search for: 3k1audio

Displaying 14 results from an estimated 14 matches for "3k1audio".

2005 Jun 24
1
BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5
...xited non-zero on 'Zap/132-1' -- Hungup 'Zap/132-1' -- Executing SetCallerPres("Zap/129-1", "prohib") in new stack -- Executing NoOp("Zap/129-1", "") in new stack -- Executing SetTransferCapability("Zap/129-1", "3K1AUDIO") in new stack -- Setting transfer capability to: 0x10 - 3K1AUDIO. -- Executing SetCIDNum("Zap/129-1", "0410612345") in new stack -- Executing Dial("Zap/129-1", "Zap/r1/039xxxx") in new stack -- Requested transfer capability: 0x10 - 3K1A...
2005 Jul 01
1
Unable to forward frame/voice
...rrect 4. Zaptel.conf is set up with Sangoma Port A as the primary clock source, and Port B to not be used as a clock source 5. LBO, switch options, etc. are correct for the environment (since 98% of outbound calls are fine, this seems fairly obvious) 6. ISDN Transfer Capability gets properly set to 3K1AUDIO for calls 7. No IRQ sharing on the system 8. IDE DMA mode is irrelevant, since there are no IDE disks in the system (other than the CDROM) We have tried the following: 1. Asterisk, libpri and zaptel versions from 6/1/2005, 6/15/2005 and 6/28/2005 - no change in behavior 2. Wanpipe drivers 2.3.3-...
2009 Sep 29
1
Fax and dial-up connection issues
...Ring continuo cadence=10000,1,60000,1 callerid="" <7875> context=fax callwaiting=no callgroup=3 pickupgroup=3 mailbox=7875 channel => 125 /etc/asterisk/extensions.conf: [fax] ignorepat => 0 include => local ;Ligacoes locais exten => _0XXXXXXXX,1,SetTransferCapability(3K1AUDIO) exten => _0XXXXXXXX,n,Dial(DAHDI/g1/${EXTEN:1},60) ;Ligacoes DDD - telefones fixos exten => _00XX[2-6]XXXXXXX,1,SetTransferCapability(3K1AUDIO) exten => _00XX[2-6]XXXXXXX,n,Dial(DAHDI/g1/021${EXTEN:2},60) # dahdi_test: svoip01:~# dahdi_test -vv Opened pseudo dahdi interface, measuring...
2005 Jun 28
0
BRIstuff/OctoBRI problem: Ring requested on unconfigured channel 255/255 span 5
...xited non-zero on 'Zap/132-1' -- Hungup 'Zap/132-1' -- Executing SetCallerPres("Zap/129-1", "prohib") in new stack -- Executing NoOp("Zap/129-1", "") in new stack -- Executing SetTransferCapability("Zap/129-1", "3K1AUDIO") in new stack -- Setting transfer capability to: 0x10 - 3K1AUDIO. -- Executing SetCIDNum("Zap/129-1", "0410612345") in new stack -- Executing Dial("Zap/129-1", "Zap/r1/039xxxx") in new stack -- Requested transfer capability: 0x10 - 3K1A...
2005 Aug 29
3
How to use * and # as part of number indialcommand
...e, asterisk logs it as a > 'warning' , but for me it looks like it is linked to the problem. See > my comments in the logs between [ ]. > > -- Executing Dial("Zap/2-1", "Zap/4/*31*040268000") in new stack > -- Requested transfer capability: 0x10 - 3K1AUDIO > -- Called 4/*31*040268000 > -- Zap/4-1 is making progress passing it to Zap/2-1 > [thus far it looks okay] > -- Channel 0/1, span 2 got hangup > [hmm, it seems that the channel was hangup, so it failed] > Aug 27 23:32:28 WARNING[17591]: app_dial.c:412 wait_for_answer...
2011 Jul 03
1
SIP Peer Name Variable
Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Aug 23
0
[Asterisk-Dev] Severe ISDN signal distortion in CVS-HEAD with octoBRI
...tuff-0.2.0-RC8f-CVS/Asterisk-CVS-05.29.05). The Stable versions of the Bristuff patch do not have the problem (tested on Bristuff-0.2.0-RC8f/Asterisk-1.0.7 and Bristuff-0.2.0-RC8n/Asterisk-1.0.9). 4.Signal distortion is limited to calls with an ISDN Transfer Capability of 0x00 (SPEECH) or 0x10 (3K1AUDIO). Calls with an ISDN Transfer Capability of 0x08 (DIGITAL) are not being effected. 5.The amount of signal distortion is of such severity that it is almost impossible to reliably send a fax call through the ISDN card. 6.The amount of signal distortion also appears to render the echo cancellatio...
2005 Aug 01
0
How to force Requested transfer capability on BRI/PRI dial?
...following: - when a GSM phone or ISDN phone calls in, the Transfer capability is Requested transfer capability: 0x00 - SPEECH - when an analog phone calls in (either from an analog line or an analog ISDN port), the Transfer capability is Requested transfer capability: 0x10 - 3K1AUDIO The problem is that the ISDN phone in the internal bus apparently ignores this transfer capability in the SETUP message: it does not answer at all. It works well in the first case (GSM, etc) AFAIK there is no difference in the codec in both cases: just a type difference. How can I make Asterisk...
2005 Aug 28
1
How to use * and # as part of number in dialcommand
> Hi Damon and others, > > Your example is still doing what I tried already, so eventually the > dial command ends like: > Dial(zap/4/*21*) > or > Dial(zap/4/*31*) > I prefer to use Dial(zap/4/*21*<thenumber>) > or Dial(zap/4/*31*<thenumber>) > > But whatever I try, the error message as in my first post shows up and > the line hangs up before the
2007 Aug 30
0
WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries!
...8', 9 retries! -- Called g2/7141278 -- Zap/33-1 is proceeding passing it to Zap/30-1 -- Zap/33-1 is ringing -- Zap/33-1 answered Zap/30-1 And interesting what difference between -- Requested transfer capability: 0x00 - SPEECH and -- Requested transfer capability: 0x10 - 3K1AUDIO ??? Thanks! asterisk-users-request at lists.digium.com wrote: ------------------------------ Message: 18 Date: Wed, 29 Aug 2007 14:46:41 -0500 From: Russell Bryant <russell at digium.com> Subject: Re: [asterisk-users] WARNING[11439] To: Asterisk Users Mailing List - Non-Commercial Discuss...
2010 Mar 03
2
Best practise for ISDN Video Conferencing..
Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's
2006 Jan 16
2
Problem with calls starting from a legacy PBX
...led Party Number) -- Executing NoOp("Zap/60-1", "--> Thematica called from 0984899220 <--") in new stack -- Executing Dial("Zap/60-1", "Zap/g1/0984465691") in new stack -- Making new call for cr 33047 -- Requested transfer capability: 0x10 - 3K1AUDIO > Protocol Discriminator: Q.931 (8) len=34 > Call Ref: len= 2 (reference 279/0x117) (Originator) > Message type: SETUP (5) > [04 03 90 90 a3] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) > Ext: 1 Tr...
2005 Aug 27
3
How to use * and # as part of number in dial command
.... E.g.: 2000,1,Dial(Zap/4/*31*040268000) When I dial 2000 , the verbose logging shows below (Zap2-1 is my internal phone , Zap/4 is connected to outside ISDN line. -- Executing Dial("Zap/2-1", "Zap/4/*31*040268000") in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called 4/*31*040268000 -- Zap/4-1 is making progress passing it to Zap/2-1 -- Channel 0/1, span 2 got hangup Aug 27 23:32:28 WARNING[17591]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/4-1' == No one is available to answer at this time --...
2006 Jan 14
1
Problem with just one number!
...led Party Number) -- Executing NoOp("Zap/59-1", "--> Thematica called from 0984899416 <--") in new stack -- Executing Dial("Zap/59-1", "Zap/g1/0984465691") in new stack -- Making new call for cr 32960 -- Requested transfer capability: 0x10 - 3K1AUDIO > Protocol Discriminator: Q.931 (8) len=35 > Call Ref: len= 2 (reference 192/0xC0) (Originator) > Message type: SETUP (5) > [04 03 90 90 a3] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) > Ext: 1 Tra...