[Asterisk-Users] make available again a zap channel after a red alarm...
Hello, I finally arrived to convince a cellsocket for Nokia phones to work with a X101P card in an asterisk v1.0.7. The problem I have now is that cellsocket usually resets after receiving a call in the mobile. If asterisk by luck notices it, it issues an error message Detected Alarm on channel 4: Red Alarm and makes unavailable the Zap channel. The problem is solved by removing and puting back the cable, but this is not a solution for my case. 1. Is there a way to make asterisk to stop monitoring the status of the line? 2. Is there a way by software to make available again the Zap channel after a red alarm ? Thank you in advance, Dimitris ___ 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] ringing CAPI SIP channels together
Hello,
I have an asterisk box with ISDN (CAPI) phones (using chan_capi that has
them as Zap channels) and SIP phones connected to it. I want to have a
extension that rings 1 CAPI and 2 SIP phones together as a group.
I tried the following but when either the capi phone either one SIP
phone is busy, asterisk goes directly to the BUSY extension.
Any idea how to overcome this ?
Thank you in advance,
Dimitris
exten = s,1,Dial(Zap/g5/111694SIP/g4,20,Ttwr)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(su111694)
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Voicemail(sb111694)
exten = s-BUSY,2,Hangup
exten = s-CHANUNAVAIL,1,Playtones(dial)
exten = s-CHANUNAVAIL,2,Hangup
exten = s-CONGESTION,1,Playtones(congestion)
exten = s-CONGESTION,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(111694)
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
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Re:[Asterisk-Users] Broadvoice alternatives
For me it was impossible to make a call with broadvoice without the patch for the file channels/chan_sip.c Try to search the list, I had posted it February or March 2005, or see www.voip-info.org Dimitris ___ 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] my hfc card does not like Siemens
Retrans: 135352460 Busy: 135351948 Overlap Dial: -1 asterias*CLI zap show channels Chan Extension Context Language MusicOnHold pseudozapcha gr 1zapcha gr 2zapcha gr end-- when making a call from a sip phone I get the following: --begin- -- Executing Dial(SIP/9591-131b, Zap/1/211694) in new stack -- Making new call for cr 131 Protocol Discriminator: Q.931 (8) len=50 Call Ref: len= 1 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] [28 13 44 69 6d 69 74 72 69 73 20 4b 6f 75 6e 61 6c 61 6b 69 73] Display (len=19) [ Dimitris Kounalakis ] [6c 06 21 80 39 35 39 31] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '9591' ] [70 07 a1 32 31 31 36 39 34] Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '211694' ] -- Called 1/211694 No response to SETUP message NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending -- Channel 0/1, span 1 got hangup NEW_HANGUP DEBUG: Destroying the call, ourstate Call Initiated, peerstate Overlap sending NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending -- Hungup 'Zap/1-1' == No one is available to answer at this time ---end- ___ 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] possible bug in chan_capi concerning context handling
Thank you for your response Marco.
I do. The problem is that all incomings calls from ISDN are handled by the
default s extension in the context [default] and not by an s extension
in the context [isdn] or by the msm numbers as extensions in the context
[isdn].
So, what is the reason for the context directive in /etc/asterisk/capi.conf
to exit? What exactly is it used for?
Dimitris
*Marco Supino wrote:*
-
Do you have an 's' extention in the default context ?
Marco.
Dimitris Kounalakis wrote:
/ Hello,
// I am trying to configure asterisk 1.0.7pre to get incoming calls from an
// ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
// that the context is not recognised in the /etc/asterisk/capi.conf
// I have in /etc/asterisk/capi.conf 's section [interfaces] the
// following directive
// context=isdn
//
// and the following directive in /etc/asterisk/extensions.conf in the
// context [isdn]
// [isdn]
// exten = s,1,Dial(SIP/${DNID:4},60,tr)
//
//
// Here follows the debug info I get when an incoming call starts:
//
//
// == CONNECT_IND
// (PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
//-- creating pipe for PLCI=0x101 msn = 2810111694
//sent ALERT_REQ PLCI = 0x101
// == Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so
// falling back to exten 's'
// == Starting CAPI[contr1/2810111694]/3 at ,s,1 still failed so falling
// back to context 'default'
// Mar 13 11:52:41 WARNING[10744]: pbx.c:1893 ast_pbx_run: Channel
// 'CAPI[contr1/2810111694]/3' sent into invalid extension 's' in context
// 'default', but no invalid handler
//-- CAPI Hangingup
// -
//
// When I move the exten = s,1,Dial(${DNID:4},60,tr) in the context
// [default] of the /etc/asterisk/extensions.conf, I get the following
// debug info and the sip phone rings ok:
// --
//
// == CONNECT_IND
// (PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
//-- creating pipe for PLCI=0x101 msn = 2810111694
//sent ALERT_REQ PLCI = 0x101
// == Starting CAPI[contr1/2810111694]/4 at ,2810111694,1 failed so
// falling back to exten 's'
// == Starting CAPI[contr1/2810111694]/4 at ,s,1 still failed so falling
// back to context 'default'
//-- Executing Dial(CAPI[contr1/2810111694]/4, SIP/111694|60|tr) in
// new stack
//-- Called 111694
// --
//
//
// Is this a bug? It does not handle the context, so, it can not find what
// to do, it works only with the default context.
//
// Thank you in advance,
// Dimitris
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Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
Hello *Martijn, Thank you for your response. *That was my opinion too, it looses the context due to a bug, and can anyone confirm it also? But I have no output from the command Show channels, and it happens so quickly that it is impossible to issue the command before falling to the default context. In the logs, I can see that the channel exists like CAPI[contr1/2810211694]/0 but this is druring call only. Any other way to debug it more (or to solve it)? My /etc/asterisk/capi.conf is: - [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] controller=1 msn=2810111694 incomingmsn=* devices=2 softdtmf=1 callgroup=1 context=isdn -- *Martijn van Oosterhout wrote: *On Sun, Mar 13, 2005 at 06:44:52PM +0200, Dimitris Kounalakis wrote: / Thank you for your response Marco. // // I do. The problem is that all incomings calls from ISDN are handled by the // default s extension in the context [default] and not by an s extension // in the context [isdn] or by the msm numbers as extensions in the context // [isdn]. / Looking at the line here: / // == Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so // // falling back to exten 's' / It looks like the context is blank. What does the show command in asterisk show the context as being (paste output please). -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] Re: possible bug in chan_capi concerning context handling - SOLVED
Hello Stefan, Thank you for response. it helped me to solve it. The statement order was the problem here. I checked the source and I found that chan_capi separes config for different capi controllers with the directive devices. So the devices must be the last directive for each controller. After having devices in the last line, everything was ok. Thank you for your time, Dimitris Stefan Tichy wrote: -- Hello, On Sun, Mar 13, 2005 at 12:21:42PM +0200, Dimitris Kounalakis wrote: / I am trying to configure asterisk 1.0.7pre to get incoming calls from an // ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is // that the context is not recognised in the /etc/asterisk/capi.conf / Is this problem specific to asterisk 1.0.7pre? The statement order in /etc/asterisk/capi.conf may be relevant. Could you send the complete section of capi.conf. -- __ My e-mail address has changed. Please use the [EMAIL PROTECTED] or [EMAIL PROTECTED] e-mail . , [EMAIL PROTECTED] [EMAIL PROTECTED] ___ 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] asterisk and Broadvoice Outgoing Again :(
I never managed to make outgoing calls to broadvoice without the
following patch to the file channels/chan_sip.c
it comes from http://edvina.net/broadvoice/ and it is the only fraction
that it is still needed for outgoing calls.
It does not cause any problems with other sip devices that are connected
to my asterisk box.
if you do not patch it, then in sip debug you will notice that
broadvoice gives you an error message:
I do not remember it anymore, but it should be unauthorised or access
not allowed something like this.
--- channels/chan_sip.c.old 2005-03-12 18:10:49.0 +0200
+++ channels/chan_sip.c 2005-03-14 07:20:18.0 +0200
@@ -3701,16 +3701,28 @@
/* If we have full contact, trust it */
strncpy(invite, p-fullcontact, sizeof(invite) - 1);
/* Otherwise, use the username while waiting for registration */
- } else if (!ast_strlen_zero(p-username)) {
- if (ntohs(p-sa.sin_port) != DEFAULT_SIP_PORT) {
- snprintf(invite, sizeof(invite),
sip:[EMAIL PROTECTED]:%d,p-username, p-tohost, ntohs(p-sa.sin_port));
+} else {
+ /* If we have set the fromdomain, this is also used
+ as the to domain for SIP calls to a peer. Fromdomain
+ is used for calls to SIP proxys mostly
+ */
+ char fromdomain[256];
+ if (!ast_strlen_zero(p-fromdomain)) {
+ strncpy(fromdomain, p-fromdomain,
sizeof(fromdomain) -1);
} else {
- snprintf(invite, sizeof(invite),
sip:[EMAIL PROTECTED],p-username, p-tohost);
+ strncpy(fromdomain, p-tohost,
sizeof(fromdomain) -1);
+ }
+ if (!ast_strlen_zero(p-username)) {
+ if (ntohs(p-sa.sin_port) != DEFAULT_SIP_PORT) {
+ snprintf(invite, sizeof(invite),
sip:[EMAIL PROTECTED]:%d,p-username, fromdomain, ntohs(p-sa.sin_port));
+ } else {
+ snprintf(invite, sizeof(invite),
sip:[EMAIL PROTECTED],p-username, fromdomain);
+ }
+ } else if (ntohs(p-sa.sin_port) != DEFAULT_SIP_PORT) {
+ snprintf(invite, sizeof(invite), sip:%s:%d,
fromdomain, ntohs(p-sa.sin_port));
+ } else {
+ snprintf(invite, sizeof(invite), sip:%s,
fromdomain);
}
- } else if (ntohs(p-sa.sin_port) != DEFAULT_SIP_PORT) {
- snprintf(invite, sizeof(invite), sip:%s:%d, p-tohost,
ntohs(p-sa.sin_port));
- } else {
- snprintf(invite, sizeof(invite), sip:%s, p-tohost);
}
strncpy(p-uri, invite, sizeof(p-uri) - 1);
/* If there is a VXML URL append it to the SIP URL */
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[Asterisk-Users] possible bug in chan_capi concerning context handling
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the /etc/asterisk/capi.conf
I have in /etc/asterisk/capi.conf 's section [interfaces] the
following directive
context=isdn
and the following directive in /etc/asterisk/extensions.conf in the
context [isdn]
[isdn]
exten = s,1,Dial(SIP/${DNID:4},60,tr)
Here follows the debug info I get when an incoming call starts:
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/3 at ,s,1 still failed so falling
back to context 'default'
Mar 13 11:52:41 WARNING[10744]: pbx.c:1893 ast_pbx_run: Channel
'CAPI[contr1/2810111694]/3' sent into invalid extension 's' in context
'default', but no invalid handler
-- CAPI Hangingup
-
When I move the exten = s,1,Dial(${DNID:4},60,tr) in the context
[default] of the /etc/asterisk/extensions.conf, I get the following
debug info and the sip phone rings ok:
--
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/4 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/4 at ,s,1 still failed so falling
back to context 'default'
-- Executing Dial(CAPI[contr1/2810111694]/4, SIP/111694|60|tr)
in new stack
-- Called 111694
--
Is this a bug? It does not handle the context, so, it can not find what
to do, it works only with the default context.
Thank you in advance,
Dimitris
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[Asterisk-Users] chan_capi
Hello, I have installed asterisk but it is impossible to get compiled the chan_capi 0.3.5. I am using slackware 10.1 (kernel 2.4.29) and I am getting the following when I issue gcc -v Reading specs from /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/specs Configured with: ../gcc-3.3.4/configure --prefix=/usr --enable-shared --enable-threads=posix --enable-__cxa_atexit --disable-checking --with-gnu-ld --verbose --target=i486-slackware-linux --host=i486-slackware-linux Thread model: posix gcc version 3.3.4 The same happened also with Slackware 9.1 but I do not have anymore an installation to send info. I am getting the following errors and in the end of this message I have part of the file stddef.h. Any suggestion is welcomed. Thank you in advance, Dimitris Kounalakis -- [EMAIL PROTECTED]:~/chan_capi-0.3.5# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=athlon -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from /usr/include/time.h:38, from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/include/stddef.h:213: error: syntax error before typedef In file included from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/time.h:60: error: syntax error before typedef /usr/include/time.h:74: error: parse error before __BEGIN_NAMESPACE_STD /usr/include/time.h:76: error: syntax error before typedef /usr/include/time.h:129: error: parse error before __BEGIN_NAMESPACE_STD /usr/include/time.h:131: error: syntax error before struct /usr/include/time.h:178: error: parse error before __BEGIN_NAMESPACE_STD /usr/include/time.h:181: error: syntax error before extern /usr/include/time.h:181: error: parse error before __THROW /usr/include/time.h:184: error: parse error before __THROW /usr/include/time.h:188: error: parse error before __THROW /usr/include/time.h:191: error: parse error before __THROW /usr/include/time.h:199: error: parse error before __THROW /usr/include/time.h:226: error: parse error before __BEGIN_NAMESPACE_STD /usr/include/time.h:229: error: syntax error before extern /usr/include/time.h:229: error: parse error before __THROW /usr/include/time.h:233: error: parse error before __THROW /usr/include/time.h:248: error: parse error before __BEGIN_NAMESPACE_STD /usr/include/time.h:251: error: syntax error before extern /usr/include/time.h:251: error: parse error before __THROW /usr/include/time.h:254: error: parse error before __THROW /usr/include/time.h:272: error: syntax error before extern In file included from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/signal.h:31: error: parse error before __BEGIN_DECLS In file included from /usr/include/signal.h:33, from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sigset.h:23: error: syntax error before typedef In file included from /usr/include/bits/pthreadtypes.h:23, from /usr/include/pthread.h:25, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sched.h:83: error: syntax error before struct In file included from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: ... File: stddef.h 209: #ifndef __SIZE_TYPE__ 210: #define __SIZE_TYPE__ long unsigned int 211: #endif 212: #if !(defined (__GNUG__) defined (size_t)) 213: typedef __SIZE_TYPE__ size_t; 214: #ifdef __BEOS__ 215: typedef long ssize_t; 216: #endif /* __BEOS__ */ ___ 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
