[asterisk-users] Query1
Hi, I am facing problem in configuring D-channel for TE120P card.I did the following things /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 then running the command ztcfg -vv I get the following output Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. Asterisk is running on host machine with DIGIUM card. DIGIUM card is connected through cable to E1 card running application on other end.On placing call from E1 card running application to asterisk PBX ( through DIGIUM card ) following error messages is coming on console mode of asterisk Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! NOTE- The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION Can anybody tell me how to overcome this error. Thanx and Regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query2
Hi, Do the following steps are required while configuring D-channel for TE120P card ( TE120P card is connected through cable to E1 card running application). These steps are written in TE120P card documnetation. 1) In zconfig.h file of zaptel package uncomment #define CONFIG_ZAPATA_NET make sethdlc-new make install 2) modprobe wcte12xp ztcfg 3) sethdlc hdlc0 cisco Step 3 is giving error hdlc0: Unable to set Cisco HDLC protocol information: No such device Can anybody tell, how to overcome this error. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query2
Hi, Do the following steps are required while configuring D-channel for TE120P card. Te120P card is connected to E1 card running application through cable.The following steps are written in the documentation of TE120P card 1) In zconfig.h file of zaptel package uncomment #define CONFIG_ZAPATA_NET make sethdlc-new make install 2) modprobe wcte12xp ztcfg 3) sethdlc hdlc0 cisco Step 3 is giving error hdlc0: Unable to set Cisco HDLC protocol information: No such device Can anybody tell, how to overcome this error. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] better subject needed [was: Re: Query1]
On Sat, Jul 28, 2007 at 12:03:33PM +0530, [EMAIL PROTECTED] wrote: Hi, I am facing problem in configuring D-channel for TE120P card.I did the following things /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf [ snip ] /etc/asterisk/zaptel.conf . Hmm.. sounds familiar. Haven't I answered it already. I also recall someone replying to it just today... You have already posted that question. Two of us have already posteated follow-ups on it. Please reply to (at least one) of them rather than re-posting your question. Furthermore, your posts have no meaningful subject. This post could use a subject such as: problem in configuring D-channel for TE120P card Or even just: configuring D-channel for TE120P card -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CONSOLE=Phone/phone0 and CONSOLE=Console/dsp and Zap/g2
Dear Jared; Thanks a lot for your kindly answer. Yes, but what does it mean: Phone/phone0 and Consol/dsp? Regards Bilal On Fri, 2007-07-27 at 06:46 -0700, bilal ghayyad wrote: What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 These are global variables as defined in the [global] section of extensions.conf. They're simply variables that can be used later on in your dialplan. I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? The g in is case is the syntax for telling Asterisk to dial out on a group of channels, not on an individual channel. (See the group= setting in zapata.conf.) The g option tells Asterisk to dial out on the lowest-numbered available channel in that channel group. You could also use: G - dial out on the highest-numbered available channel in the group. r - dial out on the lowest-numbered available channel in the group, but remember where we left off last time, and start back there when searching for a channel next time (round-robin) R - round-robin, highest to lowest -- Jared Smith Community Relations Manager Digium, Inc. Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Users Conference Friday at 12:30 PM EDT
On 7/27/07, dave cantera [EMAIL PROTECTED] wrote: randulo, I could not get into the conference today... the SIP line was busy, no matter what I do, the website thinks I'm not logged in and gives me the login page. after I login, anything I want to do brings me back to the login page... so I tried to re-setup the account thinking I wasn't logging in, and the user name was taken so I know I'm signed up. Dave, I answered privately to get more details, but if anyone is having SIP problems, for reference: I have logged in successfully using asterisk hundreds of times. That info is at http://x2z.eu I've also used numerous SIP clients including the Java one called ShoePhone built in to the conference interface (Win/Mac only) and X-Lite, Gizmo project, Idefisk/Zoiper and some group meet freeware for the Mac, so it wouldn't seem to be a problem on the SIP server side. I know tzafrir has had problems with the SIP and we can't figure out why it doesn't work for him. However, IIRC, his issue is not an apparent busy signal, but an auth problem. That busy signal means you are not reaching the server (unless you see other messages). You can call the SIP server anytime and it will always answer. You can not however enter a conference until the host is there. hth, randy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] global variables and updates
Sorry if this appears twice - I originally sent it nearly 18 hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as 1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
Julian.
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Re: [asterisk-users] Queue stats
Jay Moore wrote (received 2007-07-28): My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. So why drop them out of the queue? How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. There are various ways to solve this. I would suggest to get familiar with the queue log (/var/log/asterisk/queue_log). You could use a script in whatever language you like to read that file. Or you could write a script to import the log entries into an SQL database. (There are some scripts around to do that, search for asterisk queue_log mysql.) Or you could call a custom logger script directly from the dialplan with TrySystem() for example. Another option would be to listen to events from the manager interface (AMI) but that's probably not what you are looking for. Hope that helps. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] global variables and updates
Julian Lyndon-Smith wrote:
Sorry if this appears twice - I originally sent it nearly 18 hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as 1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
Someone more knowledgeable about Asterisk than I can correct me, but I
would look at it from the perspective any development environement:
Global variables are typically bad in a threaded environment without
some form of queuing/locking/critical section functionality to avoid
collisions.
If I needed a globally unique, sequential number, I'd push it out to
AGI/FastAGI so I could use a language with support for locking/queuing.
A DB like MySQL or FirebirdSQL would easily handle this need as you
know, but then is the overhead of establishing DB connections worth it
for simply getting a incremented int?
--
Warm Regards,
Lee
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Re: [asterisk-users] global variables and updates
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose it
to anyone else. If you received it in error please notify us immediately and
then destroy it.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian Lyndon-Smith
Sent: Saturday, July 28, 2007 5:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] global variables and updates
Sorry if this appears twice - I originally sent it nearly 18
hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I
have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as
1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
The changing of variables is not atomic as would hope, but there is a
solution for you. Look the application MacroExclusive. Put your Set to
increment the global variable inside of a macro and call it using this,
and you will get the behavior you desire. One caveat, however, is that
you will want as little logic as possible inside of this macro.
MacroExclusive will block all other calls to this macro until the first
one exits. But this is not an issue if all you are doing is a quick
var++ and then leaving.
- Brad
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[asterisk-users] New York Asterisk Meetup Aug 9th
Thursday, August 9, 2007, 7:00 PM http://asteriskpbx.meetup.com/2/calendar/6012673/ See you there Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] global variables and updates
Watkins, Bradley wrote:
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose
it to anyone else. If you received it in error please notify us immediately
and then destroy it.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian Lyndon-Smith
Sent: Saturday, July 28, 2007 5:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] global variables and updates
Sorry if this appears twice - I originally sent it nearly 18
hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I
have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as
1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
The changing of variables is not atomic as would hope, but there is a
solution for you. Look the application MacroExclusive. Put your Set to
increment the global variable inside of a macro and call it using this,
and you will get the behavior you desire. One caveat, however, is that
you will want as little logic as possible inside of this macro.
MacroExclusive will block all other calls to this macro until the first
one exits. But this is not an issue if all you are doing is a quick
var++ and then leaving.
That's a very nice feature. A quick Google search on the wiki didn't
turn up any topics. Does it queue subsequent calls or just block them
and then logic in the dialplan must be used against a return value?
---
Warm Regards,
Lee
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Re: [asterisk-users] Need help with inbound IAX
Nope, that's good. It means you have registered to their server no problem. Firstly, which version of Asterisk are you using? Version 1.2.7 If you turn on iax2 debug, and then say call from your cellphone to the DDI you have registered do you get anything at all? No, I do not see anything on the console when i use iax2 debug, not when I call the number. I do see some messages when I do a reload chan_iax2.so that I think are related to registration. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue stats
Hello Jay, you may want to have a look at QueueMetrics - everything you're looking for is already there. :-) l. On Thu, 26 Jul 2007 16:37:56 +0200, Jay Moore [EMAIL PROTECTED] wrote: Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. Thanks in advance, Jay -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN: Problems starting off
On Fri, 27 Jul 2007, Bertram Scharpf wrote: Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: please provide your capi.conf. Which chan-capi version do you use? [capi-in] exten = 9876543,1,Goto(demo,1000,1) where 9876543 is my MSN without the area prefix. `demo' is the context that plays `demo-congrats'. The debug output I yield ends with (after a pause) DISCONNECT_IND ID=001 #0x0027 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 DISCONNECT_RESP ID=001 #0x0027 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup. CAPI/ISDN1/9876543-2: set channel task to 1 == ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 4 == ISDN1#02: Interface cleanup PLCI=0x101 CAPI devicestate requested for ISDN1/9876543 Seems that the MSN or even `capi-in' cannot be found at all. Yes, chan-capi seems to wait because of no match. Could anyone give me a hint what is going wrong here or at least what I have to diagnose next? The full debug log together with capi.conf should help. Armin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] global variables and updates
Not sure what you are doing with meetme but,
i Always used AstDB() for this type of needs.
On 7/28/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Watkins, Bradley wrote:
The contents of this e-mail are intended for the named addressee only.
It contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose
it to anyone else. If you received it in error please notify us immediately
and then destroy it.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian Lyndon-Smith
Sent: Saturday, July 28, 2007 5:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] global variables and updates
Sorry if this appears twice - I originally sent it nearly 18
hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I
have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as
1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
The changing of variables is not atomic as would hope, but there is a
solution for you. Look the application MacroExclusive. Put your Set to
increment the global variable inside of a macro and call it using this,
and you will get the behavior you desire. One caveat, however, is that
you will want as little logic as possible inside of this macro.
MacroExclusive will block all other calls to this macro until the first
one exits. But this is not an issue if all you are doing is a quick
var++ and then leaving.
That's a very nice feature. A quick Google search on the wiki didn't
turn up any topics. Does it queue subsequent calls or just block them
and then logic in the dialplan must be used against a return value?
---
Warm Regards,
Lee
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Re: [asterisk-users] Configuring Sangoma A101D with Asterisk 1.2.18 zaptel-1.2.17.1
Deepak Naidu wrote: Hi, I have a Dell Power Edge server planning yo buy Sangoma A101D card. To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX setup. It would help to know exactly what Dell Poweredge you were considering. They do vary. If you compile your kernel with SMP and IO-APIC support, you shouldn't have any problems. The Sangoma cards are very tolerant. So I wanted to know the steps any issue which I may come accross if any. I have googled have some docs handy wrt Trixbox-2.2. Just wanted to get some notes from user with custom install setup when used with Asterisk+freepbx+Sangoma. On the hardware side, the experience shouldn't be any different. Also how do I enable DTMF hardware detection. As far as I know, that is the default. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Viable Alternatives to TDM400P
Corporate IT Solutions - Michael Dunne wrote: I have now within 18 months had a second TDM400P die, the first time was random call drops, and now it will not go off hook when making a call. To summarise, the card stopped making calls, I replaced the computer hardware, installed new OS and new Asterisk (from 1.2 to 1.4) without making a difference, the only factor in common is the TDM400P ... oh the card will receive calls just fine, so it's not a surge that has blown anything. Anyway, are there any viable alternatives to the Digium cards for analogue termination as yet. I need a minimum of 3 FXO ports. Are the Sangoma cards any good (I noticed a 5 year warranty on those ones)? I've used both and I can recommend them. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID from POTS to SIP
[EMAIL PROTECTED] wrote: Thanks for the reply. Unfortunately that didn't work. What's confusing is that for the line without any distinctive ring that works correctly with callerid, the only thing it does is dial the phones, so here's the entire context: [add-incoming] exten = s,1,Dial(SIP/ht1SIP/ht2SIP/gxp1,20) The other context, the one with distinctive ring that's not passing caller id, actually does a little more: [main-open] exten = s,1,Answer exten = s,n,Wait(3) exten = s,n,Background(opengreeting) exten = s,n,Dial(SIP/ht1SIP/gxp3,20) But even if I remove those extra bits from that context and make it look like the other one, it still doesn't work. Any more suggestions? What happens when you connect a regular caller ID device to the line and call the distinctive ring number? Is the telco even supplying CID info? This sounds like a programming problem on the telco side. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming call detected
Erick Perez wrote: Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. Let's see some CLI debug output. Do asterisk -r and try and make a call in, then post the result. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can someone Stop this autoreply stuff?????
Cheikhou DIAW wrote: hi , i think everybody is receiving theses mails from rp. can someone unsubscribe or do something , its really annoying Now I know why I had 600 messages in my Asterisk folder after only three days away. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
If you do not have any alarms and PRI debug span 1 still gives you nothing then you need to call your telco and say I'm not getting any Q.931 messages on the D-Channel. Stephen Bosch wrote: Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help with inbound IAX
Patrick Buller wrote: Nope, that's good. It means you have registered to their server no problem. Firstly, which version of Asterisk are you using? Version 1.2.7 That is super old. Did you install it from a package? I recommend you upgrade now, because you will have to later, I guarantee it. -stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help with inbound IAX
On 7/26/07, Patrick Buller [EMAIL PROTECTED] wrote: This is what callwithus is supposed to forward the call to: IAX/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED] Does that need to be IAX2/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED] ? Notice the 2? It used to be that IAX referred to v1 of the IAX protocol, which has since been obsoleted. I vaguely recall discussions that IAX2 and IAX were to be made synonymous now that v1 is obsolete for quite a while. Didn't follow those discussions to know whether that happened or not. Might be worth a try changing your config with callwithus to IAX2/... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling to users in other asterisk servers
Hello, Do you have porf forwardin for SIP protocol in your firewall? SIP: 5060 udp rtp 1 - 2 udp (default) and IAX2 4569 udp Best Regards Carlos Rojas On 7/28/07, Ary Junior [EMAIL PROTECTED] wrote: Hi, Im a asterisk newbie and I've configured an asterisk server here in my house... in my LAN two users can login and call to each other, but when I try to call an user in another asterisk server outside my LAN ( sip:[EMAIL PROTECTED] ) it dont work... if the person outside is conected on my server it works fine... My asterisk server is behind a firewall and portfowarding... it is possible? Thanks very much!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling to users in other asterisk servers
aryjunior, is your dialplan and registration configured to connect to another * server?...include your config so we can analyze it... daveC Carlos Rojas wrote: Hello, Do you have porf forwardin for SIP protocol in your firewall? SIP: 5060 udp rtp 1 - 2 udp (default) and IAX2 4569 udp Best Regards Carlos Rojas On 7/28/07, *Ary Junior* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Im a asterisk newbie and I've configured an asterisk server here in my house... in my LAN two users can login and call to each other, but when I try to call an user in another asterisk server outside my LAN ( sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] ) it dont work... if the person outside is conected on my server it works fine... My asterisk server is behind a firewall and portfowarding... it is possible? Thanks very much!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.22/921 - Release Date: 07/26/2007 11:16 PM -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connections broken
michael, this is what I use for centOS 4, but I think its too loose... let me know if you don't know where to put it... daveC # for asterisk -A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 4569 -j ACCEPT IAX -A RH-Firewall-1-INPUT -p udp -m udp --dport 5036 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5004 -j ACCEPT Michael Munger wrote: It did change, which is what caused this problem in the first place, but all the updates have been applied, propagated, and are working….well, with the exception of this one. Does anyone know what the iptables command would be to forward these IAX packets to a specific LAN ip? Michael Munger High Powered Help, Inc [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 404-438-2128 x 101 *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Dave Bour *Sent:* Thursday, July 26, 2007 12:29 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] IAX connections broken Are sites listed by IP or DN. If IP, dumb question but did it change? If DN, can you resolve it from the respective boxea? Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: [EMAIL PROTECTED] [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu Jul 26 10:17:23 2007 Subject: Re: [asterisk-users] IAX connections broken Not likely. #1, I have a public IP on that firewall. #2. If I block 4569 at our firewall, then it goes from closed to stealth. If I forward the port, it goes from stealth to closed. The iaxping tool (http://www.bpvn.com/asterisk/iaxping.zip) has no problems pinging the box from the lan, and our test machine can make an IAX connection to the box. From outside the network, however, it times out. It has to be a NAT problem, but forwarding doesn't appear to be working. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Thursday, July 26, 2007 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connections broken what if your internet provider is blocking inbound 4569 ? -- On 7/26/07, Michael Munger wrote: Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says Request sent. The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but we've got 4569 TCP UDP forwarded to our Asterisk box. When I use Shields up from GRC.com to test the port, it is showing up as closed rather than open, which normally means the port is open, but the service is not running, yet Asterisk is up and running just fine, and my outbound connections to Voicepulse work fine. I see voicepulse, voicepulse sees me. There is something I am not seeing here. Any thoughts? -Michael ___ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.22/921 - Release Date: 07/26/2007 11:16 PM -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] queue stats
I am submitting a patch to the Bug tracker next week that will have a manager event fired alongside every queue log write. You can then send the queue information to the database in realtime if you have a manager interface script. If anyone is willing to test this patch once posted, I would appreciate it. Anthony -- Original Message -- From: Philipp Kempgen [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sat, 28 Jul 2007 12:13:41 +0200 Jay Moore wrote (received 2007-07-28): My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. [message truncated] Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.X support for Solaris 10?
I've been trying to get Asterisk 1.4.X running under Solaris 10 x86 with
limited success.
I can build Asterisk and get it started but have run in to a problem
with a segmentation fault with the help command in the CLI.
When I start Asterisk:
# ./asterisk -vvvgc
Asterisk 1.4.9, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
== Parsing '/var/etc/asterisk/asterisk.conf': Found
.
.
.
== Registered application 'Skel'
app_skel.so = (Skeleton (sample) Application)
Asterisk Ready.
*CLI
If I type help
*CLI help
! Execute a shell command
abort halt Cancel a running halt
ael debug contexts Enable AEL contexts debug (does nothing)
.
.
.
say load set/show the say mode
show parkedcalls Lists parked calls
Segmentation Fault - core dumped
#
This problem only seems to occur with the help command in the CLI.
gdb shows this:
gdb ./asterisk core
GNU gdb 6.2.1
Copyright 2004 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain
conditions.
Type show copying to see the conditions.
There is absolutely no warranty for GDB. Type show warranty for details.
This GDB was configured as i386-pc-solaris2.10...
Core was generated by `./asterisk -vvvgc'.
Program terminated with signal 11, Segmentation fault.
Reading symbols from /usr/lib/libcurses.so.1...done.
.
.
.
Loaded symbols for /opt/asterisk/lib/modules/app_skel.so
#0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1
(gdb) bt
#0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1
#1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1
#2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1
#3 0x080e994a in ast_dynamic_str_thread_build_va (buf=0x817625b, max_len=0,
ts=0x8149720, append=0, fmt=0x811eefd %25.25s %s\n,
ap=0x8046f18 Pb\027\b) at utils.c:969
#4 0x08089ad8 in ast_cli (fd=1, fmt=0x811eefd %25.25s %s\n) at cli.c:69
#5 0x0808d33e in help1 (fd=1, match=0x8047084, locked=1) at cli.c:1746
#6 0x0808d45f in handle_help (fd=1, argc=0, argv=0x8047080) at cli.c:1773
#7 0x0808e05c in ast_cli_command (fd=1, s=0x0) at cli.c:1979
#8 0x08074127 in main (argc=135688218, argv=0x80471fc) at asterisk.c:1388
(gdb) q
#
The segmentation fault is caused by the call to vsnprintf in this
function in utils.c:
int ast_dynamic_str_thread_build_va(struct ast_dynamic_str **buf, size_t
max_len,
struct ast_threadstorage *ts, int append, const char *fmt,
va_list ap)
{
int res;
int offset = (append (*buf)-len) ? strlen((*buf)-str) : 0;
#if defined(DEBUG_THREADLOCALS)
struct ast_dynamic_str *old_buf = *buf;
#endif /* defined(DEBUG_THREADLOCALS) */
res = vsnprintf((*buf)-str + offset, (*buf)-len - offset, fmt,
ap);
/* Check to see if there was not enough space in the string
buffer to prepare
* the string. Also, if a maximum length is present, make sure
the current
* length is less than the maximum before increasing the size. */
if ((res + offset + 1) (*buf)-len (max_len ? ((*buf)-len
max_len) : 1)) {
/* Set the new size of the string buffer to be the size
needed
* to hold the resulting string (res) plus one byte for the
* terminating '\0'. If this size is greater than the
max, set
* the new length to be the maximum allowed. */
if (max_len)
(*buf)-len = ((res + offset + 1) max_len) ?
(res + offset + 1) : max_len;
else
(*buf)-len = res + offset + 1;
if (!(*buf = ast_realloc(*buf, (*buf)-len +
sizeof(*(*buf)
return AST_DYNSTR_BUILD_FAILED;
if (append)
(*buf)-str[offset] = '\0';
if (ts) {
pthread_setspecific(ts-key, *buf);
#if defined(DEBUG_THREADLOCALS)
__ast_threadstorage_object_replace(old_buf,
*buf, (*buf)-len + sizeof(*(*buf)));
#endif /* defined(DEBUG_THREADLOCALS) */
}
/* va_end() and va_start() must be done before calling
* vsnprintf() again. */
return AST_DYNSTR_BUILD_RETRY;
}
return res;
}
I think the fault is caused by a NULL pointer somewhere,but I can't
figure-out what's wrong.
Can anyone help?
Frank
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