) i can see the value of variable cellphone,
but on my secondcontext (after Dial) the variable CDR(cellphone) is blank.
Is there something that i can do to pass the value after Dial application?
You can't depend on CDR to hold this value because you create a new CDR
instance with the Dial application. You can set a local variable and reload
CDR(cellphone) after the Dial, like this
[firstcontext]
exten = s,1,Set(CDR(cellphone)=${CELLPHONE})
exten = s,n,Set(holdcellphone=${CELLPHONE})
exten =
s,n,Dial(IAX2/user:pass@otherasterisk/${CELLPHONE},30,G(secondcontext^s^1))
[secondcontext]
exten = s,1,Hangup()
exten = s,n,Playback(something)
Exten = s,n,Set(CDR(cellphone)=${holdcellphone})
exten = s,n,NoOp(CDR(cellphone)
exten = s,n,Hangup()
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[asterisk-users] asterisk behind nat
I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the Outside NIC Some of the phones are being disconnected with Asterisk saying no reply to critical packet How is Asterisk supposed to be configured? Currently this: externip = 94.18.x.x ; Address that we're going to put in outbound SIP messages ; if we're behind a NAT localnet = 192.168.5.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask ; The externip, localnet and localmask is used ; when registering and communication with other proxies ; that we're registered with tcpbindaddr=0.0.0.0 bindaddr = 0.0.0.0 Leif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Call progress for Zaptel 1.4.3.1?
On Wed, 02 Mar 2011 15:03:46 +0100, Gilles codecompl...@free.fr
wrote:
Does someone know if it's OK to use all those for a TDM with an FXO
module, or should only some be used together?
In addition, based on people's experience, is CHANNEL() reliable to
detect call progress?
;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing
Offhook, Pre-ring, Unknown
;diff between OffHook and Up?
exten = start,n,While($[${CHANNEL(state)} != Up ${INDEX} 10])
Thank you.
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Re: [asterisk-users] asterisk behind nat
On 3/2/2011 9:46 AM, Leif Neland wrote: Some of the phones are being disconnected with Asterisk saying no reply to critical packet What kind of phones are they? I might have nothing to do with your network configuration; try adding to sip.conf [general]: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Doubt about cdr on asterisk
Thanks for your answer Danny,
I thought there was another solution using some cdr options.
Best Regards.
Luiz Gustavo Chiaretto
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 2, 2011 11:42:58 AM
Subject: Re: [asterisk-users] Doubt about cdr on asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luiz Gustavo
Chiaretto
Sent: Wednesday, March 02, 2011 8:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Doubt about cdr on asterisk
I have the following situation
I'm using Action Originate to originate a call for a costumer. Originate goes
to a context that call the dial application. Before the application (Dial using
the G option) to be invoked i'm setting the variable cellphone like this:
[firstcontext]
exten = s,1,Set(CDR(cellphone)=${CELLPHONE})
exten =
s,n,Dial(IAX2/user:pass@otherasterisk/${CELLPHONE},30,G(secondcontext^s^1))
[secondcontext]
exten = s,1,Hangup()
exten = s,n,Playback(something)
exten = s,n,NoOp(CDR(cellphone)
exten = s,n,Hangup()
When the costumer answer the call, caller party goes to secondcontex on
extension 1 and the called party goes to secondcontex on extension 2. On
firstcontext (before the Dial) i can see the value of variable cellphone, but
on my secondcontext (after Dial) the variable CDR(cellphone) is blank.
Is there something that i can do to pass the value after Dial application?
You can’t depend on CDR to hold this value because you create a new CDR
instance with the Dial application. You can set a local variable and reload
CDR(cellphone) after the Dial, like this
[firstcontext]
exten = s,1,Set(CDR(cellphone)=${CELLPHONE})
exten = s,n,Set(holdcellphone=${CELLPHONE})
exten =
s,n,Dial(IAX2/user:pass@otherasterisk/${CELLPHONE},30,G(secondcontext^s^1))
[secondcontext]
exten = s,1,Hangup()
exten = s,n,Playback(something)
Exten = s,n,Set(CDR(cellphone)=${holdcellphone})
exten = s,n,NoOp(CDR(cellphone)
exten = s,n,Hangup()
--
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Re: [asterisk-users] records inbound and outbound calls
ok thanks for your response i have created an agent in sip sip.conf [222] type=friend context=agents host=dynamic dtmfmode=auto disallow=all allow=alaw allow=ulaw qualify=yes context=test i have add in extensions.conf the fil below but when i check in *var/spool/asterisk/monitor there is no record call* ** *could you please chek these configuration and tell me if there is any issue or wrong * *tahnks a lot * extensions.conf [test] exten = 100,1,Answer() exten = 100,2,MixMonitor(test.wav|av(0)V(0)) exten = 100,3,Dial(SIP/222) exten = 100,4,Hangup() 2011/3/1 Fellipe ... fellipe...@hotmail.com Hi, here is an example: http://www.asteriskguru.com/tutorials/mixmonitor.html Enjoy it! Best regards, Fellipe -- Date: Tue, 1 Mar 2011 17:06:32 + From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] records inbound and outbound calls thank you so much but i don't know how can i do could you please give an example to record an external call or which file I must to configure Thanks a lot 2011/3/1 Danny Nicholas da...@debsinc.com -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Tuesday, March 01, 2011 10:35 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] records inbound and outbound calls Hello List i have asterisk installed in our call centre i have configured the snom phone 320 and 370 with in sip.conf and dialplan.com and extenssion.com i have just one question how can i do in order to record all the calls automatically in our server Thanks and regards Just put a mixmonitor command after your Answer for incoming and add a macro to your dial command to start mixmonitor when dialing out. -- _ -- 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] records inbound and outbound calls
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, March 02, 2011 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] records inbound and outbound calls
snip
I copied the test context into my dialplan and ran it.
/var/spool/asterisk/monitor/test.wav was created with the audio of my call
(including the ringing).
How I did it
exten = 3009,1,Answer()
exten = 3009,2,MixMonitor(test.wav|av(0)V(0))
exten = 3009,3,Dial(SIP/144)
exten = 3009,4,Hangup()
How I would suggest you do it
exten = 100,1,Answer()
exten = 100,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 100,3,Dial(SIP/222)
exten = 100,4,Hangup()
This way, you get a new file for each call instead of overwriting
/v/s/a/m/test.wav each time.
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Re: [asterisk-users] records inbound and outbound calls
thank you i have one question waht is 3009 is the called
Regards
2011/3/2 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Wednesday, March 02, 2011 11:40 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] records inbound and outbound calls
snip
I copied the test context into my dialplan and ran it.
/var/spool/asterisk/monitor/test.wav was created with the audio of my call
(including the ringing).
How I did it
exten = 3009,1,Answer()
exten = 3009,2,MixMonitor(test.wav|av(0)V(0))
exten = 3009,3,Dial(SIP/144)
exten = 3009,4,Hangup()
How I would suggest you do it
exten = 100,1,Answer()
exten = 100,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 100,3,Dial(SIP/222)
exten = 100,4,Hangup()
This way, you get a new file for each call instead of overwriting
/v/s/a/m/test.wav each time.
--
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Re: [asterisk-users] records inbound and outbound calls
I made a sub-context 3009 in default to let me call from my phone sipphone
to my phone 144 and record the conversation.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, March 02, 2011 11:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] records inbound and outbound calls
thank you i have one question waht is 3009 is the called
Regards
2011/3/2 Danny Nicholas da...@debsinc.com
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, March 02, 2011 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] records inbound and outbound calls
snip
I copied the test context into my dialplan and ran it.
/var/spool/asterisk/monitor/test.wav was created with the audio of my call
(including the ringing).
How I did it
exten = 3009,1,Answer()
exten = 3009,2,MixMonitor(test.wav|av(0)V(0))
exten = 3009,3,Dial(SIP/144)
exten = 3009,4,Hangup()
How I would suggest you do it
exten = 100,1,Answer()
exten = 100,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 100,3,Dial(SIP/222)
exten = 100,4,Hangup()
This way, you get a new file for each call instead of overwriting
/v/s/a/m/test.wav each time.
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Re: [asterisk-users] records inbound and outbound calls
Un-top-posting...
2011/3/2 Danny Nicholas da...@debsinc.com
How I did it
exten = 3009,1,Answer()
exten = 3009,2,MixMonitor(test.wav|av(0)V(0))
exten = 3009,3,Dial(SIP/144)
exten = 3009,4,Hangup()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
salaheddine elharit
thank you i have one question waht is 3009 is the called
On Wed, 2 Mar 2011, Danny Nicholas wrote:
I made a sub-context 3009 in default to let me call from my phone
“sipphone” to my phone “144” and record the conversation.
3009 is an extension, not a [sub]context.
I'd add a suggestion to use the 'n' priority to make maintenance easier.
I use
'mixmonitor(/tmp/${EXTEN}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}.wav)'
on my dev box so the file name has the number I dialed as well as the
timestamp.
Also, recording calls without warning is illegal in [many|most|all]
countries.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] Failover Routing
On Wednesday 02 March 2011 07:06:31 Andrew Thomas wrote:
It seems like it is a v1.8 only function at present (unless a backport
is released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}
Asterisk 1.8 also comes with a 'use_q850_reason' configuration option
for generating and parsing, if available: -
That will give you what you want if you consider upgrading to v1.8.
A backport on this is not possible. It depends upon some core
functionality introduced in the 1.8 branch.
--
Tilghman
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Re: [asterisk-users] records inbound and outbound calls
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, March 02, 2011 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] records inbound and outbound calls
Un-top-posting...
2011/3/2 Danny Nicholas da...@debsinc.com
How I did it
exten = 3009,1,Answer()
exten = 3009,2,MixMonitor(test.wav|av(0)V(0))
exten = 3009,3,Dial(SIP/144)
exten = 3009,4,Hangup()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
salaheddine elharit
thank you i have one question waht is 3009 is the called
On Wed, 2 Mar 2011, Danny Nicholas wrote:
I made a sub-context 3009 in default to let me call from my phone
sipphone to my phone 144 and record the conversation.
3009 is an extension, not a [sub]context.
I'd add a suggestion to use the 'n' priority to make maintenance easier.
I use
'mixmonitor(/tmp/${EXTEN}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}.wav)'
on my dev box so the file name has the number I dialed as well as the
timestamp.
Also, recording calls without warning is illegal in [many|most|all]
countries.
--
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
I believe this satisfies all of the requirements...
exten = 3009,1,Answer()
exten =
3009,n,MixMonitor(3009-#{STRFTIME(${EPOCH},,%Y-%m-%d-$H-%M-%S)}.wav|av(0)V(0
))
exten = 3009,n,Dial(SIP/144,30,A(this-call-may-be-monitored-or-recorded))
exten = 3009,n,Hangup()
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[asterisk-users] Question on Asterisk 1.8 and Wait()
When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry -- _ -- 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] Question on Asterisk 1.8 and Wait()
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Wednesday, March 02, 2011 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on Asterisk 1.8 and Wait() When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+c ommands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry Best guess is that syntax changed from 1.4 to 1.8. Change line to Exten = s,1,Wait(1) -- _ -- 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] Question on Asterisk 1.8 and Wait()
On Wed, 2011-03-02 at 16:33 -0500, Jerry Geis wrote: When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry Could you post the dialplan usage? Maybe you have a typo. It should be something like: exten = s,1,Wait(1) https://wiki.asterisk.org/wiki/display/AST/Application_Wait -- 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] Question on Asterisk 1.8 and Wait()
Best guess is that syntax changed from 1.4 to 1.8. Change line to Exten = s,1,Wait(1) Danny Your correct. it was a syntax change. the above works. jerry -- _ -- 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] Functionality Questions
Hello, I am looking at implementing Asterisk for a project I'm working on. I need to authenticate a user against a database, and implement CC processing; shouldn't be a problem with PHP-AGI. In addition, I need to do several other things. After a user has been authenticated, they will be prompted to be sent to a queue. Before they're put in the queue, they need the option of recording a short message. When an agent picks up the phone, the message should be played to the agent (user still on hold) and the agent will then have the option of accepting or rejecting the call. Once in the call, a timer needs to be started. The timer will be a variable length (depending on the client). After that many minutes, the call needs to be disconnected. A minute prior to disconnection, I'd like a tone to be played on both ends signaling the impending termination of the call. Is this possible with Asterisk? Where do I start? Thank you -- Kyle Baczynski -- _ -- 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] Question on Asterisk 1.8 and Wait()
Jerry Geis wrote: Your correct. it was a syntax change. the above works. I've always used Wait(#) in my 1.4.x dial plans. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Functionality Questions
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Baczynski Sent: Wednesday, March 02, 2011 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Functionality Questions Hello, I am looking at implementing Asterisk for a project I'm working on. I need to authenticate a user against a database, and implement CC processing; shouldn't be a problem with PHP-AGI. In addition, I need to do several other things. After a user has been authenticated, they will be prompted to be sent to a queue. Before they're put in the queue, they need the option of recording a short message. When an agent picks up the phone, the message should be played to the agent (user still on hold) and the agent will then have the option of accepting or rejecting the call. Once in the call, a timer needs to be started. The timer will be a variable length (depending on the client). After that many minutes, the call needs to be disconnected. A minute prior to disconnection, I'd like a tone to be played on both ends signaling the impending termination of the call. Is this possible with Asterisk? Where do I start? Thank you -- Kyle Baczynski As I interpret this, this is all possible and has been covered in somewhat recent threads. -- _ -- 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] Functionality Questions
Hi Danny, Thank you. I will look through the archives, then. If anybody can provide any specific threads or key phrases I might search for (sometimes there are buzzwords for these things, you know), I would much appreciate it. I'm new to Asterisk. Thank you --- Kyle Baczynski On Wed, 2 Mar 2011 16:07:49 -0600, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Baczynski Sent: Wednesday, March 02, 2011 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Functionality Questions Hello, I am looking at implementing Asterisk for a project I'm working on. I need to authenticate a user against a database, and implement CC processing; shouldn't be a problem with PHP-AGI. In addition, I need to do several other things. After a user has been authenticated, they will be prompted to be sent to a queue. Before they're put in the queue, they need the option of recording a short message. When an agent picks up the phone, the message should be played to the agent (user still on hold) and the agent will then have the option of accepting or rejecting the call. Once in the call, a timer needs to be started. The timer will be a variable length (depending on the client). After that many minutes, the call needs to be disconnected. A minute prior to disconnection, I'd like a tone to be played on both ends signaling the impending termination of the call. Is this possible with Asterisk? Where do I start? Thank you -- Kyle Baczynski As I interpret this, this is all possible and has been covered in somewhat recent threads. -- _ -- 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] how to use qualify times to route calls
I'm using 1.8.3, and have 2 sip providers. Both are set with qualify=yes. Each of them sometimes have qualify times 10+ times the other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the lower qualify time? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
On 3/03/11 11:29 AM, sean darcy wrote: I'm using 1.8.3, and have 2 sip providers. Both are set with qualify=yes. Each of them sometimes have qualify times 10+ times the other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the lower qualify time? Traditionally you'd use a value you consider to be good enough for calls and set qualify to that. I.E. if you think 30ms is ok then set qualify=30 and then just route via the first then the second depending on status. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Wednesday, March 02, 2011 4:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to use qualify times to route calls
I'm using 1.8.3, and have 2 sip providers. Both are set with
qualify=yes. Each of them sometimes have qualify times 10+ times the
other. For instance, one will be at 10-15ms, the other at 200ms.
Is there a way I can route an outgoing call to the provider with the
lower qualify time?
sean
You could do a context using an AGI to do a sip show peers and select the
provider from that. Something like this
[pick_prov]
exten = s,1,AGI(getprov.agi)
exten = s,n,Dial(SIP/${EXTEN}@${BESTPROV},30,m)
getprov.agi does sip show peers and gets the qualify time from status.
The low value is returned in the variable BESTPROV.
Should be about 50 lines of PERL or PHP.
--
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Re: [asterisk-users] how to use qualify times to route calls
On 3/03/11 11:34 AM, Danny Nicholas wrote: getprov.agi does sip show peers and gets the qualify time from status. The low value is returned in the variable BESTPROV. If you're going to do that, you could probably knock something up with the SIPPEER function - SIPPEER(status). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
On 03/02/2011 05:34 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Wednesday, March 02, 2011 4:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to use qualify times to route calls
I'm using 1.8.3, and have 2 sip providers. Both are set with
qualify=yes. Each of them sometimes have qualify times 10+ times the
other. For instance, one will be at 10-15ms, the other at 200ms.
Is there a way I can route an outgoing call to the provider with the
lower qualify time?
sean
You could do a context using an AGI to do a sip show peers and select the
provider from that. Something like this
[pick_prov]
exten = s,1,AGI(getprov.agi)
exten = s,n,Dial(SIP/${EXTEN}@${BESTPROV},30,m)
getprov.agi does sip show peers and gets the qualify time from status.
The low value is returned in the variable BESTPROV.
Should be about 50 lines of PERL or PHP.
That would be a great idea, but would stretch my limits.
I'll try qualify=30 and qualifyfreq=20 to start.
sean
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Re: [asterisk-users] how to use qualify times to route calls
On Wed, 2 Mar 2011, sean darcy wrote: That would be a great idea, but would stretch my limits. Isn't that what makes it fun? -- 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] Failover Routing
What value do you get from the hangup cause, are they different?
I think can you use a gotoif checking the hangup cause.
On Wed, Mar 2, 2011 at 12:43 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote:
On Wednesday 02 March 2011 07:06:31 Andrew Thomas wrote:
It seems like it is a v1.8 only function at present (unless a backport
is released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}
Asterisk 1.8 also comes with a 'use_q850_reason' configuration option
for generating and parsing, if available: -
That will give you what you want if you consider upgrading to v1.8.
A backport on this is not possible. It depends upon some core
functionality introduced in the 1.8 branch.
--
Tilghman
--
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Robert
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Re: [asterisk-users] How do I find a phone numbers issued by Rogers?
Call them. On Wed, Mar 2, 2011 at 8:17 PM, Robert Augustyn robert.augus...@linqone.com wrote: Hi, Is there a way of finding out what block of phone numbers were issued to Roger’s business customers in my end of the woods? Thanks, Sincerely, Robert Augustyn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I find a phone numbers issued by Rogers?
Hm, I did not think about that I just assumed that they would not give it as this are the contact number of their clients ... I believe that I have seen it somewhere on the web cannot find it though. Sincerely, Robert Augustyn p:519.997.3106 ext:802 m:519.817.2503 e:robert.augus...@linqone.com -Original Message- From: shma...@gmail.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Wednesday, March 02, 2011 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do I find a phone numbers issued by Rogers? Call them. On Wed, Mar 2, 2011 at 8:17 PM, Robert Augustyn robert.augus...@linqone.com wrote: Hi, Is there a way of finding out what block of phone numbers were issued to Roger?s business customers in my end of the woods? Thanks, Sincerely, Robert Augustyn -- _ -- 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
[asterisk-users] Converting MP3 files to wav for Asterisk
Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk but it wasnt any better. #!/bin/bash for i in `ls $1/*mp3` do lame -a $i $i.wav mplayer -quiet -vo null -vc dummy -ao pcm:waveheader:file=$i.h.wav $i.wav sox $i.h.wav -t raw -r 8000 -s -2 -c 1 `echo $i|sed s/.mp3/.sln/` done - Any thoughts please? Regards, 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
[asterisk-users] chan_skinny and Cisco 793X (7936) support in 1.8
Is there any way to make a Cisco 7936 conference phone work in version 1.8? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I find a phone numbers issued by Rogers?
On Wednesday 02 March 2011 19:17:03 Robert Augustyn wrote: Is there a way of finding out what block of phone numbers were issued to Roger’s business customers in my end of the woods? You can find out from NANPA, the registry which assigns blocks of phone numbers. Note that due to phone number portability, however, this only will tell you the numbers that were originally allocated to Rogers, as customers are free to request existing numbers to be ported to them, and former customers are free to port their numbers away from Rogers. -- Tilghman -- _ -- 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] Testing from where number is...
Hi! My customer want's to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy... Is there a service that allows checking phone number? Maybe some specific Enum? I ask for number and server responds with info, for example: Cell Phone, Belgium or Land Line, Germany. -- Piotr Gorski -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I find a phone numbers issued by Rogers?
I don't remember exact name but there are two authorities which provide real-time portability information online but you need subscription. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Thursday, March 03, 2011 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do I find a phone numbers issued by Rogers? On Wednesday 02 March 2011 19:17:03 Robert Augustyn wrote: Is there a way of finding out what block of phone numbers were issued to Roger’s business customers in my end of the woods? You can find out from NANPA, the registry which assigns blocks of phone numbers. Note that due to phone number portability, however, this only will tell you the numbers that were originally allocated to Rogers, as customers are free to request existing numbers to be ported to them, and former customers are free to port their numbers away from Rogers. -- Tilghman -- _ -- 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] Testing from where number is...
www.numberingplans.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski Sent: Thursday, March 03, 2011 12:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Testing from where number is... Hi! My customer want's to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy... Is there a service that allows checking phone number? Maybe some specific Enum? I ask for number and server responds with info, for example: Cell Phone, Belgium or Land Line, Germany. -- Piotr Gorski -- _ -- 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] Converting MP3 files to wav for Asterisk
Try to convert into gsm instead wav. sox test.wav -r 8000 -c1 test.gsm Am 03.03.2011 06:20, schrieb Timothy Smith: Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk but it wasnt any better. #!/bin/bash for i in `ls $1/*mp3` do lame -a $i $i.wav mplayer -quiet -vo null -vc dummy -ao pcm:waveheader:file=$i.h.wav $i.wav sox $i.h.wav -t raw -r 8000 -s -2 -c 1 `echo $i|sed s/.mp3/.sln/` done - Any thoughts please? Regards, 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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
