Re: [asterisk-users] ztdummy, zttest
Hello, On Nov/11/2007, Tzafrir Cohen wrote: On Sun, Nov 11, 2007 at 08:51:40PM +0100, Carles Pina i Estany wrote: I also tried using bristuff 0.3y, 0.3s, etc. (is it 0.3 bristuff when Asterisk is 1.2.X?). Always without any result :-( Latest bristuff for 1.2 is y-k . See http://bristuff.org/ . However the bristuff Zaptel patch has no effect on ztdummy or Zaptel timing. I also tried y-k (last week) with no results in that machine. That computer is a Dell PowerEdge 860. I just checked now in other machine (standard Pentium 4) and just loading ztdummy and running zttest is working. In other machine, running ztdummy and zttest too. I don't know why this PowerEdge 860 is not working in that way :-( -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --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] Wanted: tutorial on troubleshooting SIP issues
On Nov 9, 2007 1:11 AM, Philip Prindeville [EMAIL PROTECTED] wrote: For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? It's true that using Ethereal (is that what Wireshark is nowadays?) will show you a lot and teach you a lot, but, I totally agree with you, it would be a popular site indeed that posted a commented cleaned up packet dump with examples of sequences for one or more popular phones and full explanations. Such a project would help a lot of people become more aware of SIP and bring attention to the person or company who does it. ___ --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] Wanted: tutorial on troubleshooting SIP issues
On Mon, Nov 12, 2007 at 10:20:16AM +0100, randulo wrote: It's true that using Ethereal (is that what Wireshark is nowadays?) Ethereal was the original name of Wireshark. -- 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] Wanted: tutorial on troubleshooting SIP issues
I thought Wireshark was the cute Mac OS X name. On Nov 12, 2007 10:32 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Nov 12, 2007 at 10:20:16AM +0100, randulo wrote: It's true that using Ethereal (is that what Wireshark is nowadays?) Ethereal was the original name of Wireshark. -- 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 ___ --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] Wanted: tutorial on troubleshooting SIP issues
And by the way, other than a dump, I'm surprised no one suggested studying the source code. However, I don't think either would be as useful as a good paper about these SIP transactions, especially an asterisk-centric one that could even mention differences in newer versions of asterisk. ___ --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] Record() : How to get filename created with %d?
In article [EMAIL PROTECTED],
Vincent [EMAIL PROTECTED] wrote:
Hello
About Record(), ATFT 2nd Edition says that if the filename
contains %d, these characters will be replaced with a number
incremented by one each time the file is recorded.
Problem is, the documentation doesn't explain how to refer to this
filename later in the dialplan :-/
I'm a little surprised at the variety of band-aid suggestions that have
been posted. All you need to do is refer to show application record,
and you uwill see that the generated filename is available by using
${RECORDED_FILE}
---
-= Info about application 'Record' =-
[Synopsis]
Record to a file
[Description]
Record(filename.format|silence[|maxduration][|options])
Records from the channel into a given filename. If the file exists it will
be overwritten.
- 'format' is the format of the file type to be recorded (wav, gsm, etc).
- 'silence' is the number of seconds of silence to allow before returning.
- 'maxduration' is the maximum recording duration in seconds. If missing
or 0 there is no maximum.
- 'options' may contain any of the following letters:
'a' : append to existing recording rather than replacing
'n' : do not answer, but record anyway if line not yet answered
'q' : quiet (do not play a beep tone)
's' : skip recording if the line is not yet answered
't' : use alternate '*' terminator key instead of default '#'
If filename contains '%d', these characters will be replaced with a number
incremented by one each time the file is recorded. A channel variable
named RECORDED_FILE will also be set, which contains the final filemname.
Use 'show file formats' to see the available formats on your system
User can press '#' to terminate the recording and continue to the next priority.
If the user should hangup during a recording, all data will be lost and the
application will teminate.
---
Cheers
Tony
--
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
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[asterisk-users] Grandstream GXP2020 + Asterisk 1.4.11
Hi, I`m using several GXP2020 phones with newest Firmware 1.1.4.18. Asterisk Version: 1.4.11. It happens several times that users complain that the caller cannot hear the transmitted voice from the phones Also now it happens quite often that callers on hold beeing dropped. Environment: ISDN with chan_misdn and SIP internal calls. No NAT no DNS name (only IPS configured). I configured in sip.conf and on the phone now that alaw is preferred. As I saw in the FMW Bug list that GSM is not a good option Also I set the canreinvite=no as it is also configured in a Grandstream manual. I use on every phone the 1 as local port and in the rtp.conf I allowed a range from 1 - 5. As far my SIP knowledge is up to date the local port has not to differ from phone to phone or I´m wrong? Any idea or useres which had the same problems and fixed it? My sip.conf: [test1] type=friend context=outgoing username=test1 secret=987454 qualify=yes host=dynamic nat=yes canreinvite=no disallow=all allow=alaw allow=ulaw callerid=Test 0 insecure=very Kind Regards, Erik ___ --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] detect asterisk pbx via sip
Use tcpdump to investigate that Giedrius Augys wrote: Hello, My situation is that , I can't make calls with asterisk, but with x-lite works fine. Asterisk shows , that successfully registers with another SIP server, asterisk sends invite, gets trying, and after 30 secs asterisk gets 408 Request timeout. And as I said , with x-lite no problems. I heard that for comercial purposes, this SIP server detects asterisk , and ignores him. Or maybe it check is it server or device? Maybe somebody can give me some advices... 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 ___ --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] 'h' extension on call-out
Hello! I would like to store ISDNCAUSE on automatic call-out campaign (possibly gives more detail on failed call). How is it possible? I have tried 'failed' and 'h' extension. No luck. Extension 'failed' does not know anything about ISDNCAUSE and 'h' extension is not called at all. Any idea? I am using Asterisk 1.2.14 on FC4 if it counts. Cheers, a ___ --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] Video Call
On Nov 7, 2007 11:43 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 7 Nov 2007, Marek B wrote: On Nov 3, 2007 9:03 PM, Bert Haverkamp [EMAIL PROTECTED] wrote: This is generally not possible. The 3G phones (GPRS will be a strech wrt bandwidth) that do video telephony, do not support any SIP. So the (...) Not true - Nokia N95, 3G phone with video telephony, SIP support included. Makes no difference though - I haven't heard about any possibility to use builtin video connectivity on top of SIP. I'd love to be able to do this on my Nokia E90 too... Maybe one day! The voice SIP interface actually seems to work quite well over Wi-Fi though. It seems a lot more reliable than my UTS F1000G toy phone (And to be able to use SIP via it's 3G interface, but I'm not sure if that's possible - again, maybe one day!) SIP over UMTS using N95: tried and it works... in general... ;) Packet timings over UMTS networks (at least in Poland) are not acceptable. I was getting around 40-60sec delays in transfering voice up direction (from me). In the opposite direction it was ok most of the time. Regards -- Marek ___ --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] Internal CallerID problem
Hey Guys, I have something that just started happening. When my users call each other on their 5 digit extensions their CallerID is showing as [EMAIL PROTECTED] (X would be their Ext. and 10.25.2.50 is my server) Calls in an out to the outside world are fine. I have scoured my configuration and can't find what would be causing it. I have checked the sip.cfg in the polycom's and URI dialing is disabled. What am I missing? Trixbox 2.2.3 Asterisk 1.2.18 Polycom IP550's with 2.2 software Regards Mark ___ --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] 'h' extension on call-out
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is not called
at all. Any idea?
I am using Asterisk 1.2.14 on FC4 if it counts.
I think you will find the 'h' extension is only called at the end of
a successful call.
If the call is unsuccessful, the Dial command will return, and you can
then check ${ISDNCAUSE} on the next line of your dialplan.
Cheers
Tony
--
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] sip_chan - how to use value of the SIP 'To:' header field for extension logic
Hi, I have the following situation. I have one account created in my VoIP provider. Asterisk registers this account with the usage of 'register = ' command in the sip.conf file. I have a number of aliases assigned to my user which correspond to different public/PSTN numbers through which I am accessible. When there is an incoming call from my sip provider 'some_extension' which corresponds to my registered user 'rings'. this is because of such registration: register = user:[EMAIL PROTECTED]/some_extension. How can I now evaluate the value of the To header and perform further routing logic. What will happen if I don't specify the /extension value in the register command? Will it in such situation analyze the 'To' header to find matching extension?? I kindly ask for your help. Cheers Tomasz ___ --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] 'Traditional' Faxing
Greg Cockburn wrote: Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there maybe someway to bridge an Analogue FXS port to a channel on the E1? Basically I want to mimic what we have now. 1. Any person can send a fax using the fax machine, and the PBX picks the next free channel on the E1. 2. A fax call can come over any channel on the E1, and the dialed number is matched and sent to the analogue FXS port of the PBX to be received by the fax machine. Is there anyway I can do this in Asterisk that will work seamlessly? snip From what I've heard, I think your best bet is to buy a multi-port T1/E1 card for asterisk, put your E1 in one port and a channel bank in the other port, then plug your fax extension into an FXS port on the channel bank. Since both legs of the call pass through the same E1 interface card asterisk can bridge the call on the card itself and the timing issue should become moot. I have not done this nor have any hands-on experience to share, but I have done some research into this in the past. This is also the method Fonality recommends for customers of their asterisk based system: http://help.fonality.com/index.php/Fax_Machine_and_Modem_Support_in_PBXtra -Dave ___ --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] 'Traditional' Faxing
Dave Fullerton wrote: snip From what I've heard, I think your best bet is to buy a multi-port T1/E1 card for asterisk, put your E1 in one port and a channel bank in the other port, then plug your fax extension into an FXS port on the This is what we do for our fax machines along with using iaxmodem and HylaFAX+ It just works(tm). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Wanted: tutorial on troubleshooting SIP issues
On 12/11/2007, randulo [EMAIL PROTECTED] wrote: I thought Wireshark was the cute Mac OS X name. The author changed the name of the codebase last year due to employment changes: http://en.wikipedia.org/wiki/Wireshark#History Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] 'h' extension on call-out
Artifex Maximus wrote: Hello! I would like to store ISDNCAUSE on automatic call-out campaign (possibly gives more detail on failed call). How is it possible? I have tried 'failed' and 'h' extension. No luck. Extension 'failed' does not know anything about ISDNCAUSE and 'h' extension is not called at all. Any idea? I am using Asterisk 1.2.14 on FC4 if it counts. There is no such dialplan variable. Maybe you are looking for the HANGUPCAUSE variable? ___ --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] sip_chan - how to use value of the SIP 'To:' header field for extension logic
In article [EMAIL PROTECTED],
Tomasz Zieleniewski [EMAIL PROTECTED] wrote:
I have the following situation.
I have one account created in my VoIP provider.
Asterisk registers this account with the usage of
'register = ' command in the sip.conf file.
I have a number of aliases assigned to my user which
correspond to different public/PSTN numbers through which I am
accessible. When there is an incoming call from my sip provider
'some_extension' which corresponds to my registered user 'rings'.
this is because of such registration:
register = user:[EMAIL PROTECTED]/some_extension.
How can I now evaluate the value of the To header and perform
further routing logic.
What will happen if I don't specify the /extension value in the
register command?
I think it sill use the 's' extension in your incoming context.
Will it in such situation analyze the 'To' header to find matching extension??
Asterisk never uses the To header itself, expect to set the variable
SIP_HEADER(TO).
You can extract the number from the To header like this:
exten = some_extension,1,Set(DEST=${CUT(CUT(SIP_HEADER(TO),:,2),@,1)})
And then use ${DEST} in some way to route the call.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Wanted: tutorial on troubleshooting SIP issues
Brilliant program, whatever it's called this week. http://www.wireshark.org/faq.html#q1.2 http://en.wikipedia.org/wiki/Wireshark#History ___ --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] ztdummy, zttest
Debian as well as everyone else 2.6.18-5 Zaptel is branch/1.4 latest. The issue is not with Zaptel though...IMHO. If you look at /proc/driver/rtc, I find: rtc_time : 14:34:27 rtc_date : 2007-11-12 rtc_epoch : 1900 alarm: 16:30:31 DST_enable : no BCD: yes 24hr: yes square_wave : no alarm_IRQ : no update_IRQ : no periodic_IRQ : no periodic_freq : 1024 batt_status : okay If you notice, there is no periodic_IRQ. That is the issue. To me (and the kernels reported here) it is a Debian kernel problem. If I turn off ACPI, Zaptel works, but the box performance is awful because other things, which depend on ACPI do not have interrupts. I have read some places that if you have the Hard Drive Suppend in the BIOS enabled, you will get this situation. However, I would check the /proc/driver/rtc to see that you have periodic_IRQ. On Fri, Nov 09, 2007 at 04:59:37PM -0600, Tony Plack wrote: The thing is that this works, but The performance of the box becomes really bad. It seems that the problem, at least in my case is that the HPET timer from the cpu does not get an IRQ for the RTC. Does anyone else have a solution for this issue where the RTC does not get an interrupt when HPET is turned on? What kerenl version? What version of Zaptel? ___ --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] Polycom Speakerphone
Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric ___ --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 sound from playback and voicemail
Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] exten = 99,1,ANSWER() exten = 99,2,PLAYBACK(tt-monkeys) exten = 99,3,HANGUP() The phone has access to this context, and the file exists, all codecs are allowed. I have tried to load either chan_alsa.so or chan_oss.so but it doesn't change anything. Does anyone have an idea what could be wrong? This is not the first Asterisk system that I set up, but I never had a problem like this. Asterisk is version 1.4.13 Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] Wanted: tutorial on troubleshooting SIP issues
For general SIP understanding, there's also Sip Scenario from IPtel ( http://www.iptel.org/~sipsc/ ). It will generate sort of human-readable web stuff from captures, allowing you to click on the graphical portions of the call and see the actual SIP packets that correspond to that. N. randulo wrote: Brilliant program, whatever it's called this week. http://www.wireshark.org/faq.html#q1.2 http://en.wikipedia.org/wiki/Wireshark#History ___ --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] Polycom Speakerphone
On Monday November 12 2007 9:38 am, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric I have found the polcom speaker phone to be very good on the 320's, 330's, and the 501's. Clear clean voice even in relatively noisy areas. JohnM ___ --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] Polycom Speakerphone
Eric Jacksch wrote: For those of you using the polycom desk phones, how do you find the built-in speakerphone? Excellent! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Polycom Speakerphone
On Mon, 12 Nov 2007 09:38:57 -0500, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric Actually, IMHO, the Polycom speakerphone is the standard by which all other should be judged. I have a number of 500, 600 and 430 models in service and they're all very good. Even the little CS100 USB speakerphone device has been excellent. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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] 'h' extension on call-out
On Nov 12, 2007 3:22 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Artifex Maximus wrote: Hello! I would like to store ISDNCAUSE on automatic call-out campaign (possibly gives more detail on failed call). How is it possible? I have tried 'failed' and 'h' extension. No luck. Extension 'failed' does not know anything about ISDNCAUSE and 'h' extension is not called at all. Any idea? I am using Asterisk 1.2.14 on FC4 if it counts. There is no such dialplan variable. Maybe you are looking for the HANGUPCAUSE variable? Sorry, you are right. I mean HANGUPCAUSE... bye, a ___ --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] 'h' extension on call-out
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is not called
at all. Any idea?
I am using Asterisk 1.2.14 on FC4 if it counts.
I think you will find the 'h' extension is only called at the end of
a successful call.
Extension 'h' is executed on every hangup event from both side.
If the call is unsuccessful, the Dial command will return, and you can
then check ${ISDNCAUSE} on the next line of your dialplan.
There is no dial command because I only wrote a call file to
/var/spool/asterisk/outgoing and actually dial has been made by
pbx_spool module. There is a 'failed' extension but no HANGUPCAUSE
there.
Bye,
a
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Re: [asterisk-users] Grandstream GXP2020 + Asterisk 1.4.11
I`m using several GXP2020 phones with newest Firmware 1.1.4.18. I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22 and have eliminated that. Asterisk Version: 1.4.11. Me too. Still testing 1.4.13 on a non-production system. I use on every phone the 1 as local port and in the rtp.conf From my knowledge of IP I don't think this is a problem since the address/port would be unique. However the example config I originally had from Grandstream indicated that each phone should use a different port and recommended to use the random port option on the phones. I have since assigned the port number on each phone to 1 plus the extension number. This was done to create a unique port number and to help with troubleshooting when using Wireshark or tcpdump. I set this in the config file for each phone. 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] 'h' extension on call-out
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is not called
at all. Any idea?
I am using Asterisk 1.2.14 on FC4 if it counts.
I think you will find the 'h' extension is only called at the end of
a successful call.
Extension 'h' is executed on every hangup event from both side.
But your experience as described directly contradicts that assertion.
My statement that 'h' is only called at the end of a successful call was
not accurate. It depends how the dialplan is written. The actual condition
is this:
When a channel is hung up, IF and ONLY IF that channel is executing in
the dial plan, the 'h' extension is called in whatever context that
channel is in currently.
If the call is unsuccessful, the Dial command will return, and you can
then check ${ISDNCAUSE} on the next line of your dialplan.
There is no dial command because I only wrote a call file to
/var/spool/asterisk/outgoing and actually dial has been made by
pbx_spool module. There is a 'failed' extension but no HANGUPCAUSE
there.
That is why you are not executing the 'h' extension. You are placing
the call directly to a channel. That channel does not start executing
the dial plan until it is answered. In that respect, my original
statement was correct. Because it only starts the dialplan when the
call is successfully answered, there is no context in which to execute
an 'h' extension on a failed call.
You might find it useful to use a Local channel. Instead of doing this:
Channel: Zap/g1/123456789
Context: mycontext
Extension: s
Priority: 1
Try doing this:
Channel: Local/[EMAIL PROTECTED]
Context: mycontext
Extension: s
Priority: 1
(replace mycontext and s with whatever you are using currently)
And in extensions.conf:
[outgoing]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten =
_X.,n,NoOp(${EXTEN}:DIALSTATUS=${DIALSTATUS};HANGUPCAUSE=${HANGUPCAUSE})
exten = h,n,NoOp(${EXTEN}:DIALSTATUS=${DIALSTATUS};HANGUPCAUSE=${HANGUPCAUSE})
In that case you should find the 'h' in [outgoing] gets executed on both
successful
and failed calls.
(Replace Zap/g1 with whatever channel technology you are using)
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] No sound from playback and voicemail
On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote: Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. I once had this exact problem on a new installation of Asterisk and it was because one of the cards was not properly seated in the motherboard. Once I pushed the card all the way in I could hear voicemail and the IVR. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Polycom Speakerphone
Michael Graves wrote: On Mon, 12 Nov 2007 09:38:57 -0500, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? I am using the Polycom Communicator C100S on Ubuntu Linux. And despite most of the echo cancellation and noise reduction technology being only available in their Windows XP driver, the sound quality is excellent. I sometimes need to plug in a headset, but the desktop mic(s) work brilliantly. HTH Alan ___ --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] No sound from playback and voicemail
Stefan Guenther wrote: Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] exten = 99,1,ANSWER() exten = 99,2,PLAYBACK(tt-monkeys) exten = 99,3,HANGUP() The phone has access to this context, and the file exists, all codecs are allowed. I have tried to load either chan_alsa.so or chan_oss.so but it doesn't change anything. Does anyone have an idea what could be wrong? This is not the first Asterisk system that I set up, but I never had a problem like this. Asterisk is version 1.4.13 Do you have zaptel loaded? Could this be related? http://lists.digium.com/pipermail/asterisk-users/2007-November/199784.html If you do - you can try downgrading zaptel version. Regards, Atis ___ --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] 'Traditional' Faxing
On Monday 12 November 2007 07:54:42 Dave Fullerton wrote: From what I've heard, I think your best bet is to buy a multi-port T1/E1 card for asterisk, put your E1 in one port and a channel bank in the other port, then plug your fax extension into an FXS port on the channel bank. Since both legs of the call pass through the same E1 interface card asterisk can bridge the call on the card itself and the timing issue should become moot. I have not done this nor have any hands-on experience to share, but I have done some research into this in the past. This is also the method Fonality recommends for customers of their asterisk based system: That is exactly what I do in my systems. I did have success with a TDM400--T100P but that was a long time ago and I was unable to repeat it recently. I didn't spend a whole lot of time on it, though. -A. ___ --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] No sound from playback and voicemail
This will also happen if there is a zap card installed and unconfigured in zaptel.conf zapata.conf. Forrest Beck [EMAIL PROTECTED] www.shift8.biz dCAP On Nov 12, 2007, at 9:46 AM, Stefan Guenther wrote: Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] exten = 99,1,ANSWER() exten = 99,2,PLAYBACK(tt-monkeys) exten = 99,3,HANGUP() The phone has access to this context, and the file exists, all codecs are allowed. I have tried to load either chan_alsa.so or chan_oss.so but it doesn't change anything. Does anyone have an idea what could be wrong? This is not the first Asterisk system that I set up, but I never had a problem like this. Asterisk is version 1.4.13 Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] No sound from playback and voicemail
On Mon, Nov 12, 2007 at 07:14:31PM +0200, Atis Lezdins wrote: Stefan Guenther wrote: Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] exten = 99,1,ANSWER() exten = 99,2,PLAYBACK(tt-monkeys) exten = 99,3,HANGUP() The phone has access to this context, and the file exists, all codecs are allowed. I have tried to load either chan_alsa.so or chan_oss.so but it doesn't change anything. Does anyone have an idea what could be wrong? This is not the first Asterisk system that I set up, but I never had a problem like this. Asterisk is version 1.4.13 Do you have zaptel loaded? Could this be related? http://lists.digium.com/pipermail/asterisk-users/2007-November/199784.html If you do - you can try downgrading zaptel version. Downgrading? just properly configure the card. If you have an ISDN card, it should sync Asterisk. As a quick check for zaptel timing issue, please run zttest to see that you have a working zaptel timing source. -- 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] No sound from playback and voicemail (Carlos Chavez)
On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote: Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. I once had this exact problem on a new installation of Asterisk and it was because one of the cards was not properly seated in the motherboard. Once I pushed the card all the way in I could hear voicemail and the IVR. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 I made my first attempts with an onboard Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01). Then I used an old Multimedia audio controller: Ensoniq 5880 AudioPCI. Here is a list of the kernel modules that are loaded, maybe a relevant one is missing: snd_dummy 10496 0 snd_opl3_lib9984 0 snd_hwdep 8068 1 snd_opl3_lib snd_mpu401_uart 7680 0 snd_hda_intel 252576 0 snd_seq_dummy 3460 0 snd_seq_oss30720 0 snd_seq_midi8064 0 snd_ens137123712 0 gameport 12424 1 snd_ens1371 snd_ac97_codec 96672 1 snd_ens1371 snd_rawmidi22528 3 snd_mpu401_uart,snd_seq_midi,snd_ens1371 ac97_bus2304 1 snd_ac97_codec snd_pcm_oss41344 0 snd_mixer_oss 15488 1 snd_pcm_oss snd_pcm74504 5 snd_dummy,snd_hda_intel,snd_ens1371,snd_ac97_codec,snd_pcm_oss snd_seq_midi_event 7424 2 snd_seq_oss,snd_seq_midi snd_seq48368 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 20484 3 snd_opl3_lib,snd_pcm,snd_seq snd_seq_device 7820 6 snd_opl3_lib,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd48772 15 snd_dummy,snd_opl3_lib,snd_hwdep,snd_mpu401_uart,snd_hda_intel,snd_seq_oss,snd_ens1371,snd_ac97_codec,snd_rawmidi,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq,snd_timer,snd_seq_device soundcore 6880 1 snd snd_page_alloc 8584 2 snd_hda_intel,snd_pcm Regards, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] Polycom Speakerphone
At 08:38 11/12/2007, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric Excellent speakerphone. Extremely cumbersome to configure. ___ --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] No sound from playback and voicemail (Atis Lezdins)
Hello, I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] exten = 99,1,ANSWER() exten = 99,2,PLAYBACK(tt-monkeys) exten = 99,3,HANGUP() The phone has access to this context, and the file exists, all codecs are allowed. I have tried to load either chan_alsa.so or chan_oss.so but it doesn't change anything. Asterisk is version 1.4.13 Do you have zaptel loaded? Could this be related? http://lists.digium.com/pipermail/asterisk-users/2007-November/199784.html If you do - you can try downgrading zaptel version. I have started asterisk with and without loading zaptel and ztdummy, still no playback or voicemail. Regards, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] Polycom Speakerphone
On Monday November 12 2007 1:50 pm, Doug wrote: At 08:38 11/12/2007, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric Excellent speakerphone. Extremely cumbersome to configure. I do not understand how you can say that the Polycoms are Extremely cumbersome to configure. I find them rather nice. Once you have one working config it is very easy to copy that config over to the mac address files for the other phones that you have and only change the per phone bits. Set up a site wide sip.cfg and then use phone-(macaddress).cfg files for the individual settings for each phone. real nice when you have more than a couple phones to configure. It is not my intention to start any war here just giving my 2 cents worth. JohnM ___ --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] Polycom Speakerphone
Doug wrote: At 08:38 11/12/2007, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric Excellent speakerphone. Extremely cumbersome to configure. I agree about the speakerphone, and disagree with the claim about configuration. The XML is extremely to generate through scripts, and once the framework is built, I find it to be far simpler to manage the deployment than other IP phones. Of course, YMMV. ___ --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] VoiceMail hangup
Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this 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] VoiceMail hangup
Hi additional information if I am going to wait at least 3 seconds after the voicemail starts to give me the instruction I am able to listen my messages. But why I need to wait? On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote: Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this 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] Polycom Speakerphone
At 13:05 11/12/2007, John Millican, wrote: Excellent speakerphone. Extremely cumbersome to configure. I do not understand how you can say that the Polycoms are Extremely cumbersome to configure. I find them rather nice. Once you have one working config it is very easy to copy that config over to the mac address files for the other phones that you have and only change the per phone bits. Set up a site wide sip.cfg and then use phone-(macaddress).cfg files for the individual settings for each phone. real nice when you have more than a couple phones to configure. It is not my intention to start any war here just giving my 2 cents worth. JohnM At 13:13 11/12/2007, David Gomillion wrote: Excellent speakerphone. Extremely cumbersome to configure. I agree about the speakerphone, and disagree with the claim about configuration. The XML is extremely to generate through scripts, and once the framework is built, I find it to be far simpler to manage the deployment than other IP phones. Of course, YMMV. I agree that once the .cfg files are working, duplicating them to use on other phones if fairly straightforward. That having been said, getting the .cfg files hammered into a usable form is quite tedious. Also, getting the Polycoms to accept the new configs frequently involve defaulting the phone, or resetting the local configuration. Upgrading firmware on older phones may require many steps by upgrading through intermediate versions. Compared to an analog ATA, Polycoms are about 10 times more difficult and time consuming to get running well. If you haven't had to deal with these problems, count yourself very lucky. ___ --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] Polycom Speakerphone
Hi, Excellent ! For me, Polycom have the best audio. Just behind, I like also Aastra. Best Regards, Francois -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Eric Jacksch Envoyé : lundi 12 novembre 2007 15:39 A : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Polycom Speakerphone Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric ___ --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: 7.5.503 / Virus Database: 269.15.30/1125 - Release Date: 11/11/2007 21:50 ___ --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] VoiceMail hangup
Am Montag, den 12.11.2007, 15:14 -0500 schrieb Il Neofita: Hi additional information if I am going to wait at least 3 seconds after the voicemail starts to give me the instruction I am able to listen my messages. But why I need to wait? On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote: Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this problem? Read voicemail.conf. Look for minmessage setting - it will remove messages that are shorter than the given number of seconds. See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail See http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf BR Anselm ___ --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] VoiceMail hangup
Thank you for your answer. The problem is quite different for example, I am leaving a message of 5 seconds when I call to listen the message , asterisk answer and pass the call to voicemailmain and it plays the welcome message now if I press 1 before 3 or 4 seconds the voicemailmain gives me then information of the message send the command to play the message and it exists. If I wait more the 3 or 4 seconds and then I press 1 everything is going well for the same kind of message On Nov 12, 2007 3:53 PM, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Montag, den 12.11.2007, 15:14 -0500 schrieb Il Neofita: Hi additional information if I am going to wait at least 3 seconds after the voicemail starts to give me the instruction I am able to listen my messages. But why I need to wait? On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote: Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this problem? Read voicemail.conf. Look for minmessage setting - it will remove messages that are shorter than the given number of seconds. See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail See http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf BR Anselm ___ --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
[asterisk-users] ACD Queue LOG RINGNOANSWER Content 0
In my queue log I see that on the RINGNOANSWER Event I get different content. Some events soe the ring timeout (15000). Other events show 0. Other yet show 1000 Doens anyone know what 0 means? Did it try to ring the phone, but it was busy? Thanks Doug ___ --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] ACD Queue LOG RINGNOANSWER Content 0
Yes. That's supposed to to be the timeout value. In the case where it's 0 are you seeing a call reject or something else? asterisk wrote: In my queue log I see that on the RINGNOANSWER Event I get different content. Some events soe the ring timeout (15000). Other events show 0. Other yet show 1000 Doens anyone know what 0 means? Did it try to ring the phone, but it was busy? Thanks Doug ___ --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 -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] 'h' extension on call-out
Hello,
On Nov 12, 2007 5:52 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is not called
at all. Any idea?
I think you will find the 'h' extension is only called at the end of
a successful call.
Extension 'h' is executed on every hangup event from both side.
But your experience as described directly contradicts that assertion.
That's why I don't understand and asking here. :-)
If the call is unsuccessful, the Dial command will return, and you can
then check ${ISDNCAUSE} on the next line of your dialplan.
There is no dial command because I only wrote a call file to
/var/spool/asterisk/outgoing and actually dial has been made by
pbx_spool module. There is a 'failed' extension but no HANGUPCAUSE
there.
That is why you are not executing the 'h' extension. You are placing
the call directly to a channel. That channel does not start executing
the dial plan until it is answered. In that respect, my original
statement was correct. Because it only starts the dialplan when the
call is successfully answered, there is no context in which to execute
an 'h' extension on a failed call.
I see. Thanks for clarifying! In short there is no 'extra' extension
for 'hangup' like 'failed'.
Channel: Local/[EMAIL PROTECTED]
Context: mycontext
Extension: s
Priority: 1
Thank you! I will try this! Is local redirection have any
performance/resource hit on calls?
[outgoing]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
I see. Therefore Dial is executed within my context and I have
extension 'h' for it. Tricky.
exten =
_X.,n,NoOp(${EXTEN}:DIALSTATUS=${DIALSTATUS};HANGUPCAUSE=${HANGUPCAUSE})
Do I really need this? Because extension 'h' is executed on hangup so
this is redundancy I think.
exten =
h,n,NoOp(${EXTEN}:DIALSTATUS=${DIALSTATUS};HANGUPCAUSE=${HANGUPCAUSE})
In that case you should find the 'h' in [outgoing] gets executed on both
successful
and failed calls.
So I don't need NoOp for _X.. Right?
Many thanks for your informations!
Bye,
a
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Re: [asterisk-users] ztdummy and BackGround
Hello, On Nov/02/2007, Atis Lezdins wrote: On 11/2/07, Tony Plack [EMAIL PROTECTED] wrote: We are going to implement MeetMe, but this should still work right? I had similar issues with 1.4.12 just one time (also topmost zaptel at are you using Dell PowerEdge servers? -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --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] ztdummy, zttest
Hello, On Nov/12/2007, Tony Plack wrote: Debian as well as everyone else 2.6.18-5 Zaptel is branch/1.4 latest. The issue is not with Zaptel though...IMHO. If you look at /proc/driver/rtc, I find: periodic_IRQ: no If you notice, there is no periodic_IRQ. That is the issue. To me (and If I correctly remember, I have this in yes :-| the kernels reported here) it is a Debian kernel problem. Here working with Debian, in a Dell server. I have found some references about Dell and RTC. If I turn off ACPI, Zaptel works, but the box performance is awful because Not here :-( other things, which depend on ACPI do not have interrupts. I have read some places that if you have the Hard Drive Suppend in the BIOS enabled, you will get this situation. However, I would check the /proc/driver/rtc to see that you have periodic_IRQ. I will change BIOS options and other things on Wednesday, since we don't have right now physical access to this server. In other boxes ztdummy worked like a charm... -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --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] ODBC connection to Microsoft SQL Server
Hi, I wish to integrate a Microsoft SQL server with Asterisk for CDRs and for dialplan routing based on database values, and have this application scale to a large number of simultaneous calls: The Asterisk: The Future of Telephony 2nd edition book states that: ‡ The pooling and limit options are quite useful for MS SQL Server and Sybase databases. These permit you to establish multiple connections (up to limit connections) to a database while ensuring that each connection has only one statement executing at once (this is due to a limitation in the protocol used by these database servers). Does this suggest any kind of performance issue with scaling? I am assuming not as all this indicates is that DB queries are pooled from the ODBC connection on the Asterisk Server side rather than the SQL Server? Has anyone done this before in a large implementation? Any advice appreciated. Cheers Robert McNaught ___ --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] 'h' extension on call-out
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
If the call is unsuccessful, the Dial command will return, and you can
then check ${ISDNCAUSE} on the next line of your dialplan.
There is no dial command because I only wrote a call file to
/var/spool/asterisk/outgoing and actually dial has been made by
pbx_spool module. There is a 'failed' extension but no HANGUPCAUSE
there.
That is why you are not executing the 'h' extension. You are placing
the call directly to a channel. That channel does not start executing
the dial plan until it is answered. In that respect, my original
statement was correct. Because it only starts the dialplan when the
call is successfully answered, there is no context in which to execute
an 'h' extension on a failed call.
I see. Thanks for clarifying! In short there is no 'extra' extension
for 'hangup' like 'failed'.
Channel: Local/[EMAIL PROTECTED]
Context: mycontext
Extension: s
Priority: 1
Thank you! I will try this! Is local redirection have any
performance/resource hit on calls?
No, the Local channel is just used to set up the call, and once it is
answered, the extra channels are optimised out so that you end up with
the same channel as if you had called it directly.
[outgoing]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
I see. Therefore Dial is executed within my context and I have
extension 'h' for it. Tricky.
Yes, but this is a different context from the one you want the answered
call to execute. In my example, outgoing and mycontext must be different,
unless there will be no confusion between possible extensions. I always
keep them different to avoid confusion.
exten =
_X.,n,NoOp(${EXTEN}:DIALSTATUS=${DIALSTATUS};HANGUPCAUSE=${HANGUPCAUSE})
Do I really need this? Because extension 'h' is executed on hangup so
this is redundancy I think.
exten =
h,n,NoOp(${EXTEN}:DIALSTATUS=${DIALSTATUS};HANGUPCAUSE=${HANGUPCAUSE})
In that case you should find the 'h' in [outgoing] gets executed on both
successful
and failed calls.
So I don't need NoOp for _X.. Right?
The NoOps were just so you could see which conditions caused each to be
called. You would replace or follow the NoOp with some conditional statements
that act on the status values.
It doesn't hurt to have the NoOps in both places. At least until you are sure
'h' catches all the conditions you are interested in. For example you might
want _X. to go on and try again or try a different operation that you couldn't
do within 'h'.
Many thanks for your informations!
Glad to help. Hope you get it working the way you want!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Record() : How to get filename created with %d?
On Mon, 12 Nov 2007 09:58:50 + (UTC), [EMAIL PROTECTED]
(Tony Mountifield) wrote:
I'm a little surprised at the variety of band-aid suggestions that have
been posted. All you need to do is refer to show application record,
and you uwill see that the generated filename is available by using
${RECORDED_FILE}
Thanks for the tip. The reason I was looking for another solution is
that I couldn't get the value of the variable... but it's working now.
I'm not comfortable yet with functions/applications, and have no idea
what I did wrong :-/
BTW, what's the difference between functions and applications?
For those interested, here's some working code:
==
exten = 555,1,Record(/tmp/msg%d.wav,3,30)
exten = 555,n,Verbose(${RECORDED_FILE})
exten = 555,n,TrySystem(mv ${RECORDED_FILE}.wav /var/www/asterisk/)
exten = 555,n,ExecIf($[${SYSTEMSTATUS} != SUCCESS],Verbose,Failed
moving WAV file)
==
Another way to generate a filename dynamically, using the current date
+ time:
==
exten =
_[1-4],n,Set(CALLTIME=${STRFTIME(${EPOCH},GMT+1,%d-%b-%Y-%Hh%M)})
exten = _[1-4],n,Record(/tmp/${CALLTIME}.wav,3,30)
exten = _[1-4],n,TrySystem(mv /tmp/${CALLTIME}.wav
/var/www/asterisk/)
exten = _[1-4],n,ExecIf($[ ${SYSTEMSTATUS} != SUCCESS
],Verbose,Failed moving WAV file)
==
Thanks a lot guys for the help.
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Re: [asterisk-users] Record() : How to get filename created with %d?
On Sun, 11 Nov 2007 11:18:30 -0600, Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: You need to look at the files in /path/to/src/asterisk/doc (or /docs, I don't recall) there is much information there, including a file named README.variables (1.2) or channelvariables.txt (1.4). Will do. 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] Record() : How to get filename created with %d?
On Sun, 11 Nov 2007 13:16:35 -0400, Baji Panchumarti [EMAIL PROTECTED] wrote: you can generate your own name using a combo of STRFTIME() CALLERID() CDR() ( and RAND() if you like ) Thanks for the tip. That's what I'll end up doing, as the filename is more descriptive than just using a monotonously incremented integer. ___ --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] sangoma zaptel patches
- Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 11, 2007 8:21 PM Subject: Re: [asterisk-users] sangoma zaptel patches On Sunday 11 November 2007 11:07:04 Steve Totaro wrote: Tzafrir Cohen wrote: Sangoma's s setup process includes a small patch to Zaptel. I have some technical reservations with that patch. It seems that under certain circumstances it may cause unexpected behavior when used with other Zaptel channel drivers. I also don't understand why a safer method is not used. Just out of curiosity, I have yet to see any issues with Sangoma cards and the way they ride on top (and patch) the Zaptel drivers. This personal dataset is around one hundred productions boxes. How many of those boxes are of the type that Tzafrir is worried about? Specifically, how many of those boxes contain a combination of telephony hardware from vendors other than Sangoma? I have a box that now has a TDM400P. I will be installing a sangoma card in it soon and I actually need support for 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] 'Traditional' Faxing
- Original Message - From: Jonn R Taylor [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, November 10, 2007 5:45 PM Subject: Re: [asterisk-users] 'Traditional' Faxing Greg Cockburn wrote: Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there maybe someway to bridge an Analogue FXS port to a channel on the E1? Basically I want to mimic what we have now. 1. Any person can send a fax using the fax machine, and the PBX picks the next free channel on the E1. 2. A fax call can come over any channel on the E1, and the dialed number is matched and sent to the analogue FXS port of the PBX to be received by the fax machine. Is there anyway I can do this in Asterisk that will work seamlessly? I have not yet purchased any hardware, so recommendations would be greatly appreciated. (I believe some of the problem exists due to timing, does any hardware; E1 card / Analogue card; support linking a timing signal together?) Sangoma, Digium, Pika? Thanks all for any help on this one. Greg. ___ --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 Greg, There are alot of option for handeling faxes. One is to use iaxmodem and hylafax. This option works the best. You can try to use an analog adapter or card to connect a conventional fax to but this is not allways reliable. I have spent alot of time working on faxing with asterisk. If you need any help you can email me and I will send the links and scripts that I have to help you in your setup. FYI, They are all for RH/CentOS. Hardware, how many phone and trunks do you plan on using? Digium cards for analog phone's and faxes work very well, linksys makes very good ATA's too. Digium or Sangoma T1 cards are the most suppoted that I have seen. but there are others. OS, there are alot of different *nix OS's that are out there. Pick the one that you are the most comfortable to use. Asterisk was developed on RedHat though. Depending on your needs for support I would suggest either EL4 or CentOS4 with Asterisk 1.2. There are alot of people running 1.4 in production but the commercaial version of Asterisk is still on 1.2 John, Do you mind posting a link to those CentOS scripts here ? Thanks. Dovid ___ --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] 'Traditional' Faxing
I am also very interested in these scripts. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, 13 November 2007 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 'Traditional' Faxing - Original Message - From: Jonn R Taylor [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, November 10, 2007 5:45 PM Subject: Re: [asterisk-users] 'Traditional' Faxing Greg Cockburn wrote: Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there maybe someway to bridge an Analogue FXS port to a channel on the E1? Basically I want to mimic what we have now. 1. Any person can send a fax using the fax machine, and the PBX picks the next free channel on the E1. 2. A fax call can come over any channel on the E1, and the dialed number is matched and sent to the analogue FXS port of the PBX to be received by the fax machine. Is there anyway I can do this in Asterisk that will work seamlessly? I have not yet purchased any hardware, so recommendations would be greatly appreciated. (I believe some of the problem exists due to timing, does any hardware; E1 card / Analogue card; support linking a timing signal together?) Sangoma, Digium, Pika? Thanks all for any help on this one. Greg. ___ --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 Greg, There are alot of option for handeling faxes. One is to use iaxmodem and hylafax. This option works the best. You can try to use an analog adapter or card to connect a conventional fax to but this is not allways reliable. I have spent alot of time working on faxing with asterisk. If you need any help you can email me and I will send the links and scripts that I have to help you in your setup. FYI, They are all for RH/CentOS. Hardware, how many phone and trunks do you plan on using? Digium cards for analog phone's and faxes work very well, linksys makes very good ATA's too. Digium or Sangoma T1 cards are the most suppoted that I have seen. but there are others. OS, there are alot of different *nix OS's that are out there. Pick the one that you are the most comfortable to use. Asterisk was developed on RedHat though. Depending on your needs for support I would suggest either EL4 or CentOS4 with Asterisk 1.2. There are alot of people running 1.4 in production but the commercaial version of Asterisk is still on 1.2 John, Do you mind posting a link to those CentOS scripts here ? Thanks. Dovid ___ --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] asterisk and installing chan_h323.so rpm
- Original Message - From: Bincy K. Philip [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 08, 2007 2:13 PM Subject: [asterisk-users] asterisk and installing chan_h323.so rpm Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same files? I had this issue in the past. I do not remember what I did to resolve it. In the end I went with the h323 channel driver located in the asterisk add-ons. It was a lot easier to work with and worked with out any issues. ___ --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 direct dialing
- Original Message -
From: Gopal krishnan [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, November 10, 2007 12:46 PM
Subject: [asterisk-users] Asterisk direct dialing
Hi,
I am using Asterisk 1.2.24, I have written my dialplan to land
with an IVR with the same time if the customer knows the parties
extensions they can dial directly, but what happens is sometimes its
working and sometime its not working.
My extensions.conf as follows,
[incoming]
exten = 052477302,1,Wait(2)
exten = 052477302,2,NoOp(${CALLERIDNUM})
exten = 052477302,3,Goto(from-internal,s,1)
[from-internal]
exten = s,1,Answer()
exten = s,2,Background(welcome_pride5)
exten = 501,1,Dial(Zap/1)
exten = 502,1,Dial(Zap/2)
exten = 503,1,Dial(Zap/3)
exten = 504,1,Dial(Zap/4)
exten = 505,1,Dial(Zap/5)
I dont know what could be the reason. Is there any other way that i can
use.
--
Thank you, with regards,
Gopal,
What comes up in the CLI when it does not work ?
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Re: [asterisk-users] Help: Static and dropped calls
You need to provide more information that just that. Maybe a CLI output ? Have you tested with any other providers ? SIP debug ? Ran a trace ? We aren't mind readers here. - Original Message - From: Jarga Jallow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 05, 2007 8:27 PM Subject: [asterisk-users] Help: Static and dropped calls Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters. Thank you Jarga Jallow -- ___ --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-usersimage001.jpg___ --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 delete voice mail messages?
I wrote this for a client a while back:
[del-all-vm]
exten = s,1,Set(TIMEOUT(digit)=3)
exten = s,2,Set(TIMEOUT(response)=6)
exten = s,3,Background(enter-exten-for-vm-to-delete)
exten = _XX,1,Set(THIER_EXTEN=${EXTEN})
exten = _XX,2,Goto(del-all-vm-confirm,s,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = t,1,Goto(s,1)
[del-all-vm-confirm]
exten = s,1,Set(TIMEOUT(digit)=3)
exten = s,2,Set(TIMEOUT(response)=6)
exten = s,3,Background(are-you-sure-del-all-vm)
exten = s,4,Saynumber(${THIER_EXTEN})
exten = s,5,Background(1-for-yes-2-for-no)
exten = 1,1,System(rm -rf
/var/spool/asterisk/default/techmast/${THIER_EXTEN}/INBOX/*.*)
exten = 1,2,Playback(all-vm-deleted)
exten = 1,3,Congestion
exten = 1,4,Hangup
exten = 2,1,Playback(close-call-not-deleted)
exten = 2,2,Congestion
exten = 2,3,Hangup
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = t,1,Goto(s,1)
- Original Message -
From: voip crazy
To: asterisk-users@lists.digium.com
Sent: Monday, November 05, 2007 1:15 PM
Subject: [asterisk-users] How to delete voice mail messages?
Hello all,
Could I create a script to delete the first messages on my voice mail? In
this script should I update any messages index file or there isn't any file
to index them? Could you share any script to do that?
Thanks in advance.
VoipCrazy.
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Re: [asterisk-users] asterisk and installing chan_h323.so rpm
- Original Message - From: Bincy K. Philip [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 08, 2007 2:13 PM Subject: [asterisk-users] asterisk and installing chan_h323.so rpm Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same files? I had this issue in the past. I do not remember what I did to resolve it. In the end I went with the h323 channel driver located in the asterisk add-ons. It was a lot easier to work with and worked with out any issues. It seems to me that you need to run ldconfig so as to pick up the location of the specified libraries. Do a google on it to see syntax of man ldconfig. You could also hack things by linking to the libraries from the expected directories (What the rpm is expecting) if executing ldconfig doesn't work. db ___ --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] ztdummy and BackGround
No, but if you have a hint, I would love it. This is still plaguing me. Hello, On Nov/02/2007, Atis Lezdins wrote: On 11/2/07, Tony Plack [EMAIL PROTECTED] wrote: We are going to implement MeetMe, but this should still work right? I had similar issues with 1.4.12 just one time (also topmost zaptel at are you using Dell PowerEdge servers? ___ --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] MOH Codec Issue
Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/10.97.1.33-09f0cfc8, sounds) in new stack [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a codec translation path from alaw to unknown [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' -- Started music on hold, class '?S?', on channel 'SIP/10.97.1.33-09f0cfc8' [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8 Have attempted to use an alternate Music On Hold context and forced a format= within musiconhold.conf. Otherwise all other audio (Playback, voice etc) seems fine. Anyone seen this before? Can not see anything in the tracker regarding this issue in 1.4.7 specifically. Cheers Nick. ___ --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] RTP traffic not being forwarded
Hi Vivek, Thanks for the link. I had a look through and couldn't find anything that worked. There are no NAT problems as this is all taking place on my internal network. The rtp.conf is used to configure the ports. There are no firewalls or gateways in between these devices. Asterisk is listening on the correct ports, and receiving the traffic, as no ICMP messages are being generated to say that the packets could not be delivered. Ryan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vivek Shrivastava Sent: Monday, 12 November 2007 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded well i think rtp port range is defined in rtp.conf and correct me if i am wrong, these ports must be opened/forwarded to communicate. http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Let me know if you need more information. Thanks, Vivek On 11/11/07, Ryan Newington [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: Hi Vivek, I'm not sure what you mean, could you explain further? Regards Ryan From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]mailto:[EMAIL PROTECTED]] On Behalf Of Vivek Shrivastava Sent: Monday, 12 November 2007 1:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, I was just wondering if they need to be according rtp.conf. ( or you may need to modify rtp.conf) Regards, Vivek On 11/11/07, Ryan Newington [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: Hi Vivek, The SIP port is set to the standard port 5060. The RTP ports as far as I know are random ephemeral ports between 63000 and 64000. I can change the port range on the media server, asterisk and the device, but neither seems to help. My diagram below is probably misleading. The RTP traffic flow that I see is as follows (one way traffic into Asterisk) SIP Phone --- Media Gateway --- Asterisk --- SIP Phone Ryan From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]mailto:[EMAIL PROTECTED]] On Behalf Of Vivek Shrivastava Sent: Sunday, 11 November 2007 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking random port selection option on the device/softphone may help. --Vivek On 11/10/07, Ryan Newington [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] wrote: Hi Luki, Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success. Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway. SIP Phone - Media Gateway - Asterisk - SIP Phone An asterisk internal call will work fine. Eg; SIP Phone - Asterisk - SIP Phone Regards Ryan -Original Message- From: [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Luki Sent: Sunday, 11 November 2007 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded When using 'rtp debug' on the asterisk console, it shows that it is receiving traffic from one endpoint, but not the other. A wireshark trace reveals it is actually receiving traffic from both ends. Sounds like a firewall issue. Wireshark shows what's on the wire, i.e. before iptables. The packets are being dropped for whatever reason and never reach the asterisk process. Check your iptables and RTP port range, and perhaps try changing it. Luki ___ --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-- 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-- 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--http://www.api-digital.com--/ asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] MOH Codec Issue
What format is your music on hold in? PaulH On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/10.97.1.33-09f0cfc8, sounds) in new stack [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a codec translation path from alaw to unknown [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' -- Started music on hold, class '?S?', on channel 'SIP/10.97.1.33-09f0cfc8' [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8 Have attempted to use an alternate Music On Hold context and forced a format= within musiconhold.conf. Otherwise all other audio (Playback, voice etc) seems fine. Anyone seen this before? Can not see anything in the tracker regarding this issue in 1.4.7 specifically. Cheers Nick. ___ --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
[asterisk-users] Chatterbug
Does anyone know anything about the Chatterbug product? I can't tell if it's an ATA with a modem or some sort of LCR proxy or somesuch. Anyone? ___ --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] MOH Codec Issue
It was using the 3 wav's from Freeplay. I have just recompiled and told it to pull down the ULAW versions, then removed the Wav's however it has made no difference. Cheers Nick On 13/11/07 3:56 PM, Paul Hales wrote: What format is your music on hold in? PaulH On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/10.97.1.33-09f0cfc8, sounds) in new stack [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a codec translation path from alaw to unknown [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' -- Started music on hold, class '?S?', on channel 'SIP/10.97.1.33-09f0cfc8' [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8 Have attempted to use an alternate Music On Hold context and forced a format= within musiconhold.conf. Otherwise all other audio (Playback, voice etc) seems fine. Anyone seen this before? Can not see anything in the tracker regarding this issue in 1.4.7 specifically. Cheers Nick. ___ --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 Regards, Nick Brown Ipera Communications Pty Ltd Level 1, 9 Denison Street, Newcastle West NSW 2302 PO Box 2115, Dangar NSW 2309 Ü P: +61 2 4910 1000 Ü F: +61 2 4910 1099 Ü ABN: 31 090 964 104 ___ --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] Chatterbug
http://www.oldskoolphreak.com/tfiles/voip/chatter_bug.pdf PaulH On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote: Does anyone know anything about the Chatterbug product? I can't tell if it's an ATA with a modem or some sort of LCR proxy or somesuch. Anyone? ___ --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] MOH Codec Issue
Is it possibly a funny zaptel issue? Paul Hales AsteriskIT On Tue, 2007-11-13 at 16:20 +1100, Nick Brown wrote: It was using the 3 wav's from Freeplay. I have just recompiled and told it to pull down the ULAW versions, then removed the Wav's however it has made no difference. Cheers Nick On 13/11/07 3:56 PM, Paul Hales wrote: What format is your music on hold in? PaulH On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/10.97.1.33-09f0cfc8, sounds) in new stack [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a codec translation path from alaw to unknown [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' -- Started music on hold, class '?S?', on channel 'SIP/10.97.1.33-09f0cfc8' [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8 Have attempted to use an alternate Music On Hold context and forced a format= within musiconhold.conf. Otherwise all other audio (Playback, voice etc) seems fine. Anyone seen this before? Can not see anything in the tracker regarding this issue in 1.4.7 specifically. Cheers Nick. ___ --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 Regards, Nick Brown Ipera Communications Pty Ltd Level 1, 9 Denison Street, Newcastle West NSW 2302 PO Box 2115, Dangar NSW 2309 Ü P: +61 2 4910 1000 Ü F: +61 2 4910 1099 Ü ABN: 31 090 964 104 ___ --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
[asterisk-users] Stress-Testing Asterisk
Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Thanks, Jeng ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ --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] Stress-Testing Asterisk
Jeng Yu wrote: Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Try SIPp from HP (http://sipp.sourceforge.net/index.html) or SIP swiss army knife - SIPSAK (http://sipsak.org/) -- Suich ___ --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
