Re: [asterisk-users] SET with pipe symbol
Tilghman,
Tx, That was the solution.
Kind Regards,
Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: dinsdag 29 januari 2008 16:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SET with pipe symbol
On Tuesday 29 January 2008 08:32:44 Arjan Kroon | Mobillion wrote:
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see the following output in my CLI :
Ignoring entry '612345678' with no = (and not last 'options'
entry)
(in my test call ${CALLERID(number) = 061234578)
I tried to escape the pipe symbol by using \ (backslash)
With the same result
Also I tried to place the variable between single or double quotes,
but
with the same result.
Does anybody now how place a pipe symbol in variable.
You can't, in 1.4. This is by design. We have removed this restriction
in
1.6. As a workaround, in 1.4, use the NoOp instruction with the SET
dialplan
function, i.e.
NoOp(${SET(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))})
--
Tilghman
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Re: [asterisk-users] Problem with DTMF dialing
On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost routing. The problem I am having is dialing out using DTMF signalling. At the moment I am making do with Pulse dialing through the 3 analog lines. I can recieve calls on the Cellphone line without any problems, but cant dial out through it, as a cellphone cant do pulse dialing. I have run ztmonitor 1 -f gains, where 1 is the zap channel where the cellphone is located, while dialing the number 072 031 1294. I then went to audacity, on my own pc, and converted the raw file into mp3 format, mp3 is a compressed format, and hence may lose some quality. Generally you should stick with wav. ztmonitor should spit the appropriate sox command to do the conversion. Maybe it would look slightly different in the original format. which is available for download at http://www.iancoetzee.za.net/tone_dial.mp3. After listening to the playback I concluded that the DTMF signals being sent is totally wrong. Is that the whole tone? It is too short to be a valid DTMF. -- Tzafrir Cohen icq#16849755 jabber:[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
[asterisk-users] Facing problem in installing asterisk-addons
Hi, I have installed GNU gatekeeper. Then I am trying to install asterisk addons. I gave make and then make clean. I worked properly. Then I gave make install. It gave following error. make[1]: Entering directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c' make: *** [install] Error 2 Please help me in understanding the solution for this. Thanking you, Regards, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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] Can't read environment variable
On Wed, Jan 30, 2008 at 01:38:48PM +0100, Joost Kuif | Mobillion wrote:
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME' gives the
hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone know what i am missing?
Is that variable set?
cat /proc/PID_OF_ASTERISK/environ | tr '\0' '\n' | grep ^HOSTNAME=
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Can't read environment variable
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME' gives the
hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone know what i am missing?
Ipv een saaie e-mail een leuk videobericht? Ga naar www.KletsKoppies.nl
http://www.kletskoppies.nl/
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Re: [asterisk-users] Can't read environment variable
You're using the variables wrong.
This is what you could do is either:
exten = s,1,NoOp(${ENV(HOSTNAME)})
or
In globals section
;; Defing hostname
host=${ENV(HOSTNAME)})
In you dailplan section
exten = s,1,NoOp(${host})
You will manage,
Greets
Joris
Joost Kuif | Mobillion wrote:
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME'
gives the hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone know what i am missing?
Ipv een saaie e-mail een leuk videobericht? Ga naar/
//www.KletsKoppies.nl http://www.kletskoppies.nl//
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Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
On Jan 29, 2008 8:36 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Jan 29, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote: Franklin, Because ChanSpy() is a passive monitor, there is nothing about the implementation that would cause Asterisk to shunt the speech back to itself. Asterisk only does this in situations where it is out of the media path and needs to insinuate itself back into it for the purpose of generating media, such as on-hold music, IVR, etc. What you're wanting should, in my opinion, basically be submitted as a feature request. Perhaps the developers can add a flag to the ChanSpy() invocation repertoire to make this work. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Alex, he was not asking why, it is obvious he knows why. He was asking for a solution or idea on how to work around this issue. Are you using Sangoma cards? If so, I might have a very good answer for you, as well as another very possible different solution. Both would be outside of Asterisk so some kind of magic would have to happen to associate the call being spied on to the channel but that should not be that difficult if you even need it. Another solution is to track down the code referenced here http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a reinvite back to asterisk before starting the spy. Anyways, I am sure it can be done. The question is how much time is it worth to make it happen. Maybe we should meet for lunch this week. I can meet you in cow country or Philly if you want, your choice. I have to go to both this week anyways and would like to catch up with things since Astricon. Thanks, Steve Totaro I just confirmed that there is a solution that is perfect for this that has been developed with a web interface to select the call to monitor. A little added code and you can pretty easily look up who the agent handling the call is. Let's test it out on your call center. Again, it is not an Asterisk app and would have no impact on your operations if it does not work. Thanks, Steve Totaro ___ -- 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't read environment variable
Maybe...
exten = s,n,NoOp(SET(host=${ENV(HOSTNAME))})
?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joost Kuif
| Mobillion
Sent: 30 January 2008 12:39
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Can't read environment variable
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME'
gives the hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone know what i am missing?
Ipv een saaie e-mail een leuk videobericht? Ga naar
www.KletsKoppies.nl http://www.kletskoppies.nl/
kuif_joost.gif___
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Re: [asterisk-users] Can't read environment variable
This pointed me into the right direction, thanks Tzafrir!
i added a export HOSTNAME=$HOSTNAME into my .bash_profile
Grtz,
Joost
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Tzafrir Cohen
Verzonden: Wednesday, January 30, 2008 1:58 PM
Aan: asterisk-users@lists.digium.com
Onderwerp: Re: [asterisk-users] Can't read environment variable
On Wed, Jan 30, 2008 at 01:38:48PM +0100, Joost Kuif | Mobillion wrote:
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME' gives
the hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone know what i am missing?
Is that variable set?
cat /proc/PID_OF_ASTERISK/environ | tr '\0' '\n' | grep ^HOSTNAME=
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Can't read environment variable
On Wed, 30 Jan 2008, Joost Kuif | Mobillion wrote: This pointed me into the right direction, thanks Tzafrir! i added a export HOSTNAME=$HOSTNAME into my .bash_profile Grtz, Joost I explicitly pass it to Asterisk in this snippet from my /etc/init.d/asterisk file. daemon\ env -i\ HOST=$HOST\ PATH=/usr/local/bin/:/bin/\ nice --adjustment=-20\ $ASTERISK $START_OPTIONS Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Dialogic card
Steve Totaro wrote: I was under the impression that only ABE supports Dialogic boards. I thought I saw that in passing so I could be totally wrong. There was talk but ABE has never supported Dialogic cards. If anyone would be interested, I would recommend expressing it to [EMAIL PROTECTED] -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] sipsock_read: BAD! BAD! BAD!
Douglas Garstang wrote: Does anyone know the cause of these BAD BAD BAD messages? I think I lost all my calls when it happened too. We have nagios running against IAX and nagios reports that IAX is down. It would seem that the entire application locks up when this happens and calls are dropped. Yes, that message generally indicates a deadlock in Asterisk. Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846) There have been 380 fixes made to Asterisk 1.2 since 1.2.14, 61 of which were made to chan_sip. :) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] Using two SIP-Domains with asterisk
Hi, is it possible to use asterisk to serve two SIP-domains with different users? It does work to define two domains with 'domain=' in sip.conf, but that allows all sip-users to register with both domains. I want to define users for a one domain only and not allow them to use the second. Lets say I have domains domainA.org and domainB.org and a sip-user user1. User1 should be able to register as [EMAIL PROTECTED] but not as [EMAIL PROTECTED] In addition I want to be able to define a second (different) user with the same username but another domain. How can I do that? Is it possible at all? thanks Bjoern ___ -- 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] POE draw on Aastra 480i
Octavio Ruiz wrote: Allen Casteran wrote: Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Drew wrote: According to Aastra tech support, 5 watts (peak) per 480i. We are testing five phones running on a Linksys SRW208P that will only support full 15W on up to 4 of 8 ports. I can power up the switch while all phones are connected without any issues. I would expect your lower power switch will provide ample power. But, PoE class does not matter? Did you plug five Aastra phones? I'm suspicious about how that scenario worked, I mean, as far as i know Aastra phones should register as a zero PoE class, that means it would reserve up to 12.94 watts no matter how many watts uses. So, my guess here is even if the phone use only 5 watts, the switch already reserved 12.94 watts for it. I would love to see what happens if you plug a sixth phone or figure out if you used an Aastra phone. Can you tell us what model/brand you used? Dimensioning PoE devices over capable switches has been a new issue which involves many factors like those described before. Regards, PD. Sorry about the original thread break off, I've been unable to find the original one. Hi Octavio, We are using Aastra 480i phones powered by PoE. During the testing we decided to drop the Linksys SRW208P as it is extremely noisy due to the fans, not suitable for an office environment. We continued testing with Netgear FS116P switches. These have 8 PoE ports plus 8 non-powered ports. Completely silent, no fans. Our testing was done with 5 phones per switch as our call centre is laid out in pods with 5 seats per pod and will not require more devices powered by PoE. I did not look into the issues of PoE class as there were no problems meeting our requirements with the equipment we had. Perhaps the Cisco switches are more particular about PoE class, they are much more complex devices. I like Cisco gear, they make good products but they tend to be over-featured and over-priced for many applications. We have had 5 of the Netgear switches in production for almost a year now, each powering 5 Aastra 480i phones without any issues whatsoever. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] chanspy does not pull the call back to asterisk after a reinvite
Thanks to both of you for your input. I'll be in touch off list Steve. -Franklin - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 30, 2008 8:00:48 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite On Jan 29, 2008 8:36 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Jan 29, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote: Franklin, Because ChanSpy() is a passive monitor, there is nothing about the implementation that would cause Asterisk to shunt the speech back to itself. Asterisk only does this in situations where it is out of the media path and needs to insinuate itself back into it for the purpose of generating media, such as on-hold music, IVR, etc. What you're wanting should, in my opinion, basically be submitted as a feature request. Perhaps the developers can add a flag to the ChanSpy() invocation repertoire to make this work. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Alex, he was not asking why, it is obvious he knows why. He was asking for a solution or idea on how to work around this issue. Are you using Sangoma cards? If so, I might have a very good answer for you, as well as another very possible different solution. Both would be outside of Asterisk so some kind of magic would have to happen to associate the call being spied on to the channel but that should not be that difficult if you even need it. Another solution is to track down the code referenced here http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a reinvite back to asterisk before starting the spy. Anyways, I am sure it can be done. The question is how much time is it worth to make it happen. Maybe we should meet for lunch this week. I can meet you in cow country or Philly if you want, your choice. I have to go to both this week anyways and would like to catch up with things since Astricon. Thanks, Steve Totaro I just confirmed that there is a solution that is perfect for this that has been developed with a web interface to select the call to monitor. A little added code and you can pretty easily look up who the agent handling the call is. Let's test it out on your call center. Again, it is not an Asterisk app and would have no impact on your operations if it does not work. Thanks, Steve Totaro ___ -- 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 -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] ___ -- 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] No audio one way
Hi, I have one problem, i´ve a trunk sip Asterisk--- Cisco 2600. Call inbound work very good, but call outbound don´t work. Call progress but no audio. Canreinvite=no , no Nat, No problem Codec. Any idea??? Thanks in advance, D ___ -- 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] Meetme voice quality problems
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is cut.
Each voice sequence is disturbed.
Does any one have similar issue and could give me some advice??
my extension.conf for meetme:
;switch = Realtime/macro-conference
exten = s,1,NoOp(-- Macro-conference for:${MARCO_EXTEN} start --)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,MeetMe(|cdIps)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
Thank for any help.
Kind Regards
Tomasz
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[asterisk-users] func_odbc - trouble
Hello, we are migrating an Asterisk install from Trixbox/Asterisk 1.2 to Elastix 0.92 with asterisk-1.4.17 on CentOS 5. I need to migrate an funtion that consults a remote sybase database, using ODBC and freetds. On the new server I am able to connect to the database using isql without problems. When I try to connect from asterisk logs show: pbx.c: Function ODBC_SQL not registered Indeed I can find this function in /var/lib/asterisk/modules As far as I understand asterisk needs to be recompiled to include the function. Recompiling asterisk-1.4.17 I get even into more trouble: make menuselect shows XXX 4. res_config_odbc XXX 13. res_odbc ODBC Resource Depends on: unixodbc(E), ltdl(E) unixODBC and unixODBC-devel are installed. Thanks for any hint to get my ODBC_SQL included and working! Enrique -- Dirk Enrique Seiffert - Lintec S.A. Ed. Torre del Reloj - Of. 401 Plaza de los Coches, Centro Cartagena - Colombia http://www.sipcolombia.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] Using two SIP-Domains with asterisk
30 jan 2008 kl. 15.44 skrev Bjoern Haje: Hi, is it possible to use asterisk to serve two SIP-domains with different users? It does work to define two domains with 'domain=' in sip.conf, but that allows all sip-users to register with both domains. I want to define users for a one domain only and not allow them to use the second. Lets say I have domains domainA.org and domainB.org and a sip-user user1. User1 should be able to register as [EMAIL PROTECTED] but not as [EMAIL PROTECTED] In addition I want to be able to define a second (different) user with the same username but another domain. How can I do that? Is it possible at all? Everything is possible if you change the code, but at this moment we only have one address space in chan_sip - regardless of domain. So you need unique user names for all users. Normally this is not a huge issue, as you need to separate user names and extensions, so that the user names are random strings or something that will never match anything useful except the user's name in Asterisk. This is something I planned for codename pineapple - to be able to separate Asterisk per domain. /O ___ -- 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] func_odbc - trouble
See below: Dirk Enrique Seiffert wrote: Hello, we are migrating an Asterisk install from Trixbox/Asterisk 1.2 to Elastix 0.92 with asterisk-1.4.17 on CentOS 5. I need to migrate an funtion that consults a remote sybase database, using ODBC and freetds. On the new server I am able to connect to the database using isql without problems. When I try to connect from asterisk logs show: pbx.c: Function ODBC_SQL not registered Indeed I can find this function in /var/lib/asterisk/modules As far as I understand asterisk needs to be recompiled to include the function. Recompiling asterisk-1.4.17 I get even into more trouble: make menuselect shows XXX 4. res_config_odbc XXX 13. res_odbc ODBC Resource Depends on: unixodbc(E), ltdl(E) unixODBC and unixODBC-devel are installed. You also need ltdl (whatever that is) ;) Thanks for any hint to get my ODBC_SQL included and working! Enrique ___ -- 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] sipsock_read: BAD! BAD! BAD!
On Jan 30, 2008 3:43 PM, Russell Bryant [EMAIL PROTECTED] wrote: There have been 380 fixes made to Asterisk 1.2 since 1.2.14, 61 of which were made to chan_sip. :) Did the one that gives the message Yikes, we should NEVER BE HERE! get swatted? :) ___ -- 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] Meetme voice quality problems
Tomasz Zieleniewski wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? Tomasz, Have you run zttest on the system? It verifies the accuracy of your timing source. Digium recommends an accuracy of at least 99.98%. If your accuracy is less than that it's probably the source of your problem. Luckily, it's a problem with multiple solutions. The following thread documents some kernel configuration changes that you can make to improve the quality of ztdummy as a timing source: Recommendations for kernel config http://lists.digium.com/pipermail/asterisk-users/2007-October/197778.html My preferred solution is to use an empty TDM400P as a timing source. It will cost you a little bit of money, but it's an easy way to reliably solve your problem. You'll find a few posts about it if you search the list, but this one has most of the information you'll need: Empty Wildcard TDM400P as a MeetMe timer. http://lists.digium.com/pipermail/asterisk-users/2007-March/182005.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- 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] Meetme voice quality problems
ztdummy can give you issues as a timing device. Any way you could try using a
Digium card just as a timing device to see if this helps?
- Original Message -
From: Tomasz Zieleniewski [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 30, 2008 11:23:57 AM (GMT-0500) America/New_York
Subject: [asterisk-users] Meetme voice quality problems
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is cut.
Each voice sequence is disturbed.
Does any one have similar issue and could give me some advice??
my extension.conf for meetme:
;switch = Realtime/macro-conference
exten = s,1,NoOp(-- Macro-conference for:${MARCO_EXTEN} start --)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,MeetMe(|cdIps)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
Thank for any help.
Kind Regards
Tomasz
--
Franklin Webb
Asst Project Manager
Inter Medi@ Marketing Solutions
610-701-9670
[EMAIL PROTECTED]
___
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Re: [asterisk-users] func_odbc - trouble
XXX 4. res_config_odbc XXX 13. res_odbc ODBC Resource Depends on: unixodbc(E), ltdl(E) unixODBC and unixODBC-devel are installed. You also need ltdl (whatever that is) ;) I guess this libtool-ltdl-1.5.22-6.1 ... which is installed. Thanks Enrique Thanks for any hint to get my ODBC_SQL included and working! Enrique ___ -- 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 -- Dirk Enrique Seiffert - Lintec S.A. Ed. Torre del Reloj - Of. 401 Plaza de los Coches, Centro Cartagena - Colombia http://www.lintecsa.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] Parking lot
Is there any way to have Asterisk call an extension in dial plan instead of original extension after timeout? Like extension A puts the caller in parking lot, he leaves the phone and forgets about it, instead of having that phone rings after timeout, have a group of phones rings. ___ -- 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] codec_g729a.so problem...
Sorry if this was repeated, but I think the list is acting up and not accepting some emails so I wanted to resend it just in case. On Jan 29, 2008 10:15 AM, arkda [EMAIL PROTECTED] wrote: Recently with Asterisk 1.4.17 I've been running into some stability issues. I started looking through my logs, and I found this: [Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module 'codec_g729a.so' was not compiled against a recent version of Asterisk and may cause instability. I'm using the newest version of codec_g729a.so from the Digium website (v33). I've tried using the 686, 586, and 386 versions (my platform is 32 bit). Is there a version that has been properly compiled for 1.4.17? I never had this problem with previous versions of Asterisk (1.4.15 and below), and I can't seem to find any information on this. ___ -- 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] func_odbc - trouble
Dirk Enrique Seiffert wrote: I guess this libtool-ltdl-1.5.22-6.1 ... which is installed. Thanks Enrique I believe you're looking for libtool-ltdl-dev(el) -- Jason Parker Digium ___ -- 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] Best Console phone?
On 1/27/08, Michelle Dupuis [EMAIL PROTECTED] wrote: The Aastra's also have a range of interested firmware bugs that support/development just can't seem to fix. Do a search for aastra hang/lockup and you will find what I mean. Have you ever achieved those hangup cases? Which firmware/model are you using? -- Octavio H. Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Mobile: (+55 155) 5514-087790 Mobile: (+55 155) 5541-351242 ___ -- 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] Unicall CRN 32769 - far disconnected cause=Switching equipment congestion [42]
Dears, After weeks trying to contact support of my telecom about 'Seize Ack' because that is not returned, was a lock for make calls on my E1s. Now I receive back de Ack and get ready to make calls, but the technical support reports to me that my attempts to call do not send any digits to the oder site (telecom station). 8 seconds after start 'Unicall event Dialing' the line is disconnected, like when you take up the line and hold without press any digits, after some seconds you got the congestion signal. Just for consideration I receive call without any problems, provided that performed the first setup. I have use http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 to do my configuration, sources are: http://www.moythreads.com/astunicall/files/astunicall-1.4.9-0.1.tar.gz zaptel-1.4.4-6 asterisk-1.4.9 libsupertone-0.0.2-1 spandsp-0.0.4-1 libunicall-0.0.3-1 libmfcr2-0.0.3-1 The only difference is I have use the sources to make a SRPM - RPM files on CentOS 5. Here is my config files: zaptel.conf loadzone= br defaultzone = br span=1,1,0,cas,hdb3 span=2,2,0,cas,hdb3 span=3,3,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 cas=32-46:1101 cas=48-62:1101 cas=63-77:1101 cas=79-93:1101 unicall.conf [channels] loglevel=255 language=pt_BR context=from-pstn usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both protocolclass=mfcr2 ;protocolvariant=br,20,4,x,max-seize-wait-ack=1 protocolvariant=br,20,4 protocolend=cpe group=1 callerid=asreceived channel=1-15 channel=17-31 channel=32-46 channel=48-62 channel=63-77 channel=79-93 protocolclass=mfcr2 Here is the LOGS when I try do make calls [Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Executing [ [EMAIL PROTECTED]:32] Dial(SIP/4805-0935d828, UniCall/g1|300|) in new stack [Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call called - 'g1' [Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call caller id - '4805' [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call control(1) [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Make call [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Creating a new call with CRN 32769 [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 0001 - [1/DIALING /Seize /Idle ] [Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Called g1 [Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Unicall/1 event Dialing [Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Exception on 15, channel 1 [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - [1/DIALING /Seize /Idle ] [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g on - [2/DIALING /Group I /DNIS ] [Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - 4 on [2/DIALING /Group I /DNIS ] [Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g off - [2/DIALING /Group I /DNIS ] [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - 4 off [2/DIALING /Group I /DNIS ] [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Far end disconnected(cause=Switching equipment congestion [42]) - state 0x40 [Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: Unicall/1 event Far end disconnected [Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: CRN 32769 - far disconnected cause=Switching equipment congestion [42] [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- Channel 0 got hangup [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: needcongestion [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- UniCall/1-1 is circuit-busy [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Channel gains [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Channel switching [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1 [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call control(7) [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Clearing fwd [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 1001 - [2/FAR DISC/Clear fwd B /Idle ] [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- Hungup 'UniCall/1-1' [Jan 30 16:41:26] VERBOSE[10717] logger.c: == Everyone is busy/congested at this time (1:0/1/0) If someone can help me I would be very grateful. Best Regards, -- Roger C. Beraldi Martins Fone: 55 41-8828-7068 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
[asterisk-users] conf meetme all exited and still active
Running asterisk 1.2.23 on a meetme two polycom SIP phones. When we both hung up the meetme was still active. Is there something special needed to destroy the static conference? Jerry ___ -- 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 1.4.18-rc3 Now Available
Asterisk 1.4.18-rc3 is now available. The important bug fixes that made it into this RC are a couple of crash fixes for ChanSpy/MixMonitor. A few other less severe bug fixes made it in, as well. This release candidate is published for anyone that is interested in helping to test it for a couple of days before it is officially released. To download the release candidate, use the following svn command: $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc3 If you would like it in tarball format, use the following commands: $ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc3 $ tar -czvf asterisk-1.4.18-rc3.tar.gz asterisk-1.4.18-rc3/ Thanks! ___ -- 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] Unicall CRN 32769 - far disconnected cause=Switching equipment congestion [42]
Well, that's simple, the telco is not getting any digits because YOU are not sending any digits! From the logs, I see you are dialing like this: Dial(UniCall/g1|300|) Where is the number you want to reach? I'd expect to see Dial(Unicall/g1/1234567890|300) To reach number 1234567890 - Moisés Silva On Jan 30, 2008 1:21 PM, Roger C. Beraldi Martins [EMAIL PROTECTED] wrote: Dears, After weeks trying to contact support of my telecom about 'Seize Ack' because that is not returned, was a lock for make calls on my E1s. Now I receive back de Ack and get ready to make calls, but the technical support reports to me that my attempts to call do not send any digits to the oder site (telecom station). 8 seconds after start 'Unicall event Dialing' the line is disconnected, like when you take up the line and hold without press any digits, after some seconds you got the congestion signal. Just for consideration I receive call without any problems, provided that performed the first setup. I have use http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 to do my configuration, sources are: http://www.moythreads.com/astunicall/files/astunicall-1.4.9-0.1.tar.gz zaptel-1.4.4-6 asterisk-1.4.9 libsupertone-0.0.2-1 spandsp-0.0.4-1 libunicall-0.0.3-1 libmfcr2-0.0.3-1 The only difference is I have use the sources to make a SRPM - RPM files on CentOS 5. Here is my config files: zaptel.conf loadzone= br defaultzone = br span=1,1,0,cas,hdb3 span=2,2,0,cas,hdb3 span=3,3,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 cas=32-46:1101 cas=48-62:1101 cas=63-77:1101 cas=79-93:1101 unicall.conf [channels] loglevel=255 language=pt_BR context=from-pstn usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both protocolclass=mfcr2 ;protocolvariant=br,20,4,x,max-seize-wait-ack=1 protocolvariant=br,20,4 protocolend=cpe group=1 callerid=asreceived channel=1-15 channel=17-31 channel=32-46 channel=48-62 channel=63-77 channel=79-93 protocolclass=mfcr2 Here is the LOGS when I try do make calls [Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Executing [EMAIL PROTECTED]:32] Dial(SIP/4805-0935d828, UniCall/g1|300|) in new stack [Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call called - 'g1' [Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call caller id - '4805' [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call control(1) [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Make call [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Creating a new call with CRN 32769 [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 0001 - [1/DIALING /Seize /Idle ] [Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Called g1 [Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Unicall/1 event Dialing [Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Exception on 15, channel 1 [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - [1/DIALING /Seize /Idle ] [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g on - [2/DIALING /Group I /DNIS ] [Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - 4 on [2/DIALING /Group I /DNIS ] [Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g off - [2/DIALING /Group I /DNIS ] [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - 4 off [2/DIALING /Group I /DNIS ] [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Far end disconnected(cause=Switching equipment congestion [42]) - state 0x40 [Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: Unicall/1 event Far end disconnected [Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: CRN 32769 - far disconnected cause=Switching equipment congestion [42] [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- Channel 0 got hangup [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: needcongestion [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- UniCall/1-1 is circuit-busy [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Channel gains [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Channel switching [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1 [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call control(7) [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Clearing fwd [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 1001 - [2/FAR DISC/Clear fwd B /Idle ] [Jan 30
[asterisk-users] OT - Looking for used 2 FXS port pci card
My apologies for the OT request, I think more people here may have what I am looking for than on the commercial list (which I did post to also). I'm looking for a used 2 FXS port pci card, don't care who makes it as long as it works. Oh and I have a limited budget...;) If you have anything you want to get rid of please email me off list. Thanks, Glenn ___ -- 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] calls get stuck in the asterisk box
At the end of the day SIP calles keep stuck in asterisk, is there any way to prevent this or debug this? The sip calls which get stuck all are calles on a krik IP600v3 dect gateway, I cant tell if they originate of the ip600v3, probably this are calls TO the IP600v3 10.0.0.71240 2c2cfcc47ca 05593/103700 0x0 (nothing) No Tx: BYE Done 10.0.0.71238 d4b2f570e90 00105/103150 0x0 (nothing) No Rx: BYE 10.0.0.71240 5d02b0d503e 06353/102998 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 4b303fed159 16797/93872 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 181151d9010 16819/93839 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 4abf61ec5ee 18318/92482 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 43a74c2f08d 19014/91859 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 672a3a624b5 19237/91616 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 4ede9bb258e 19332/91525 0x0 (nothing) No Tx: BYE Done 9 active SIP channels -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 the sip.conf for the phones on the IP600v3 all have this settings in sip.conf [239] type=friend username = 239 callerid=name 239 host = dynamic secret = 239 context = default qualify = yes login = 239 callgroup = 3 pickupgroup = 3 disallow = all allow = alaw call-limit = 6 setting of call-limit to 1 doesn't prevent the above mentioned problem. ___ -- 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] func_odbc - trouble [solved]
Dirk Enrique Seiffert wrote: I guess this libtool-ltdl-1.5.22-6.1 ... which is installed. Thanks Enrique I believe you're looking for libtool-ltdl-dev(el) Thansk a lot, - this made the difference!!! 2 days of unhappy hacking found an end!!! -- Jason Parker Digium ___ -- 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 -- Dirk Enrique Seiffert - Lintec S.A. Ed. Torre del Reloj - Of. 401 Plaza de los Coches, Centro Cartagena - Colombia http://www.lintecsa.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] G729 version to be downloaded for my machines
Hi List; The output of cat /proc/cpuinfo giving a [Intel (R) Pentium (R) D] so what is the g729 version I have to download to work with my machine? Any help? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Meetme voice quality problems
Franklin wrote: ztdummy can give you issues as a timing device. Yes and no. See below Any way you could try using a Digium card just as a timing device to see if this helps? Tomasz wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. Your kernel is new enough that you should be able to leverage hi-res timers (you might need to patch ztdummy), or at least a RTC set to 8192 ticks/sec. What does dmesg show after ztdummy is loaded? I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Do you have internal_timing=yes in asterisk.conf? This option allows Asterisk to time the RTP stream based on zaptel/ztdummy clock and not on the received RTP stream. In a MeetMe, where callers might mute themselves, the received RTP stream is all but useless for timing. Dan ___ -- 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] Default delay time for Attended call transfer
Greetings, I have an issue with the length of time that passes from when someone hits the transfer soft key on a Cisco 7940, after doing an attended transfer, and when the recipient’s connects with the transferred call. It appears to be around 6 seconds. Is there a .conf in Asterisk where this time can be reduced? Thank you for your help Don No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.16/1251 - Release Date: 1/30/2008 9:29 AM ___ -- 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] POE draw on Aastra 480i
At 07:09 AM 1/30/2008, you wrote: We have had 5 of the Netgear switches in production for almost a year now, each powering 5 Aastra 480i phones without any issues whatsoever. I have one of the 8 port 4 with POE Netgear boxes powering 3 480i-CT phones for 2 years or so with zero probalems. Ira ___ -- 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 gateway
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm. Sam _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas Sent: Wednesday, January 30, 2008 10:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk gateway Hello everybody Anyone, to know a gateway that works with nextel simm cards? I'm looking for them, in internet, but I did'n look. Best regards ___ -- 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 gateway
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote: Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm. And how do they compare to others? -- Tzafrir Cohen icq#16849755 jabber:[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] Best Console phone?
Actualy Aastra phone are limited to control only 50 BLF, snom360 can handle in our site about 110, and only seems a bit busy time to time. adrià vidal ___ -- 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 gateway
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote: Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm. err biz again ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] Unicall CRN 32769 - far disconnected cause=Switching equipment congestion [42]
Man I am a little embarrassed now...
Actually dial plans and PBX rules is where I have less knowledge of
everything that involves the asterisk, because of this I am using freePBX
and this was my problem. I make the setup for outbound trunk to UniCall
using the freePBX and in this case has a bug causing this behavior:
http://freepbx.org/trac/ticket/634
But anyway, this mistake was very clear ... I should have seen !
Thank you Moises, now everything is working !
Best Regards.
2008/1/30, Moises Silva [EMAIL PROTECTED]:
Well, that's simple, the telco is not getting any digits because YOU
are not sending any digits!
From the logs, I see you are dialing like this:
Dial(UniCall/g1|300|)
Where is the number you want to reach?
I'd expect to see
Dial(Unicall/g1/1234567890|300)
To reach number 1234567890
- Moisés Silva
On Jan 30, 2008 1:21 PM, Roger C. Beraldi Martins
[EMAIL PROTECTED] wrote:
Dears,
After weeks trying to contact support of my telecom about 'Seize Ack'
because that is not returned, was a lock for make calls on my E1s.
Now I receive back de Ack and get ready to make calls, but the technical
support reports to me that my attempts to call do not send any digits
to
the oder site (telecom station). 8 seconds after start 'Unicall event
Dialing' the line is disconnected, like when you take up the line and
hold
without press any digits, after some seconds you got the congestion
signal.
Just for consideration I receive call without any problems, provided
that
performed the first setup.
I have use http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 to do my
configuration, sources are:
http://www.moythreads.com/astunicall/files/astunicall-1.4.9-0.1.tar.gz
zaptel-1.4.4-6
asterisk-1.4.9
libsupertone-0.0.2-1
spandsp-0.0.4-1
libunicall-0.0.3-1
libmfcr2-0.0.3-1
The only difference is I have use the sources to make a SRPM - RPM
files on
CentOS 5.
Here is my config files:
zaptel.conf
loadzone= br
defaultzone = br
span=1,1,0,cas,hdb3
span=2,2,0,cas,hdb3
span=3,3,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
cas=32-46:1101
cas=48-62:1101
cas=63-77:1101
cas=79-93:1101
unicall.conf
[channels]
loglevel=255
language=pt_BR
context=from-pstn
usecallerid=yes
hidecallerid=no
immediate=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
faxdetect=both
protocolclass=mfcr2
;protocolvariant=br,20,4,x,max-seize-wait-ack=1
protocolvariant=br,20,4
protocolend=cpe
group=1
callerid=asreceived
channel=1-15
channel=17-31
channel=32-46
channel=48-62
channel=63-77
channel=79-93
protocolclass=mfcr2
Here is the LOGS when I try do make calls
[Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Executing
[EMAIL PROTECTED]:32] Dial(SIP/4805-0935d828, UniCall/g1|300|)
in
new stack
[Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call called -
'g1'
[Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call caller id -
'4805'
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call
control(1)
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Make
call
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1
Creating a
new call with CRN 32769
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1
0001 -
[1/DIALING /Seize /Idle ]
[Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Called g1
[Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Unicall/1 event Dialing
[Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Exception on 15, channel
1
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2
UniCall/1 -
[1/DIALING /Seize /Idle ]
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g
on -
[2/DIALING /Group I /DNIS ]
[Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2
UniCall/1 - 4
on [2/DIALING /Group I /DNIS ]
[Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g off
-
[2/DIALING /Group I /DNIS ]
[Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2
UniCall/1 - 4
off [2/DIALING /Group I /DNIS ]
[Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Far
end
disconnected(cause=Switching equipment congestion [42]) - state 0x40
[Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: Unicall/1 event Far end
disconnected
[Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: CRN 32769 - far
disconnected cause=Switching equipment congestion [42]
[Jan 30 16:41:26] VERBOSE[10717] logger.c: -- Channel 0 got hangup
[Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: needcongestion
[Jan 30 16:41:26] VERBOSE[10717] logger.c: -- UniCall/1-1 is
circuit-busy
[Jan 30
[asterisk-users] CallerID shows wrong values in manager interface
Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time. Manager interface output: CallerIDName: unknown State: Ringing Event: Newstate Privilege: call,all Uniqueid: 1201748091.843 Channel: SIP/103-098500d8 CallerID: 103 SIP.conf snippets: [101] type=friend callerid=(Devraj Mukherjee 101) username=101 secret=password context=default host=dynamic allow=alaw [EMAIL PROTECTED] [103] type=friend callerid=(System admin Den 103) username=103 secret=password context=default host=dynamic allow=all [EMAIL PROTECTED] Extension.conf looks like: ; Standard POTS line configuration to pickup calls exten = _s,1,Wait(2) exten = _s,2,Queue(wagga-office-phones,90) exten = _s,3,VoiceMail([EMAIL PROTECTED]) exten = _s,4,Hangup exten = 101,1,Wait(1) exten = 101,2,SetCIDNum(101) exten = 101,3,Dial(SIP/101,30,trw) exten = 101,4,Voicemail(s101) exten = 101,5,Hangup exten = 103,1,Wait(1) exten = 103,2,Dial(SIP/103,30,trw) exten = 103,3,Voicemail(s103) exten = 103,4,Hangup -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- 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 shows wrong values in manager interface
remove the brackets around (Devraj Mukherjee 101) Regards Kev Devraj Mukherjee wrote: Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time. Manager interface output: CallerIDName: unknown State: Ringing Event: Newstate Privilege: call,all Uniqueid: 1201748091.843 Channel: SIP/103-098500d8 CallerID: 103 SIP.conf snippets: [101] type=friend callerid=(Devraj Mukherjee 101) username=101 secret=password context=default host=dynamic allow=alaw [EMAIL PROTECTED] [103] type=friend callerid=(System admin Den 103) username=103 secret=password context=default host=dynamic allow=all [EMAIL PROTECTED] Extension.conf looks like: ; Standard POTS line configuration to pickup calls exten = _s,1,Wait(2) exten = _s,2,Queue(wagga-office-phones,90) exten = _s,3,VoiceMail([EMAIL PROTECTED]) exten = _s,4,Hangup exten = 101,1,Wait(1) exten = 101,2,SetCIDNum(101) exten = 101,3,Dial(SIP/101,30,trw) exten = 101,4,Voicemail(s101) exten = 101,5,Hangup exten = 103,1,Wait(1) exten = 103,2,Dial(SIP/103,30,trw) exten = 103,3,Voicemail(s103) exten = 103,4,Hangup -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- 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] How to get called number in featuremap
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got executed on
pressing a particular DTMF key sequence.
As I want to execute my own application on pressing that key which will use
called number.
testfeature = 3,peer,AGI,StoreNumber|CalledNumber
Here I want to use called number in place of CalledNumber tag. When I use
any variable *${DIALEDPEERNUMBER} *then it does not resolve the variable in
features.conf.
if i use following then it does not work.
testfeature = 3,peer,AGI,StoreNumber|*${DIALEDPEERNUMBER}
*StoreNumber is my own application that stores the number.
*
Any idea as how I can use CalledNumber in features.conf?
Please help.
Thanks in Advance
Regards
Prashant
___
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[asterisk-users] pulling my hair out over voicemail
Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- 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] pulling my hair out over voicemail
You might need Voicemailmain([EMAIL PROTECTED]) PaulH On Thu, 2008-01-31 at 00:30 -0500, John Von Essen wrote: Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- 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] pulling my hair out over voicemail
Can you get some verbose output from your console/logs? It may be more obvious once you see what Asterisk is attempting to do when this extension is dialed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Von Essen Sent: Wednesday, January 30, 2008 21:30 To: asterisk-users@lists.digium.com Subject: [asterisk-users] pulling my hair out over voicemail Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- 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] pulling my hair out over voicemail
John Von Essen wrote: Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- 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 I would suggest showing us the extensions configs for both phones :). ___ -- 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] chanspy does not pull the call back to asterisk after a reinvite
Franklin Webb wrote: Thanks to both of you for your input. I'll be in touch off list Steve. -Franklin - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 30, 2008 8:00:48 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite On Jan 29, 2008 8:36 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Jan 29, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote: Franklin, Because ChanSpy() is a passive monitor, there is nothing about the implementation that would cause Asterisk to shunt the speech back to itself. Asterisk only does this in situations where it is out of the media path and needs to insinuate itself back into it for the purpose of generating media, such as on-hold music, IVR, etc. What you're wanting should, in my opinion, basically be submitted as a feature request. Perhaps the developers can add a flag to the ChanSpy() invocation repertoire to make this work. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Alex, he was not asking why, it is obvious he knows why. He was asking for a solution or idea on how to work around this issue. Are you using Sangoma cards? If so, I might have a very good answer for you, as well as another very possible different solution. Both would be outside of Asterisk so some kind of magic would have to happen to associate the call being spied on to the channel but that should not be that difficult if you even need it. Another solution is to track down the code referenced here http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a reinvite back to asterisk before starting the spy. Anyways, I am sure it can be done. The question is how much time is it worth to make it happen. Maybe we should meet for lunch this week. I can meet you in cow country or Philly if you want, your choice. I have to go to both this week anyways and would like to catch up with things since Astricon. Thanks, Steve Totaro I just confirmed that there is a solution that is perfect for this that has been developed with a web interface to select the call to monitor. A little added code and you can pretty easily look up who the agent handling the call is. Let's test it out on your call center. Again, it is not an Asterisk app and would have no impact on your operations if it does not work. Thanks, Steve Totaro ___ -- 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 in sip.conf do canreinvite=no, and suddenly the audio is always available to asterisk. Anthony ___ -- 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] pulling my hair out over voicemail
Tried it, but no change. A few updates. Even though I dont hear anything, if I hit a keys on the phone and then hang up, message log says: [Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password I enabled logging of everything, and the below is the snippet for when my SIP/6001 phone dial extension 1000 for Voicemail: [Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Ringing' [Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081de7a8, ) in new stack [Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Wait' [Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081de7a8, 2) in new stack [Jan 30 21:26:37] DEBUG[7917] pbx.c: Launching 'VoiceMailMain' [Jan 30 21:26:37] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081de7a8, [EMAIL PROTECTED]) in new stack [Jan 30 21:26:37] DEBUG[7917] app_voicemail.c: Before ast_answer [Jan 30 21:26:37] DEBUG[7917] devicestate.c: Notification of state change to be queued on device/channel SIP/6001-081de7a8 [Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for peer 6001 [Jan 30 21:26:37] DEBUG[7890] devicestate.c: Changing state for SIP/6001 - state 5 (Unavailable) [Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for peer 6001 [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: SIP answering channel: SIP/6001-081de7a8 [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Setting framing from config on incoming call [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: -- Done with adding codecs to SDP [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 30 21:26:37] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:37] DEBUG[7917] rtp.c: Ooh, format changed from unknown to ulaw [Jan 30 21:26:37] DEBUG[7917] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 21:26:37] VERBOSE[7917] logger.c: -- SIP/6001-081de7a8 Playing 'vm-login' (language 'en') [Jan 30 21:26:37] DEBUG[7910] app_queue.c: Device 'SIP/6001' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Jan 30 21:26:37] DEBUG[7893] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 12349: Match Not Found [Jan 30 21:26:39] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw [Jan 30 21:26:50] DEBUG[7917] app_voicemail.c: Before find user for mailbox 8563682102 [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:50] DEBUG[7917] rtp.c: Difference is 82416, ms is 10322 [Jan 30 21:26:50] VERBOSE[7917] logger.c: -- SIP/6001-081de7a8 Playing 'vm-password' (language 'en') [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw [Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Auto destroying SIP dialog '[EMAIL PROTECTED]' [Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog [EMAIL PROTECTED] [Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: Sending dtmf: 53 (5), at 76.161.192.192 [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin '5' received on SIP/6001-081de7a8 [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin ignored '5' on SIP/6001-081de7a8 [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: Sending dtmf: 53 (5), at 76.161.192.192 [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF end '5' received on SIP/6001-081de7a8, duration 120 ms [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF end passthrough '5' on SIP/6001-081de7a8 [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:57] DEBUG[7893] chan_sip.c: Setting SIP_ALREADYGONE on dialog [EMAIL PROTECTED] [Jan 30 21:26:57] DEBUG[7893] chan_sip.c: Received bye, issuing owner hangup [Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password [Jan 30 21:26:57] DEBUG[7917] app_voicemail.c: After vm_authenticate [Jan 30
Re: [asterisk-users] pulling my hair out over voicemail
How about your sip.conf for your extensions? Example: [6001] host=dynamic type=friend disallow=all allow=ulaw I usually don't see this (I'm more production and haven't done heavy debug for a long time): [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw Since it's within the same second, I'm not sure which is actually being set. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Von Essen Sent: Wednesday, January 30, 2008 22:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pulling my hair out over voicemail Tried it, but no change. A few updates. Even though I dont hear anything, if I hit a keys on the phone and then hang up, message log says: [Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password I enabled logging of everything, and the below is the snippet for when my SIP/6001 phone dial extension 1000 for Voicemail: [Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Ringing' [Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081de7a8, ) in new stack [Jan 30 21:26:35] DEBUG[7917] pbx.c: Launching 'Wait' [Jan 30 21:26:35] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081de7a8, 2) in new stack [Jan 30 21:26:37] DEBUG[7917] pbx.c: Launching 'VoiceMailMain' [Jan 30 21:26:37] VERBOSE[7917] logger.c: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081de7a8, [EMAIL PROTECTED]) in new stack [Jan 30 21:26:37] DEBUG[7917] app_voicemail.c: Before ast_answer [Jan 30 21:26:37] DEBUG[7917] devicestate.c: Notification of state change to be queued on device/channel SIP/6001-081de7a8 [Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for peer 6001 [Jan 30 21:26:37] DEBUG[7890] devicestate.c: Changing state for SIP/6001 - state 5 (Unavailable) [Jan 30 21:26:37] DEBUG[7890] chan_sip.c: Checking device state for peer 6001 [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: SIP answering channel: SIP/6001-081de7a8 [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Setting framing from config on incoming call [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: -- Done with adding codecs to SDP [Jan 30 21:26:37] DEBUG[7917] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 30 21:26:37] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:37] DEBUG[7917] rtp.c: Ooh, format changed from unknown to ulaw [Jan 30 21:26:37] DEBUG[7917] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jan 30 21:26:37] VERBOSE[7917] logger.c: -- SIP/6001-081de7a8 Playing 'vm-login' (language 'en') [Jan 30 21:26:37] DEBUG[7910] app_queue.c: Device 'SIP/6001' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. [Jan 30 21:26:37] DEBUG[7893] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 12349: Match Not Found [Jan 30 21:26:39] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw [Jan 30 21:26:50] DEBUG[7917] app_voicemail.c: Before find user for mailbox 8563682102 [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format gsm [Jan 30 21:26:50] DEBUG[7917] rtp.c: Difference is 82416, ms is 10322 [Jan 30 21:26:50] VERBOSE[7917] logger.c: -- SIP/6001-081de7a8 Playing 'vm-password' (language 'en') [Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8 to write format ulaw [Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Auto destroying SIP dialog '[EMAIL PROTECTED]' [Jan 30 21:26:53] DEBUG[7893] chan_sip.c: Destroying SIP dialog [EMAIL PROTECTED] [Jan 30 21:26:53] VERBOSE[7893] logger.c: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: Sending dtmf: 53 (5), at 76.161.192.192 [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin '5' received on SIP/6001-081de7a8 [Jan 30 21:26:55] DTMF[7917] channel.c: DTMF begin ignored '5' on SIP/6001-081de7a8 [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4) [Jan 30 21:26:55] DEBUG[7917] rtp.c: - RTP 2833 Event: 0005 (len = 4)
[asterisk-users] Server Compatibility List for Asterisk
Digium has a compatibility list of servers, however, it has not been updated since 2006. One of the servers on the list has since been taken out of production by Dell. Here are the remaining servers on the list: HP Proliant DL360IBM x206IBM x346 Does anyone has a most recent list and I will be adding the digium cards for T1 the 220 series with echo cancellation? ___ -- 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] How to get called number in featuremap
You need $dnis.
On Jan 30, 2008, at 11:08 PM, Prashant Sharma
[EMAIL PROTECTED] wrote:
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got
executed on pressing a particular DTMF key sequence.
As I want to execute my own application on pressing that key which
will use called number.
testfeature = 3,peer,AGI,StoreNumber|CalledNumber
Here I want to use called number in place of CalledNumber tag.
When I use any variable ${DIALEDPEERNUMBER} then it does not resolve
the variable in features.conf.
if i use following then it does not work.
testfeature = 3,peer,AGI,StoreNumber|${DIALEDPEERNUMBER}
*StoreNumber is my own application that stores the number.
Any idea as how I can use CalledNumber in features.conf?
Please help.
Thanks in Advance
Regards
Prashant
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