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. ___ --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 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 user[:password] @ remote_host [:port] 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
