search for: settransfercapability

...bled echo cancellation on channel 73 Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Executing Goto("Zap/73-1", "to-intertel|154|1") in new stack Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Goto (to-intertel,154,1) Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Executing SetTransferCapability("Zap/73-1", "SPEECH") in new stack Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Setting transfer capability to: 0x00 - SPEECH. Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Executing Dial("Zap/73-1", "Zap/g3/5555154|15|r") in new stack Nov 12 12:48:38...

...0,500,300,500,-4000 ;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, m...

...istered custom function SORT [app_echo.so] => (Simple Echo Application) == Registered application 'Echo' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk) == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver' [app_settransfercapability.so] => (Set ISDN Transfer Capability) == Registered application 'SetTransferCapability' [app_url.so] => (Send URL Applications) == Registered application 'SendURL' [app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/app_md5...

...t;switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile operator. What I have tried so far ist to use SetTransferCapability(VIDEO) but this does not change the behavior. Is there a way to set or preserve the bearer capability for the outgoing call leg? cheerio Steve --- environment pbx-test*CLI> show version Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q built by root @ pbx-test.bb.ic3s.de on a i686 running Linux...

...Spawn extension (from-s0-faxmodems, 00242xxx, 5) exited 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 -- Requ...

...e D-Channel hardware HDLC - no change in behavior 4. Firmware versions 24 (shipped version) and 25 (latest version) on the Sangoma card - no change in behavior 5. callprogress and busydetect both 'yes' and 'no' in zapata.conf (currently 'no') - no change in behavior 6. Added SetTransferCapability(3K1AUDIO) to our dialplan, just to be sure - no change in behavior General environment: 1. We run TDM only, no VoIP protocols are in use. SIP, IAX2, MGCP are all noload in modules.conf. 2. This problem occurs with as few as one simultaneous channel active and as many as 15 simultaneous channels...

I'm trying to setup DATA calls with Dial(Zap/g1d/12345678), but with PRI DEBUG SPAN 1 on, it seems to connect a regular SPEECH call. I'm using 1.0.6. Is this feature broken in stable release? There seems to be support in the source, but it doesn't work. Does the Telco set what each PRI channel support? Like DATA or SPEECH etc.. Do I have to specify in zapata.conf or zaptel.conf that

...ll. 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 (libpri?) force the Transfer capability to the first value in any case ? After looking a bit in the source (channels/chan_zap.c) and in apps/app_settransfercapability.c it looks like I can do it with in the call sequence (extensions.conf) SetTransferCapability('SPEECH') Is there a better way ? Thank you.

...g unusual on the cli, is like any other call ended for a real busy condition. More weird is that with the SIP route the called phone rings once, than stops and I get the 486. What have I've already tried : Set(CALLERID(number)=[a real "traditional" phone number]) before the dial SetTransferCapability(SPEECH) as far as I know the route calls follow is : linksys pap --sip--> asterisk (1.2 or 1.0) --iax--> asterisk server (1.2) --zap--> ..?.. <---- Hangup cause 17 linksys pap --sip--> asterisk (1.2 or 1.0) --iax--> asterisk server (1.2) --sip--> ..?.. <----- 486 Busy...

...ip=x.x.x.x port=5060 disallow=all allow=ulaw canreinvite=no nat=no extensions.conf: [sip] exten => _X.,1,Dial(Zap/g2/01234567890) [zap] exten => _X.,1,Dial(SIP/1234@sip_proxy) Asterisk CLI: -- Accepting call from '1234' to '5678' on channel 0/23, span 3 -- Executing SetTransferCapability("Zap/85-1", "SPEECH") in new stack -- Setting transfer capability to: 0x00 - SPEECH. -- Executing Dial("Zap/85-1", "SIP/3456@sip_proxy") in new stack -- Called 3456@sip_proxy -- SIP/sip_proxy-57c5 is ringing -- SIP/sip_proxy-57c5 answered...

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

...Spawn extension (from-s0-faxmodems, 00242xxx, 5) exited 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 -- Requ...

...10 3K1AUDIO). This apparently screws the fax data signal, thus rendering faxing through asterisk practically useless. It also is dubious if filtering on a voice call might be helpful, for in theory it can severely disrupt/disable an echo cancellation algorithm. For anyone thinking of using the SetTransferCapability application in a dial plan to circumvent the problem by setting the TC of fax calls to DIGITAL (0x10), this is not an option. As soon as that call will arrive at another PBX with an analog fax machine attached to it, the call will be rejected (as per ISDN specification). Therefor it seems the o...

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

...heck' == Registered application 'MD5' [app_readfile.so] => (Stores output of file into a variable) == Registered application 'ReadFile' [app_chanspy.so] => (Tap into any type of asterisk channel and listen to audio) == Registered application 'ChanSpy' [app_settransfercapability.so] => (Set ISDN Transfer Capability) == Registered application 'SetTransferCapability' [app_dictate.so] => (Virtual Dictation Machine) == Registered application 'Dictate' [skipping app_zapras.so] [app_meetme.so] => (MeetMe conference bridge) == Registered applicat...