Re: [Asterisk-Users] Calling number rewriting
You may change the CallerID with SetCallerID function, and the presentation with CallingPres, before dialing. In theory, you may place the CallerID you want, but your phone operator could refuse it if it doesn't belong to you (will show a default number). You also might have to use CallingPres to tune the presentation flags. For Swisscom with a ZapHFC BRI card, working values are 0: show, and 32: hide. Jean-Christophe ADEGOKE ARUNA a écrit : > Hi all, > > I will be glad if I can get response to my need. > > What I am trying to do is; > > I have a set up where my asterisk box is directly connected to a digitalk IN > platform. > > However, between the asterisk and digitalk is a104d sangoma card having e1 > with pri. The link between digitalk and my big alcatel switch is e1 with ss7 > signalling and this finally lead to my mobile operator > > What I am planning to do is to rewrite the caller id so that the calling > number presented to the mobile operator is going to be the number set on the > pri channels > > I will be glad, if anyone can just lead me on this > > goksie > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk acts as media gateway for existing pabx ?
Yes you can. I have done this for a while. First of all, you need an ISDN card that supports the NT mode. HFC ones work great with ZapHFC drivers (mode 0 = TE, mode 1 = NT). The cable between both systems MUST have 2 100 Ohm resistances. Depending on your PBX, you might need to power up the line or not (100V). A full explenation is available here: http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html Jean-Christophe Jean-Louis curty a écrit : >Hi, > >I'd like to realize the following setup traditional phones --> pbx >(non ip ) --> asterisk --- ISDN network > > >is it possible to set up an asterisk server in such way that it acts >as gateway between a traditionnal pbx and an isdn network ( I'm in >France ). this to let people continue with their existing pbx + phones >and enjoy voip features ? > >outgoing calls: >if yes,do I need isdn card dedicated to incoming calls from pbx to >asterisk and dedicated card from pbx to outside > >and what about the setup for incoming calls... the call comes in >asterisk server.. how to dispatch the call back to the pbx if it's an >extension still connected to the pbx ? > >I hope it's clear enough > >thanks >jean-louis >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LCR
In extensions.conf.
You should rather read a little bit about what file does what, and how
extensions and contexts work.
http://www.asteriskdocs.org/
http://www.voip-info.org/
http://www.digium.com/index.php?menu=documentation
If you ask questions on the mailing list that you could answer by
reading the docs, nobody will answer you.
Claudio Angeloni a écrit :
>Thank you for the quick reply, but as I'm VERY NEW to Asterisk, where do I
>make these changes?
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of
>Jean-Christophe Heger
>Sent: Sunday, 29 May 2005 18:21
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] LCR
>
>exten => _0.,1,Dial(Zap/g1/9${EXTEN:1})
>exten => _0.,102,Dial(Zap/g1/6129${EXTEN:1})
>
>:1 strip out the digit on the left.
>
>
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Re: [Asterisk-Users] LCR
exten => _0.,1,Dial(Zap/g1/9${EXTEN:1})
exten => _0.,102,Dial(Zap/g1/6129${EXTEN:1})
:1 strip out the digit on the left.
Claudio Angeloni a écrit :
> Ladies and Gents
>
> Please be patient as I try to explain what I am trying to achieve..
>
> I have a PSTN line and a Freshtel account, what I want to do is have
> the PSTN line as the first choice for outgoing calls for local calls
> and Freshtel as the second choice. The problem is that it's easy
> enough to set up both individually but how do I get the "second
> choice" drop the leading "0" and add "612" to the Freshtel trunk...
>
> In other words...
> USER - DIALS 0-9xxx
> ASTERISK 1st choice dials PSTN drops 0 dials 9xxx
> Freshtel 2nd choice drops 0 adds 612 9xxx
>
> I guess once I can do it for local calls, STD and IDD numbers would
> follow the same rules
>
> Please excuse me in advance if this is a lame question net I'm new to
> Asterisk and I am still trying to get my head around it..
>
> Thank You
> Claudio Angeloni
>
>
>
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Re: [Asterisk-Users] Dropping frame of G.729 since we already have a VAD frame at the end
I've got the same issue with a Swissvoice IP10S SIP phone. I couldn't find much information with this issue, but it seems to appear because Asterisk does not support variable length for g.729 (don't ask me what it really means). Anyway, it is recommanded to disable the silence suppression, which seems related to this issue. Unfortunately, it didn't work for me. But after setting the phone to canreinvite=no in the sip.conf, the connection worked allright. Don't ask me why. Jean-Christophe Bartosz Jozwiak a écrit : I have this showing on my cli while being in a call. Then connection gets broken. Can someone tell me what it means ? Dropping frame of G.729 since we already have a VAD frame at the end Thank you in advance. Bartosz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap' with zaphfc driver
Have you placed a group=1 in zapata.conf ? For a trial, you can use: exten => 203,1,Dial,Zap/1/onetelephonnumber Aitor a écrit : I new in asterisk world so, please, forgive me if I say something stupid. At least, and after a lot of tryes, the isdn card seems to be registered: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo channel == Starting D-Channel on span 1 And some info about channels: *CLI> zap show channels Chan Extension Context Language MusicOnHold pseudodefault 1default 2default *CLI> zap show channel 1 Channel: 1 File Descriptor: 21 Span: 1 Extension: Dialing: no Context: default Caller ID string: Destroy: 0 InAlarm: 1 Signalling Type: PRI Signalling Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF PRI Flags: PRI Logical Span: Implicit Actual Hookstate: Onhook I'm making this test in extensions.conf exten => 203,1,Dial,Zap/g1/onetelephonnumber But asterisk always says: Unable to create channel of type 'Zap' What I'm doing wrongly? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detection: Problem with extension number
For what I'm seeing in your log, the fax is detected, but you're missing
the fax extension. Here is how it works on my asterisk:
zapata.conf:
faxdetect = incoming
extensions.conf:
[pstn-in]
exten => 1234567,1,Goto(fax,s,1) ; This is a dedicated fax number
exten => fax,1,Goto(fax,s,1) ; For Zaptel fax detection (must
be answer first to be detected)
[fax]
exten => s,1,ZapEC(off)
exten => s,2,Wait(2)
exten => s,3,RxFAX(/var/lib/asterisk/fax/${UNIQUEID}.tif)
exten => s,4,Hangup
exten => fax,1,ZapEC(off)
exten => fax,2,RxFAX(/var/lib/asterisk/fax/${UNIQUEID}.tif)
exten => fax,3,Hangup
exten => h,1,System(/usr/bin/fax2pdf
/var/lib/asterisk/fax/${UNIQUEID}.tif /var/lib/asterisk/fax/${UNIQUEID}.pdf)
I'm not really sure what exten number to use in fax context, but it works.
Jean-Christophe
Jean-Yves Avenard a écrit :
>
>
> Hello
>
> I've been having the following problem today :
> I have a quite simple dialplan made to receive a fax:
>
> [answer-extension]
> exten => 1,1,Answer
> exten => 1,2,Macro(setcallerid)
> exten => 1,3,Ringing
> exten => 1,4,Wait(3)
> exten =>
> 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},${EXTENSION})
>
> exten => fax,1,Goto(faxreceive,1,1)
>
> The Wait(3) is there simply to let the system a bit of time to detect
> if it's a fax calling, this has worked so far in all cases except today.
> I received a fax from overseas and it seems that Asterisk has been
> unable to detect that it was a fax in the 3 seconds wait. Changing to
> 5s was sufficient to receive the fax. But obviously this is not a
> solution I want to adopt all the time as 1-it's a tool long wait,
> 2-what happens if 5s is still not enough another time.
>
> So Asterisk recognized that a fax was being received while executing
> the macro stdfwd3iax-notransfer (this extension simply check
>
> And I saw the following message in the console:
> -- Executing VoiceMail("Zap/10-1", "u200") in new stack
> -- Playing
> '/data/asterisk/var/spool/asterisk/voicemail/default/200/unavail'
> (language 'en')
> -- Redirecting Zap/10-1 to fax extension
> May 25 01:19:34 WARNING[17629]: pbx.c:2412 ast_pbx_run: Timeout, but
> no rule 't' in context 'answer-extension'
> -- Hungup 'Zap/10-1'
>
> It seems that Asterisk once entered in a Macro is unable to jump to
> the fax extension and gave me a timeout (which I do not handle in my
> dialplan). If I change the Wait(3) into Wait(0) the problem can be
> easily reproduced at all times.
>
> I then added a fax extension in the Macro just in case, but it made no
> difference whatsoever.
>
> Any ideas on what I'm doing wrong or is this a problem with Asterisk?
>
> The fully log is below.
> Thank you in advance
> Jean-Yves
>
> When I looked in the console on what what happening I say this:
> -- Accepting call from '' to '85735200' on channel 0/10, span 1
> -- Executing AGI("Zap/10-1", "getnumber.agi|200") in new stack
> -- Launched AGI Script
> /data/asterisk/var/lib/asterisk/agi-bin/getnumber.agi
> -- AGI Script getnumber.agi completed, returning 0
> -- Executing Set("Zap/10-1", "EXTENSION=00") in new stack
> -- Executing Goto("Zap/10-1", "answer-extension|1|1") in new stack
> -- Goto (answer-extension,1,1)
> -- Executing Answer("Zap/10-1", "") in new stack
> -- Executing Macro("Zap/10-1", "setcallerid") in new stack
> -- Executing GotoIf("Zap/10-1", "11?10:11") in new stack
> -- Goto (macro-setcallerid,s,10)
> -- Executing Set("Zap/10-1", "CALLERID(number)=''") in new stack
> -- Executing GotoIf("Zap/10-1", "0?20") in new stack
> -- Executing Ringing("Zap/10-1", "") in new stack
> -- Executing Wait("Zap/10-1", "0") in new stack
> -- Executing Macro("Zap/10-1",
> "stdfwd3iax-notransfer|200|200|100") in new stack
> -- Executing DBget("Zap/10-1", "temp=CFIM/200") in new stack
> -- DBget: varname=temp, family=CFIM, key=200
> -- DBget: Value not found in database.
> -- Executing Goto("Zap/10-1", "3") in new stack
> -- Goto (macro-stdfwd3iax-notransfer,s,3)
> -- Executing Dial("Zap/10-1", "IAX2/iax100|20|tr") in new stack
> May 25 01:19:21 NOTICE[17629]: app_dial.c:972 dial_exec_full: Unable
> to create channel of type 'IAX2' (cause 3)
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing NoOp("Zap/10-1", "CHANUNAVAIL") in new stack
> -- Executing Goto("Zap/10-1", "s-CHANUNAVAIL|1") in new stack
> -- Goto (macro-stdfwd3iax-notransfer,s-CHANUNAVAIL,1)
> -- Executing Goto("Zap/10-1", "s|400") in new stack
> -- Goto (macro-stdfwd3iax-notransfer,s,400)
> -- Executing Dial("Zap/10-1", "SIP/ipp100|20|tr") in new stack
> May 25 01:19:21 NOTICE[17629]: app_dial.c:972 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 3)
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing Goto("Zap/10-1", "s2-CHANUNAVAIL|1") in new stack
> -- Goto (macro-stdfwd3iax-notransfer,s2-CHANUNAVAIL,1)
> -- Executing Goto("Zap
Re: [Asterisk-Users] Boosting Shared Internet Bandwidth for Asterisk
Depending on you bandwidth, you might not need QoS. Priority could be enough. In you sip.conf (if you use SIP), place a tos value: [general] tos = 0x10 ; low delay or tos = 0x46 ; DiffServ Premium (EF: Expedited Forward) Remark: for un unknown reason, tos=lowdelay doesn't work anymore on my asterisk (v1.0.7), but was working in the past. I replaced it by 0x10 (hex value of lowdelay). Most of the routers support PFIFO (FIFO with priority), which means that low delay flagged packet will be sent in priority. I haven't tested the 0x46 value yet. Routers must be configured for DiffServ values, while ToS is by default. But the low delay TOS bit is also set within the 0x46 value. If a router treat the the DiffServ byte as TOS, it should be sent with priority as well (to be validated). If you want to check what priority is set inside your packets, you might use Ethereal. You might see either UDP or RTP packets, depending on the RTP ports that are used. In the branch "Internet Protocol", you will find the TOS/DiffServ decode, named "Type of service" or "Differential services Field". The TOS low delay bit is the 5th, and should be 1. If you have a low bandwidth connection (e.g. 600/100), you might have a new problem if you are using TOS as low delay. Voice will be good, but data will stall. QoS won't resolve it, because big packets take too much time to travell. The only way to share bandwidth for voice and data, on low bandwidth lines, is to fragment the data. An MTU of 700 is quite good, but you have to assume about 15% of bandwidth loss, because of twice more overheads on big packets. Allthough, a 1200/200 kbps line usually doesn't require such tricks. Remark about Grandstream: If you are using a GS device, you must know that QoS is buggy, and will have no effect at all. You must upgrade to the beta version of the firmware, which is OK. Therefore, GS recommands a QoS value of 48 (whithout "0x" on a GS device). This is a DiffServ value, which does not set the los delay TOS bit. Cisco recommands 46, which does. Jean-Christophe chawki hammoud a écrit : >--- Matt Riddell <[EMAIL PROTECTED]> wrote: > > > > >>Assuming your provider completely ignores QOS, it is >>still not a >>complete waste of time. >> >>If for example you have 5 people on the LAN, 4 >>uploading files to a >>remote server and 1 trying to make a phone call. >> >> > >My ISP has the internet connection set-up where 8 >people share the bandwidth. Would the script still >help boost my voip calls? > > > > > >__ >Yahoo! Mail Mobile >Take Yahoo! Mail with you! Check email on your mobile phone. >http://mobile.yahoo.com/learn/mail >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set CallerID in zapata.conf with QuadBri or other solution with parallel call signalling
I can't tell you how to resolve your issue, but I can tell you about
mine. I was fighting for setting my outgoing number (MSN /
bri_cpe_ptmp), and showing or hiding the number, with Swisscom operator.
Showing or hiding the number is resolved by the CallingPres command. For
me, values 0 and 32 worked.
Reference:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20CallingPres
Setting the outgoing number was another headache.While incoming
CallerIDs had 7 digits, the ougoing MSN must be set to 9 digits. The
formerly called 'zone prefix', now fully included inside the number, has
to be sent as well.
I hope this can help you.
Jean-Christophe
my conf:
[pstn-out]
exten => _.,1,Macro(setmsn,${CALLERIDNUM})
exten => _.,2,CallingPres(32) ; 0 to show, 32 to hide
exten => _.,3,SetCallerID(${MSN})
exten => _.,4,Dial(Zap/g1/${EXTEN})
exten => _.,5,Congestion
[macro-setmsn]
exten => s,1,GotoIf($[${ARG1} = 120]?200)
exten => s,2,GotoIf($[${ARG1} = 121]?200)
exten => s,3,GotoIf($[${ARG1} = 122]?200)
exten => s,4,GotoIf($[${ARG1} = 123]?200)
exten => s,6,GotoIf($[${ARG1} = 131]?202)
exten => s,7,SetGlobalVar(MSN=3)
exten => s,200,SetGlobalVar(MSN=1)
exten => s,202,SetGlobalVar(MSN=2)
Companity a écrit :
> Hi,
>
> we are using asterisk with Junghanns QuadBri and some sip phones. 2
> channels are configured in NT mode (ISDN PBX connected, internal ) and
> 2 channels are connected to the public ISDN network (bri-cpe). We use
> Bristuff 0.2.0 RC8C from Junghanns.
>
> When a call comes in from the public phone for a specific extension
> (Hotline Number), we initiate a parallelcall to some SIP phones and
> also to our PBX through the quadbri and we also do a signalling on one
> mobile phone(through the second channel which is connected to the
> oublic phone network).
>
> Problem:
>
> The sip phones and the internal phones on the PBX see the number of
> the calling party correctly (e.g. 040-987654321). Cause we can´t set a
> callerid to the public phone network (to show the calling party
> number), we want to show an extension of our numbers on our isdn-bri
> (asigned by Carrier, e.g. our numbers are 12345-0 to 12345-99). If we
> use our current configuration, everythings works good, execpt the isdn
> call to the mobile phone. As calling id it shows 12345 w/o an
> extension. We would like to set a specific extension, so that for the
> call to the mobile phone it is displayed 12345-88 (so we see that is a
> call forwarded from asterisk for a specific extension) and the rest
> (SIP an PBX) should display the ID of the calling party (e.g.
> 040-987654321). Does anybody has an idea ? I tried to set a callerid
> in the zapata.conf for the channels dialing the mobile phone with
> callerid= "" <1234588> and also same w/ setcallerid(""<1234588>).
> Result is that it´s shown 12345 to the mobile phone.
>
> thanks for help
>
> Andreas
>
>
>
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[Asterisk-Users] SMS & Grandstream ATA-286
I'ld like to test SMSes between Asterisk and an analog phone on a GS-ATA-286 (SIP). I have spent many hours trying any exemple on voip-info or mailing-list, but no message got sent from the analog phone. My goal, at now, is to send SMSes locally from and to Asterisk. Does anyone have a working configuration ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for improvements
That's funny, people having good bandwidth always have a better way to do it. You should feel lucky, because no one provides 768kbps upstreams in Switzerland, except if you want to pay 1'000 USD per month for a leased line. There is nothing complicated, just mathematics. Here is the formula: MTU: Maximum transmit unit = 1492 Bytes (ADSL) UP: Up stream t: time spent for a full framed packet (1492 Bytes) t = 8 * UP / MTU 128k upstream -> 91 ms 256k upstream -> 45 ms 512k upstream -> 23 ms 768k upstream -> 15 ms Using a codec, such as GSM or G.729, will take around 20 to 30 ms for encoding and decoding. While you wait for a full framed packet to go through the ADSL line, a voip packet will wait need 20 ms for encoding, 91 ms for waiting (at the most), 30 ms to go to the destination (at the best), and 10 ms to be decoded. Total: 20+91+30+20 = 161 ms. Even worse, the jitter balance will play around 100 ms, because of waiting on full framed packed. That's what I call "breaking the jitter", because not all equipment does support such jitters. Depending on the line and the distance (hops), you can easily add 50 ms, bringing the total around 200 ms. Therefore we consider a conversation as good, when the overall delay is under 150 ms. Bringing the MTU to 700 ms does bring back the overall delay to this target, and the jitter to 50 ms. Regarding the results, 768 kbps up stream is working even without QoS (< 100 ms). PFIFO (TOS=lowdelay) is good enough for perfect communications. So, any other magical solution ? Jean-Christophe Michael Graves a écrit : >Sometimes this all sounds so complicatedbut it needn't be. I >suppose it can vary with the size of your installation. > >I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic >shaping feature I establish inbound and outbound pipes which are >bandwidth restricted to just less than my mesured average DSL rate. I >then break my traffic into three priority ques in each direction; >highest priority, medium priority, low priority. > >I assign all IAX traffic in/out to the highest priority que, and map >all IAX ports to the * server inside the LAN. > >In fact, I just used the Magic Shaper Wizard in m0n0, then added IAX >specific entries to give it highest priority. The whole process took >about a half hour. Just as easy as the Linksys BEFSR-81 that I had >before, but more reliable and more controllable. > >Now to be fair, I only have about 6 phones. I use IAX2 to all my ITSPs >and SIP in-house only. My DSL is 3M down / 768k up. > >Michael > >On Fri, 06 May 2005 08:37:20 +0200, Jean-Christophe Heger wrote: > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS for improvements
I've spent may hours to play with HTB QoS settings on the firewall, but with absolutely no effect. In fact, this is normal, because the time required to let a data packet going through the ADSL line will break the voice jitter. The only right way to handle this issue is to modify the MTU on the router. Without setting a TOS for voip, data where going through and voice was unusable. With a lowdelay (0x10) TOS set for voip, voice was going through, but data was blocked. With a lowdelay TOS and an HTB QoS on the router, data where going through slowly and voice was scambled. After many tests, an MTU of 700 did work quite well. I did loose 15% of bandwidth for data (twice more overheads), but data and voice may be used together. Those tests have been done on a 256 kbps up stream. There is a quite good explenation about this issue on Cisco's web site, and about they're LFI technology (link fragmentation and interleaving): http://www.cisco.com/warp/public/788/voice-qos/voip-mlppp.html#link_frag Jean-Chrsitophe Kumara Jayaweera a écrit : >Hello! Everybody!!, >I want to run VoIP in the same LAN (15 windows clients) which we use for >surfing the Internet. 6-7 softphones in the same client's machines is 'the >target'. My DSL is 128kbps, (I can go to 256kbps if required). So, I am told >to install some QoS's in the LAN to improve the voice quality. Frankly, I >don't know what it (QoS= Quality of Service) is. I hope you may help me >giving "Links" to read and briefing me your ideas. >Thanks to everybody in the list. >So far my success and progress are your help. >Thanks again >Kumara > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold
What version of mpg123 do you use ? You must have a 0.59r one, else you
will ear a strange noise, and the nothing. But no warning message.
For test, you can try this version:
ftp://ftp.proxad.fr/pub/Distributions_Linux/Mandrakelinux/official/10.2/i586/media/main/mpg123-0.59r-23mdk.i586.rpm
This is a Mandrake one, that I'm sure it's working. You can extract the
binary out of the RPM with mc (Midnight Commander).
You can also test the MP3Player, i.e.:
exten => 999,1,MP3Player(http://my.stream:8000/stream1)
Jean-Christophe
Sahil Gupta a écrit :
> Hi,
> I've been trying to get music on hold going on one of our servers:
>
> Upon dialling extension 005, it plays:
> -- Executing WaitMusicOnHold("SIP/parssyd1-4dbe", "30") in new stack
> -- Started music on hold, class 'default', on SIP/parssyd1-4dbe
>
> However, no music in the background
>
> MPG123 is intalled..
>
> musiconhold.conf shows:
> default => mp3:/var/lib/asterisk/mohmp3
>
> The directory has?:
> [EMAIL PROTECTED]:~# ls -al /var/lib/asterisk/mohmp3
> total 6589
> drwxr-xr-x 2 root root 160 2005-04-21 10:25 ./
> drwxr-xr-x 8 root root 216 2005-02-17 22:48 ../
> -rw-r--r-- 1 root root 1939812 2005-04-21 10:25 fpm-calm-river.mp3
> -rw-r--r-- 1 root root 2582496 2005-04-21 10:25 fpm-sunshine.mp3
> -rw-r--r-- 1 root root 2217563 2005-04-21 10:25 fpm-world-mix.mp3
>
> Any clues ? Seems like it actions things but isn't playing the mp3
> files..
>
> Regards,
>
>
> Sahil Gupta
> VoiceValley
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Re: [Asterisk-Users] Can I hide caller id on the fly (per each use setting) on Bristuffed * and quadbri
The command you may play with, is CallingPres. The values that did work
for me, with a zaphfc an with Swisscom (telco), are:
- 0 - hide callerID
- 32 - show callerID
There is a quite good explanation you to calculate the presentation on:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20CallingPres
example (extensions.conf):
exten => .0_,1,CallingPres(32)
exten => .0_,2,SetCallerID(123456789)
exten => .0_,3,Dial(Zap/g1/${EXTEN})
exten => .0_,4,Hangup
Jean-Christophe
Robert Rozman a écrit :
> Hi,
>
> I wonder if I can hide caller id for just certain users. Can I
> override caller id setting for show or hide on the fly from dialplan ?
>
> Thanks in advance,
>
> regards,
>
> Rob.
>
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Re: [Asterisk-Users] Best Linux Distribution
Il personally use Mandrake 9.2 and it works perfectly. On Debian, we've never got the FritzCard USB2 ISDN card working, but nothing to do directly with Asterisk. The only performance issue I've got was while running X (many comments around this issue). JC [EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie to asterisk
KH Chow wrote: > Dear Sir / Madam, > I am a newbie in using Asterisk. I am interested in its SIP. > Before I start to use it, I would like to know whether the system can > work between two Linux box without any FXO and FXS card and just using > microphone which connect to the regular sound card? I am looking into > others applications and all of them are using at least one FXS card. > Sorry for such beginner problem and please help. > Thank you very much for your time. > Max Here are 2 apps you could try: KPhone: http://www.wirlab.net/kphone/what.html LinPhone: http://www.linphone.org/?lang=us&rubrique=1 If you want to try with H.323 (v4), you can give a try to GnomeMeeting, a great communication software: http://www.gnomemeeting.org/ JC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using the FLASH key on MGCP sends DTMF to the third party
This case happens on a Swissvoice IP10S phone. A key (F1) is defined as FLASH, which allows to transfer a call. This function works well, but the other phone will hear the DTMF keys, while the voice is muted. Any idea how to mute DTMF as well ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message signaling on MGCP (light on the phone)
Using the mailbox=... in mgcp.conf, Asterisk will tell the phone if there is any voicemail or not. This status is called "vmwi", and it works well. Therefore, the vmwi status is got only when the phone makes a transaction (ie. take a line and release it). At no time Asterisk sends the status to the phone while a voicemail incomes. Does anyone have any information about this ? Is it the same with SIP or other protocols ? Is it a function we could use to force the status check and send it to the phone ? Does anyone knows how it works on Cisco phones ? Thanks for any help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP native bridge
I'm desperate trying to understand the SIP native bridge. The Asterisk server get the client to bridge together, and everything is allright. But when a client hangs up, the second stay conntected forever. With any soft of hard we did get the same issue. Of course, disconnection works allright in any case with CAPI/CAPI, CAPI/SIP, CAPI/H.323, H.323-H.323, H.323/SIP, but not with SIP-SIP. Disabling the native bridge (noreinvite) works great also, but my goal is to use the native bridge. Any idea ? Jean-Christophe
Re: [Asterisk-Users] Gastman crashes on Win32
I didn't have any major trouble. Some functions seem unsupported by now, but I did play for more than 1 hour by monitoring calls, forcing redirections and connections, and it seems to be allright for such jobs. Although, I don't use a TDM card, but SIP and CAPI. Jean-Christophe - Original Message - From: "Edwin Silva" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 23, 2003 3:04 PM Subject: RE: [Asterisk-Users] Gastman crashes on Win32 Is Gastman at a usable level now? Have there been recent modifications? Last time I tried using it, it was causing strange errors on asterisk (in combination with the quad span TDM card and 2 PCI FXO's) -Original Message- From: rnc Info Lists [mailto:[EMAIL PROTECTED] Sent: Thursday, October 23, 2003 8:51 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Gastman crashes on Win32 Can anyone please point me toward the source/binary (linux and Win32) for Gastman?? Robert > Hi, > > The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all > my machines, no error, no log. > Although, the CVS version works great on Linux. > > Is it anybody who knows how to compile it with mingw32 ? Or better, could > anyone, who already has mingw32 installed, make a binary snapshot ? > > Thanks in advance, > > Jean-Christophe > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gastman crashes on Win32
by using the usual CVS: # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs co gastman or the released source and binary is available at: ftp://ftp.asterisk.org/pub/telephony/gastman/ Jean-Christophe - Original Message - From: "rnc Info Lists" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 23, 2003 2:51 PM Subject: Re: [Asterisk-Users] Gastman crashes on Win32 > Can anyone please point me toward the source/binary (linux and Win32) for > Gastman?? > > Robert > > > Hi, > > > > The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all > > my machines, no error, no log. > > Although, the CVS version works great on Linux. > > > > Is it anybody who knows how to compile it with mingw32 ? Or better, could > > anyone, who already has mingw32 installed, make a binary snapshot ? > > > > Thanks in advance, > > > > Jean-Christophe > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gastman crashes on Win32
Hi, The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all my machines, no error, no log. Although, the CVS version works great on Linux. Is it anybody who knows how to compile it with mingw32 ? Or better, could anyone, who already has mingw32 installed, make a binary snapshot ? Thanks in advance, Jean-Christophe
Re: [Asterisk-Users] what is the best codec for low bandwidth? for quality?
Working with X-Lite, iLBC is unuseable. The sound is completely scrambled, even without using Asterisk between 2 clients. While trying to use SPX, X-Lite connects to Asterisk, but no sound at all. Else, the GSM 06.10 is quite fair and works for everybody. Jean-Christophe - Original Message - From: "Jan Janak" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 23, 2003 9:49 AM Subject: Re: [Asterisk-Users] what is the best codec for low bandwidth? for quality? > From my experience iLBC is unbeatable on lossy and slow links. I have > been in situations where no other codec (GSM, Speex, G.729) worked and > iLBC was still fairly usable. > > Jan. > > On 22-10 23:29, Matthew Simpson wrote: > > The number of codecs is overwhelming to me. > > > > What do ya'll consider the best codec for conserving bandwidth? [I realize > > at the cost of quality] > > > > Secondly, what do you think the best codec for voice quality is? > > > > Yours, > > Matthew > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk and NAT on the same box?
Hi Tjardick, Do you mean that * will be used as a proxy by the way ? For the tests I have made, Asterisk tries to put both phones in relation together. Did I understand right ? Jean-Christophe - Original Message - From: "Tjardick van der Kraan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, October 22, 2003 6:14 PM Subject: Re: [Asterisk-Users] Running Asterisk and NAT on the same box? > Hi Steve, > > This is definately the way to go when you are dealing with NAT. > > I have simulair setups running like this myself and it works perfectly. > > Only think you need to do in the sip.conf entries is add > > canreinvite=no > > This will force any sip-calls from the outside to be routed thru *. > > Greetings, > > Tjardick > > -- > Tjardick van der Kraan > [EMAIL PROTECTED] > > IAXtel: 1 700 344 0522 > FWD: 26322 > IPtel: 91331 > > > - Original Message - > From: "Steven M. Sokol" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, October 22, 2003 5:22 PM > Subject: [Asterisk-Users] Running Asterisk and NAT on the same box? > > > > Has anyone tried installing * on a box with two eth interfaces which is > > acting as a NAT box? I have only one IP at this point and I would like > > to get * working without all of the NAT issues. My idea is to run * on > > my gateway (which is also running the firewall and masquerade services). > > All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the > > NAT screen, and will connect to the * using its PUBLIC (outside) > > address. > > > > Does this sound reasonable? > > > > Thanks, > > > > Steve > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music Onhold Configuration
Hi, I'm waiting for some hardware phones, so I missing some experience in such cases. But what you can do is to launch "sip debug" in the Asterisk's console, and watch at the logs. You might have a codec issue. Regards, Jean-Christophe - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, October 21, 2003 10:32 PM Subject: Re: [Asterisk-Users] Music Onhold Configuration > > Jean-Christophe, > > Thank you very much for your help. I configured the Music On Hold by > following your sample, it seemed work fine by looking at the Trace. But no > Music came up on my SIP phone SNOM200. I checked /var/lib/asterisk/mohmp3 > and found only one MP3 file there "sample-hold.mp3". Do you know what's > wrong with it? > > Thank you in advance, > Kang > > > > > > "Jean-Christophe Heger" > <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> > Sent by: cc: > [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Music Onhold Configuration > .digium.com > > > 10/20/2003 06:10 PM > Please respond to > asterisk-users > > > > > > > /etc/asterisk/musiconhold.conf > [classes] > default => mp3:/var/lib/asterisk/mohmp3 > > /etc/asterisk/extensions.conf > exten => 101,1,Answer > exten => 101,2,MusicOnHold(default) > > That's about what is said in the manual (RTFM ;-) and it works great. > > Jean-Christophe > > - Original Message - > From: <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Cc: <[EMAIL PROTECTED]> > Sent: Monday, October 20, 2003 10:56 PM > Subject: [Asterisk-Users] Music Onhold Configuration > > > > Anyone can share me with Music Onhold Configuration sample? > > > > Thanks in advance for your help, > > Kang > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call forwarding
Wipeout is right on the fact you cannot announce yourself with a phone
number which is not allowed by your operator, else anybody could announce
itself with any number. But it could be authorized for special cases.
Regarding your example, if you belong som MSNs 526xxx, and 226 matches the
MSN 526266 you could say:
exten => 910,2,Dial,CAPI/526${CALLERIDNUM}:bmymobilenr|20
or even
exten => 910,2,Dial,CAPI/${CALLERIDNUM}:bmymobilenr|20
The right syntax is ${...}, and not $(...).
Most of the time, you can use the revelant numbers where the point-to-point
switching is made. The rest of the MSN is automatically completed by the
operator.
Of course, the outgoing MSN must match the ones you've defined in capi.conf.
Try to announce the incoming MSN with the 3 last digits and see how it
works.
Jean-Christophe
- Original Message -
From: "JanM" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 21, 2003 12:05 PM
Subject: [Asterisk-Users] Call forwarding
Hi all,
I would like to forward an incoming call to my mobile with the incoming
callerID after ten seconds, I have tried with this:
exten => 910,1,Dial,SIP/[EMAIL PROTECTED]|10
exten => 910,2,Dial,CAPI/526$(CALLERIDNUM):bmymobilenr|20
exten => 910,3,Voicemail,u226
exten => 910,102,Voicemail,b226
The second row is not right and I can´t get it to work, any idea´s?
---JanM---
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Re: [Asterisk-Users] Music Onhold Configuration
/etc/asterisk/musiconhold.conf [classes] default => mp3:/var/lib/asterisk/mohmp3 /etc/asterisk/extensions.conf exten => 101,1,Answer exten => 101,2,MusicOnHold(default) That's about what is said in the manual (RTFM ;-) and it works great. Jean-Christophe - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Sent: Monday, October 20, 2003 10:56 PM Subject: [Asterisk-Users] Music Onhold Configuration > Anyone can share me with Music Onhold Configuration sample? > > Thanks in advance for your help, > Kang > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing around MSNs
Thanks, yes it helped a lot. I didn't understand the sense of contexts in such cases. Thanks and regards, Jean-Christophe - Original Message - From: "WipeOut" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, October 20, 2003 3:33 PM Subject: Re: [Asterisk-Users] Playing around MSNs > Jean-Christophe Heger wrote: > > > Let's say I have 3 IP phones (A, B, C) and 3 MSNs (1, 2, 3). > > > > How can I define that the incoming MSN 1 is redirected to A, 2 to B > > and 3 to C ? > > And how can I define that the A phone uses the outgoing MSN 1, etc ? > > > > Actually, I'm using the CAPI channel driver, but any help is welcome. > > > > Jean-Christophe > > The MSN will be passed to Asterisk when the call comes in, so in the > context where you send the inbound call create an entry similar to this.. > > exten => msn1,1,GoTo(context,exten,priority) > > So if your MSN was 555 and the extension was 22 in the [users] context > the line would be.. > > exten => 55,1,GoTo(users,22,1) > > Hope that helps.. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playing around MSNs
Let's say I have 3 IP phones (A, B, C) and 3 MSNs (1, 2, 3). How can I define that the incoming MSN 1 is redirected to A, 2 to B and 3 to C ? And how can I define that the A phone uses the outgoing MSN 1, etc ? Actually, I'm using the CAPI channel driver, but any help is welcome. Jean-Christophe
Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk
I'm also interested. - Original Message - From: "sip" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, October 17, 2003 7:55 PM Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk > count me in > - Original Message - > From: "Paulo Mannheimer" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, October 17, 2003 12:23 PM > Subject: [Asterisk-Users] Beta testers for visual configuration tool for > asterisk > > > > Hi All, > > > > We've been developing for a while an IDE for Asterisk, and the time has > > come to open it for beta testers. > > > > You can check at www.instant.com.br/viv.html for a snapshot of the > > application. > > > > Current modules are Dialplan and VoiceMail configuration. As you may > > see, it is all-visual, with drag and drop support and integrated sound > > recording, saving and cross-checking, so you dialpland doesn't crash > > because of a missing sound file. > > > > Beta users will have to download and install either a 16 Mb or a 4Mb > > Windows program, depending if you already have or not JRE 1.4.2 > > installed. This client works together with a tomcat-based application, > > which will be running on our servers during the trial. > > > > If you wish to participate, please let me know off-list. I'll get in > > touch with the first 5 answers to arrange how the test will be > > performed. > > > > Best, > > > > PauloHM > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > This message was checked by MailScan for WorkgroupMail. > www.workgroupmail.com > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
