Re: [Asterisk-Users] sound quality problem on mISDN
On Wed, Jun 14, 2006 at 10:04:04AM +0200, Kai Ober wrote: > Have you only one BN-Card? or more? I've one BN8S0 card and one TE110P - T1 connection to Rhino channelbank. > i have two cards, had compareable problems. > > PCM was the magic word ... > > from my misdn-init.conf: > > card=1,0x8,pcm_slave,ignore_pcm_frameclock //important! > option=9,master_clock // 9 > for port 9 > pcm=1,1 >//not sure, if this is really neaded Intresting I'm going to try this today . I thinking also about 'ulaw' option to 'card=' . My channelbank is T1 and this will eliminate transcoding from isdn to T1. thx for help. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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
[Asterisk-Users] sound quality problem on mISDN
Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), -> [ GSM Gateway ] ---> [ BN8S0 ] asterisk Port connected to GSM gatway is in TE mode , gateway is in NT mode , When I dialin to cellphone numer , call goes to 'from-eragsm' context, to Echo application. [from-eragsm] exten => 700,1,Goto(600,1) exten => 600,1,Answer() exten => 600,2,Playback(demo-echotest) exten => 600,n,Echo exten => 600,n,Playback(demo-echodone) misdn.conf: [eragsm-gw] ports=1ptp context=from-eragsm nationalprefix=0 internationalprefix=00 echocancel=yes echocancelwhenbridged=no dialplan=2 msns=600,700 Everything is good besides call quality, sound is choppy, with lot of noises, when I tell one , two , three ... test , I hear only three, sometimes more , I've already tried to increase rxgain/txgain for this channel , but It didn't help much. Outgoing call quality is rather normal. TIA for any help with this. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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] isdn out of band signalling
On Wed, Mar 15, 2006 at 02:01:50PM +1100, James Harper wrote: > This is more an isdn question than an asterisk specific one, but is > there any end to end signalling channel available during call setup? Eg > if AParty dials BParty, can any information be conveyed (in both > directions preferably, and in addition to CLI[PR]) before the call is > answered? > There is not such thing like A to B signaling, your phone/pbx is sending signaling messages to your provider pstn switch not to BParty. > The only way I can think of doing this is for the AParty to use CLIP to > present a different number to the BParty, and for the BParty to > terminate the call while ringing after a certain time (the time taken to > terminate forms the information from B to A). > On some E1/T1 , you can spoof callerid , everything depends on your provider. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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
[Asterisk-Users] isdn problem
Hi I have beronet BN8S0 isdn card in my asterisk and , card is working fine, but when I try to dial to special number 118913 ( telephone number information) from polish telecom TPSA, I always geting timeout . Bellow is isdn signaling dump : --> * CallGrp: PickupGrp: --> rxgain:0 txgain:0 --> * dad:118913 tech:mISDN/2-u25 ctx:default --> * Setting Context to from-tpnet --> TON: Unknown --> TON: Unknown --> PRES: Allowed (0x0) --> SCREEN: Unscreened (0x0) --> * adding2newbc ext 118913 --> * adding2newbc callerid 717201234 I SEND:SETUP oad:717201234 dad:118913 port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:0 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:0 addr:51400102 l3id:11000c --> new_l3id 11000e --> * SEND: State Dialing pid:43 I IND :SETUP_ACKNOWLEDGE oad:717201234 dad:118913 port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:1 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:43 addr:51400102 l3id:11000e I IND :TIMEOUT oad:717201234 dad:118913 port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:1 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:43 addr:51400102 l3id:11000e I IND :RELEASE oad:717201234 dad:118913 port:2 --> mode:TE cause:-1 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:1 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:43 addr:51400102 l3id:11000e Idx:0 stack->cchan:0 Chan:1 Idx:1 stack->cchan:0 Chan:2 Idx:0 stack->cchan:0 Chan:1 Idx:1 stack->cchan:0 Chan:2 I IND :CLEAN_UP oad: dad: port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:0 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:0 addr:51400102 l3id:11000e Trying to Release bc with l3id: 11000e * RELEASING CHANNEL pid:0 ctx:from-tpnet dad:118913 oad:118913 state: (null) --> * State Down --> Setting AST State to down * --> In State Dialin * --> Queue Hangup What I've tried to do: - set correct CallerID - use Dial app with option 's', 'n:h' , 'h' Without luck, but when I connect analog telephone to NT R-interface , after dialing number I have connection. Other thing is there is second similar number 118912 (abroad telephone number information ) , and I call to this number without problems. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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
[Asterisk-Users] sangoma a104 cards and ss7 signaling
Hi Sangoma a104 card have in product specyfication support for Line protocol SS7 , http://www.sangoma.com/products/p_aft-104-specs.htm [..] Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. [..] Anyone of you guys use line protocol SS7 for E1/T1 termination in asterisk ? As far I know asterisk don't have support for SS7 signaling, but my telco wants to setup E1 link with SS7 signaling and suggest sangoma a104. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Large country based dialplan
On Wed, Oct 12, 2005 at 09:13:44AM +0200, Erik wrote: > Dinesh Nair wrote: > > > > On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following: > > > >> Where I got the data from and all is also on that page if anyone wanted > >> to make their own lists. I would appreciate any updates or corrections > >> that anyone happens to notice. > > > > > > a simple modification which would make this a lot more international > > friendly would be the definition of a variable to hold the international > > access code and then using this code instead of _011 which is US-centric. > > > > Seems to be missing a lot of extensions for the Netherlands and my own region > code is listed as KPN Mobile :) The same for Poland, in list I've found only 6 major cities in Poland (Krakow/Rzeszow/Warsaw/Katowice/Gdansk/Wroclaw) but there is lot more zones : http://www.itu.int/itudoc/itu-t/number/p/pol/81563_ww9.doc or this : http://www.ertel.com.pl/python/prefkraj.py /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] iax problem
On Mon, Sep 26, 2005 at 11:02:47AM -0600, Rich Adamson wrote:
>
> > > For #2, incoming calls would be handled with:
> > > exten => 6789,1,Dial(SIP/1235)
> > >
> > Besides that :
> >
> > *CLI> iax2 show registry
> > Host UsernamePerceived Refresh State
> > X.X.X.X:4569 Username1 [MYIP]:456960 Registered
> > X.X.X.X:4569 Username2 [MYIP]:456960 Registered
> > X.X.X.X:4569 Username3 [MYIP]:456960 Registered
> >
> > source and destination ports for all 3 iax registrations are the same ,
> > and my isp see only one, becouse rest is overwriten.
>
> Have you tried using three different contexts for those in iax.conf?
>
>
Yes and result is as I suppose :
-- Accepting UNAUTHENTICATED call from X.X.X.X:
> requested format = ilbc,
> requested prefs = (ilbc|gsm|ulaw|alaw),
> actual format = ilbc,
> host prefs = (),
> priority = caller
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1", "IAX2/1237") in new stack
-- Called 1237
-- Call accepted by 192.168.57.238 (format gsm)
-- Format for call is gsm
-- IAX2/1237-8 is ringing
-- Hungup 'IAX2/1237-8'
Everything enters via last registred username 'Username3'.
/pch
--
Dyslexia bug unpatched since 1977 ...
exploit has been leaked to the underground.
___
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Re: [Asterisk-Users] iax problem
On Sun, Sep 25, 2005 at 07:26:12AM -0600, Rich Adamson wrote: > > Two approaches that have been rather common are: > 1. use the separate contexts for each did, > 2. in the register statement, add /1234 at the end; like > register => username:[EMAIL PROTECTED]/6789 > I don't think it will work , iax statement don't have exten on end. [..] register [:] @ [:] To register with another IAX server. [..] This is true for SIP but not for IAX. > For #2, incoming calls would be handled with: > exten => 6789,1,Dial(SIP/1235) > Besides that : *CLI> iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569 Username1 [MYIP]:456960 Registered X.X.X.X:4569 Username2 [MYIP]:456960 Registered X.X.X.X:4569 Username3 [MYIP]:456960 Registered source and destination ports for all 3 iax registrations are the same , and my isp see only one, becouse rest is overwriten. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] iax problem
Hi I've 3 iax connections to my provider , each of them have own DID , PH1<| | \/ PH2<-->|-| <---> ||<-- DID1 | A1 | <---> |ISP |<-- DID2 PH3<-->|-| <---> ||<-- DID3 I had iax phone on each of this connection , but now I want to terminate all on my asterisk box , and send calls to phones connected to my asterisk depending to incoming username/DID . for example : Call to DID1 must be directed to PH1 , DID2 to PH2 and DID3 to PH3 etc In iax.conf I have : [Username1] ;DID1 type=user username=Username11 ;secret=blah host=X.X.X.X context=fromisp1 [Username2] ;DID2 type=user username=Username2 host=X.X.X.X context=fromisp1 [Username3] ;DID3 type=user username=Username3 host=X.X.X.X context=fromisp1 For each of the iax connection I have defined section with type user. In extension.conf I have : [fromisp1] exten => s,1,Dial(SIP/1235) exten => _X.,1,Dial(SIP/1235) exten => h,1,Hangup Every incoming call enters context fromisp1 with exten 's' . I can't distinguish incoming DID or username, of couse I've figure out that I can create context for each iax connection , but for me I would be wast of cpu cycles :) Some other ideas for my problem ?:) /pch PS: This is my first post , don't shot me :) -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
