Re: [asterisk-users] asterisk and cisco 2800
Hi Neeraj, my problem is not terminating but making the Cisco accept the calls coming from my Asterisk. The bad news is I cannot have access to the Cisco sw, it is like a black box for me. The only thing I can have access to is the T1/E1 port on the back of the Cisco 2800. I made a custom cable too: 1 -- 5 2 -- 4 4 -- 2 5 -- 1 and it seems to work because I get all alarms off after plugging it in. Thank you Giorgio Incantalupo Neeraj Chand wrote: Hi Giorgio, Why don't you terminate calls on the cisco router via SIP? -- Message: 11 Date: Fri, 02 Jul 2010 18:54:31 +0200 From: Giorgio Incantalupo gincantal...@fgasoftware.com Subject: [asterisk-users] asterisk and cisco 2800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4c2e19c7.5090...@fgasoftware.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives no errros, the span is up and active, green light on the card) but when I make a test with my iax phone, there's no way to dial the PBX and I get this WARNING: [Jul 2 15:20:36] VERBOSE[15004] logger.c: -- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:1] Dial(IAX2/1-1024, DAHDI/g2/X|60|tT) in new stack [Jul 2 15:20:36] WARNING[15031] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:2] Hangup(IAX2/1-1024, ) in new stack [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Spawn extension (inbound, , 2) exited non-zero on 'IAX2/1-1024' [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Hungup 'IAX2/1-1024' Any hints? Thank you. Giorgio Incantalupo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 2 July, 2010 5:08:14 PM
Subject: Re: [asterisk-users] Voiceprompts i.e. voicemail and conferencing in
multiple codecs
On Fri, 2 Jul 2010, Kenny Watson wrote:
for i in `ls -R /var/lib/asterisk/sounds/uk/*wav`; # do recursive ls and
only list wav files and loop through each one
do # start do loop
CONV=`echo $i|sed 's/.wav/.g729/g'` # set CONV variable as filename with
wav swapped for G729
asterisk -rx file convert $i $CONV # run convert command placing
original filename and new filename in command
done # end loop
First off, Digium provides core prompts in many encodings at
http://downloads.asterisk.org/pub/telephony/sounds/
Assuming those packages don't meet your needs, this script doesn't do what
you intended.
Because you specified *wav, the ls -R is not recursing through the
directories, it's only listing the wav files in the specified directory.
Without the *wav, ls would recurse, but it wouldn't display the path.
The find command is a better tool for this.
Compare:
$ ls -R /var/lib/asterisk/sounds/*wav | wc -l
241
and
$ find /var/lib/asterisk/sounds/ -name '*.wav' | wc -l
1596
And then, a few suggestions* on better practices.
) Use $( and ) instead of back-ticks. They can be nested and they
survive emailing, printers (the human kind), telephones, and strange fonts
better.
) Use shell functions instead of processes. You create 2 processes for
every file just to substitute the file type.
) Use upper case for variables** so they stand out.
) Don't use single character variable names -- we don't code in FORTRAN
any more do we? Also, they're obtuse and a bitch to search for in an
editor.
) Use comments before a block of code instead of line by line.
# Recurse through the input path, [ab]using Asterisk*** to
# convert WAV files to G729
INPUT_PATH=/var/lib/asterisk/sounds/
for INPUT in $(find ${INPUT_PATH} -name '*.wav')
do
OUTPUT=${INPUT%.wav}.g729
asterisk -r -x file convert ${INPUT} ${OUTPUT}
done
# (end of snippet)
I'm sure some shell weenie could code it all up in a single arcane line of
cruft, but the goal should be clarity and maintainability.
*) I recognize some of these suggestions reflect my religious beliefs.
If you all would get over it and just convert we can all just get
along.
**) Being an old-school c weenie I don't know why I prefer upper case
for variables in shell scripts, but I do. Whatever case you prefer, be
consistent.
***) I know command line g729 encoders are hard to come by, but do you
really want to explain to your boss that you crashed the PBX trying to
convert some really funny Alison prompts you found on the net but one of
them had a funky header and it exposed a hither-to unknown bug?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Steve,
Thank you for the extensive response, I'm using some UK prompts on my system so
would like to convert them to g729.
The new script and tips on scripting will be helpful.
Thanks
Kenny
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ARA : Realtime or not ?
Hello list, what is the use of realtime SIP peers when you always need to reload the sip configuration as if you were just putting your SIP peers in sip.conf ?? My SIP peers are now defined in a mysql-DB and when I add a mailbox in the field 'mailbox', the change is not active untill a do a sip reload or a module reload chan_sip.so. Doing a sip reload or a module reload chan_sip.so makes that all the SIP peers need to re-register again to Asterisk so that they are 'available' again. So what is realtime about the ARA ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARA : Realtime or not ?
On 06/07/10 10:34, Jonas Kellens wrote: Hello list, what is the use of realtime SIP peers when you always need to reload the sip configuration as if you were just putting your SIP peers in sip.conf ?? My SIP peers are now defined in a mysql-DB and when I add a mailbox in the field 'mailbox', the change is not active untill a do a sip reload or a module reload chan_sip.so. Doing a sip reload or a module reload chan_sip.so makes that all the SIP peers need to re-register again to Asterisk so that they are 'available' again. So what is realtime about the ARA ?? Kind regards, Jonas. If you are using RealTime and make any changes to a peer entry in your DB you need to prune the peer from the realtime cache using sip prune realtime peer name The peer will need to re register but only that one and it will pick up the updated settings from the DB Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARA : Realtime or not ?
On 6 Jul 2010, at 10:34, Jonas Kellens wrote: what is the use of realtime SIP peers when you always need to reload the sip configuration as if you were just putting your SIP peers in sip.conf ?? Did you enable caching by any chance? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARA : Realtime or not ?
Hello, this is my configuration : ;- REALTIME SUPPORT ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read realtime.txt and extconfig.txt in the /doc directory of the ; source code. ; rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) ;rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. Note: realtime peers will ; probably not function across reloads in the way that you expect, if ; you turn this option off. ;rtautoclear=yes; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|seconds) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage Kind regards, Jonas. On 07/06/2010 11:56 AM, Steve Howes wrote: Did you enable caching by any chance? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARA : Realtime or not ?
On 07/06/2010 12:00 PM, Ishfaq Malik wrote: On 06/07/10 10:34, Jonas Kellens wrote: Hello list, what is the use of realtime SIP peers when you always need to reload the sip configuration as if you were just putting your SIP peers in sip.conf ?? My SIP peers are now defined in a mysql-DB and when I add a mailbox in the field 'mailbox', the change is not active untill a do a sip reload or a module reload chan_sip.so. Doing a sip reload or a module reload chan_sip.so makes that all the SIP peers need to re-register again to Asterisk so that they are 'available' again. So what is realtime about the ARA ?? Kind regards, Jonas. If you are using RealTime and make any changes to a peer entry in your DB you need to prune the peer from the realtime cache using sip prune realtime peer name The peer will need to re register but only that one and it will pick up the updated settings from the DB Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 OK, that's when changing peer-attributes. What can I do when adding new peers to the database ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi - alarm which clears itself - Should I care ?
Hi, When reading logs, I can see a couple of lines such as : full.6:[Jun 30 15:53:26] NOTICE[6599] chan_dahdi.c: PRI got event: Alarm (4) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6599] chan_dahdi.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6607] chan_dahdi.c: Alarm cleared on channel 1 full.6:[Jun 30 15:53:32] NOTICE[6607] chan_dahdi.c: Alarm cleared on channel 2 In a 3 BRI spans powered system, this occurred 4 times last week : 1 time for one span, 3 times for another one. Each time, there is a 6 seconds delay between Alarm and Alarm clearance. My setup is: Asterisk 1.6.1.18, Dahdi 2.3.0 with OSLEC Libpri 1.4.10.2 Junghanns OctoBRI with wcb4xxb driver I'm not really worried about this but should I care ? Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARA : Realtime or not ?
On 06/07/10 11:26, Jonas Kellens wrote: On 07/06/2010 12:00 PM, Ishfaq Malik wrote: On 06/07/10 10:34, Jonas Kellens wrote: Hello list, what is the use of realtime SIP peers when you always need to reload the sip configuration as if you were just putting your SIP peers in sip.conf ?? My SIP peers are now defined in a mysql-DB and when I add a mailbox in the field 'mailbox', the change is not active untill a do a sip reload or a module reload chan_sip.so. Doing a sip reload or a module reload chan_sip.so makes that all the SIP peers need to re-register again to Asterisk so that they are 'available' again. So what is realtime about the ARA ?? Kind regards, Jonas. If you are using RealTime and make any changes to a peer entry in your DB you need to prune the peer from the realtime cache using sip prune realtime peer name The peer will need to re register but only that one and it will pick up the updated settings from the DB Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 OK, that's when changing peer-attributes. What can I do when adding new peers to the database ?? Kind regards, Jonas. You don't need to do anything when adding new peers to the database, when the peer first tries to register the config will be taken from the DB and put into the realtime cache. You can see the exact config of any peer in the cache by doing the following in the console sip show peer peer name Seeing that might help, I know it did for me. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
On Tue, Jul 6, 2010 at 1:09 AM, C.Savinovich c.savinov...@itntelecom.com wrote: I am writing to you privately because I am an asterisk consultant and if you need any help I can help you for a fee. Unfortunately your email is not private, now that it is on a public list. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to secure Configuration files
Hello Community, I have a question , I have been working with asterisk and developed some successful applications. I am facing an issue of security i.e. We deploy servers to client end. Now i dont want the client to see my configuration files (Of course copy and distribute or replicate the logic with out permission). Now the configuration files are stored in /etc/asterisk/* (Of course we can specify a different location but at end we specify this in a configuration file). Is there a way that the configuration files get encrypted or some thing else so that some one who have system access can not copy the configuration files data or look into that files. -- Best Regards Shakeel Abbas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi - alarm which clears itself - Should I care ?
On Tue, Jul 06, 2010 at 12:37:25PM +0200, Olivier wrote: Hi, When reading logs, I can see a couple of lines such as : full.6:[Jun 30 15:53:26] NOTICE[6599] chan_dahdi.c: PRI got event: Alarm (4) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6599] chan_dahdi.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6607] chan_dahdi.c: Alarm cleared on channel 1 full.6:[Jun 30 15:53:32] NOTICE[6607] chan_dahdi.c: Alarm cleared on channel 2 In a 3 BRI spans powered system, this occurred 4 times last week : 1 time for one span, 3 times for another one. Each time, there is a 6 seconds delay between Alarm and Alarm clearance. It should be harmless. The provider took the line down. If outgoing and incoming calls work well, nothing to worry about. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and cisco 2800
That's not right. Should be 1245 - 4512: http://www.voip-info.org/wiki/view/crossover+T1+cable -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Tuesday, July 06, 2010 2:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 2800 Hi Neeraj, my problem is not terminating but making the Cisco accept the calls coming from my Asterisk. The bad news is I cannot have access to the Cisco sw, it is like a black box for me. The only thing I can have access to is the T1/E1 port on the back of the Cisco 2800. I made a custom cable too: 1 -- 5 2 -- 4 4 -- 2 5 -- 1 and it seems to work because I get all alarms off after plugging it in. Thank you Giorgio Incantalupo Neeraj Chand wrote: Hi Giorgio, Why don't you terminate calls on the cisco router via SIP? -- Message: 11 Date: Fri, 02 Jul 2010 18:54:31 +0200 From: Giorgio Incantalupo gincantal...@fgasoftware.com Subject: [asterisk-users] asterisk and cisco 2800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4c2e19c7.5090...@fgasoftware.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives no errros, the span is up and active, green light on the card) but when I make a test with my iax phone, there's no way to dial the PBX and I get this WARNING: [Jul 2 15:20:36] VERBOSE[15004] logger.c: -- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:1] Dial(IAX2/1-1024, DAHDI/g2/X|60|tT) in new stack [Jul 2 15:20:36] WARNING[15031] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:2] Hangup(IAX2/1-1024, ) in new stack [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Spawn extension (inbound, , 2) exited non-zero on 'IAX2/1-1024' [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Hungup 'IAX2/1-1024' Any hints? Thank you. Giorgio Incantalupo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to secure Configuration files
On Tue, Jul 6, 2010 at 7:40 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello Community, I have a question , I have been working with asterisk and developed some successful applications. I am facing an issue of security i.e. We deploy servers to client end. Now i dont want the client to see my configuration files (Of course copy and distribute or replicate the logic with out permission). You have no problems using Asterisk and GPL, but not distributing your settings? Is there a way that the configuration files get encrypted or some thing else so that some one who have system access can not copy the configuration files data or look into that files. No, asterisk still needs permission to view your config files. Lock down the box and deny your customer access to the hardware. $ chmod -R 600 /etc/asterisk Of course, this is all a moot point if they have physical access to the machine. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi on solaris
Thanks Bruce, I also think dahdi is not able to compile there, I see it requires linux kernel. I'll give a try with zaptel.. but... Do you know if most up to date version of zaptel/solaris is at solarisvoip.com? Claudio On Mon, 5 Jul 2010, Bruce McAlister wrote: Hi Claudio, As far as I am aware, dahdi is not able to compile on Solaris, although I've not attempted to compile it. There may be others out there that may have better experience than I with dahdi on Solaris. Thanks Bruce -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Claudio Furrer Sent: 05 July 2010 22:11 To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi on solaris Hello all, Does anybody know if is it possible to install dahdi on solaris 10? I've only found a zaptel modified code for solaris at solarisvoip site. I'd appreciate any comment or experience about asterisk + dahdi/zaptel on solaris.. Best regards, Caio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to secure Configuration files
On Tue, 6 Jul 2010, ABBAS SHAKEEL wrote: Hello Community, I have a question , I have been working with asterisk and developed some successful applications. I am facing an issue of security i.e. We deploy servers to client end. Now i dont want the client to see my configuration files (Of course copy and distribute or replicate the logic with out permission). Now the configuration files are stored in /etc/asterisk/* (Of course we can specify a different location but at end we specify this in a configuration file). Is there a way that the configuration files get encrypted or some thing else so that some one who have system access can not copy the configuration files data or look into that files. The simple answer is that you can't prevent anyone copying it if they have physical access. All you can do is make it hard. If you wanted to encrypt them, you'd need to alter asterisk. You could use something like trucrypt, or another whole disk encryption technology, but that'll require someone typing in a password at boot time making unattended reboots impossible. Another way which I have seen is to do away with the dialplan entirely and do it all in a single big compiled AGI C program. (Ok, you have minimal dialplan to pump everything into it, but...) and don't distribute the source to the C program... You need to work out just what it's worth to you if someone does copy it. Realistically, what's your target audience? Are your clients the sort of people likely to copy and and sell it on? For most businesses, I'd guess not. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On Tue, Jun 29, 2010 at 10:39 AM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: I use SecureCRT+FX , and use ansi graphics. Putty is nice w/WinSCP as well. I'll +1 this - SecureCRT+FX is the first thing I got my employer to buy a license of for me when I had to start using Windows on my desktop for other reasons instead of a Linux Distro. I do keep a copy of PuTTY handy on too though, the great thing about putty is that it doesn;t require it to be installed on a desktop, so you can just keep a copy of the executable on a USB flash drive or Windows share that you can then run from any desktop if you happen to be at a computer other than your own. PuTTY works well, but there are some things in it that just drive me absolutely crazy, like right-click in the window is an automatic paste... i have pasted into a putty window by accident more times than I can count, most of the time its from doing a right-click expecting a context menu to get a 'copy' action and then i just end up pasting what i actually wanted to copy :| -- Matt -- Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and cisco 2800
Hi Peder, I'make a new cable following the info on that webpage. I hope it works with Cisco 2800 too! :) Thank you! Giorgio Incantalupo Peder wrote: That's not right. Should be 1245 - 4512: http://www.voip-info.org/wiki/view/crossover+T1+cable -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Tuesday, July 06, 2010 2:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 2800 Hi Neeraj, my problem is not terminating but making the Cisco accept the calls coming from my Asterisk. The bad news is I cannot have access to the Cisco sw, it is like a black box for me. The only thing I can have access to is the T1/E1 port on the back of the Cisco 2800. I made a custom cable too: 1 -- 5 2 -- 4 4 -- 2 5 -- 1 and it seems to work because I get all alarms off after plugging it in. Thank you Giorgio Incantalupo Neeraj Chand wrote: Hi Giorgio, Why don't you terminate calls on the cisco router via SIP? -- Message: 11 Date: Fri, 02 Jul 2010 18:54:31 +0200 From: Giorgio Incantalupo gincantal...@fgasoftware.com Subject: [asterisk-users] asterisk and cisco 2800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4c2e19c7.5090...@fgasoftware.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives no errros, the span is up and active, green light on the card) but when I make a test with my iax phone, there's no way to dial the PBX and I get this WARNING: [Jul 2 15:20:36] VERBOSE[15004] logger.c: -- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:1] Dial(IAX2/1-1024, DAHDI/g2/X|60|tT) in new stack [Jul 2 15:20:36] WARNING[15031] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:2] Hangup(IAX2/1-1024, ) in new stack [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Spawn extension (inbound, , 2) exited non-zero on 'IAX2/1-1024' [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Hungup 'IAX2/1-1024' Any hints? Thank you. Giorgio Incantalupo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR
How can I match any_num_of_digits#any_num_of_digits in an IVR?
I want users to be able to type, eg., 123#4567
I tried the following but it hangs up immediately. If I uncomment WaitExten
then it hangs up right when the user dials #.
As a side question, can I play a background message while using the Read()
command?
[FILTER-validate]
exten = h,1,Hangup()
exten = hang,1,Hangup()
exten = s,1,Set(CANCALL=1)
exten = s,n,Set(LOOPCOUNT=0)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=5)
exten = s,n(repeatme),Background(TEST/FILTER_VALIDATE_1)
;exten = s,n,WaitExten(5,m(default))
exten = _X.#XXX.,1,Playback(one-moment-please)
exten = _X.#XXX.,n,AGI(filter-validate.agi|${EXTEN})
exten = _X.#XXX.,n,GotoIf($[${CANCALL} =
1]?outbound,${CANCALL_EXTEN},filterok)
exten = _X.#XXX.,n,Playback(TEST/FILTER_VALIDATE_3)
exten = _X.#XXX.,n,Hangup()
exten = t,1,Hangup()
exten = i,1,Playback(invalid)
exten = i,n,Goto(loop,1)
exten = loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten = loop,n,GotoIf($[${LOOPCOUNT} 2]?hang,1)
exten = loop,n,Goto(FILTER-validate,s,repeatme)
Thanks,
Vieri
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Tuesday, July 06, 2010 8:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IVR How can I match any_num_of_digits#any_num_of_digits in an IVR? I want users to be able to type, eg., 123#4567 I tried the following but it hangs up immediately. If I uncomment WaitExten then it hangs up right when the user dials #. As a side question, can I play a background message while using the Read() command? snip Thanks, Vieri -- #1. You will have to make a custom read to handle # since the out-of-the-box read only handles 0-9 and *. #2. No. because the first DTMF will stop the background play -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update the LCD with the callee's name after dialing
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone. Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
ant to verify the header is being added run tcpdump and analyze it
ith Wireshark. I know that Polycom phones have support for this. I
ust put a modified version of the Asterisk 1.6.1 patch into
roduction for 25 Polycom phones, soon to be 150 phones. I changed the
eturn -1 to return 0 so that the call continues even if they
IPCalledRPID args are invalid.
Ryan
--
ust to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
ia: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
ax-Forwards: 70
ontact: sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
o: Callee Name sip:2...@192.168.1.10:5060
rom: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.2: Using hints on multiple parking lots
Hi, How do I specify to which parking lot the hints refer to? For exemple, on 1.4 I use this to see whether a call is parked in 800: exten = 800,hint,park:8...@parkedcalls But on 1.6 I have multiple parking lots working apparently sucessfully. How do I build the hint for parkinglot1 and parkingloit2 so that my phone , which is subscribing to 800, only see parkinglot1 and NOT parkinglot2? I tried the obvious answer exten = 800,hint,park:8...@parkinglot1 but that didnt seem to do anything. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
On Tue, 6 Jul 2010, C.Savinovich wrote: I am writing to you privately... [snip] Doh! Need another cup of coffee? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update the LCD with the callee's name after dialing
On Tue, Jul 6, 2010 at 10:19 AM, unsero...@aol.com wrote:
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)
})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone.
Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.
Ryan
--
Just to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his
client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something
like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
Max-Forwards: 70
Contact:
sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
To: Callee Name sip:2...@192.168.1.10:5060
From: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
Yes that is what the patch addresses. The phones will only display the
name of the called extension if Remote-Party-ID or P-Asserted-Identity
is set.
Ryan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi - alarm which clears itself - Should I care ?
2010/7/6 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Jul 06, 2010 at 12:37:25PM +0200, Olivier wrote: Hi, When reading logs, I can see a couple of lines such as : full.6:[Jun 30 15:53:26] NOTICE[6599] chan_dahdi.c: PRI got event: Alarm (4) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6599] chan_dahdi.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6607] chan_dahdi.c: Alarm cleared on channel 1 full.6:[Jun 30 15:53:32] NOTICE[6607] chan_dahdi.c: Alarm cleared on channel 2 In a 3 BRI spans powered system, this occurred 4 times last week : 1 time for one span, 3 times for another one. Each time, there is a 6 seconds delay between Alarm and Alarm clearance. It should be harmless. The provider took the line down. If outgoing and incoming calls work well, nothing to worry about. OK ! -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi - Which process to swap from Octo to QuadBRI ?
Hi, I'll soon replace a Junghanns OctoBRI with a Junghanns QuadBRI. As both use wcb4xxp driver (dahdi 2.3.0, libpri 1.4.10.2 and asterisk 1.6.1), I'm planning to proceed this way : 1. Edit 2 versions of files /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf (one for each card). 2. Link current system.conf and dahdi-channels.conf to QuadBRI files. 3. Power PC off 4. Swap cards 5. Power PC on Will this succeed ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax_digium and T.38 error correction
On 07/05/2010 09:02 AM, Kristijan Vrban wrote: Hello, i just had some fax abortions because of some packet loss. so i startet to examine in the pcap recording from the res_fax_digium, if the T.38 EC mode redundancy was really used. So i watched into it, and compared it with a t.38 pcap from spandsp (same asterisk setup, but with app_fax) and i see differences in t38.error_recovery (error-recovery: secondary-ifp-packets) With spandsp here are three items, and with res_fax_digium zero items. (t38.secondary_ifp_packets) I this the the t.38 error correction? I ask this questions, because the fax for asterisk admin manual, there are no information about the T.38 error correction, and if i better use Redundancy or FEC. Please contact Digium Support with questions about Fax For Asterisk's operations and features. Thanks. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
Just downloaded PrivateSHELL and it seems to be what everyone is looking for in Putty. It's much better than putty in terms of not being sluggish and scrolling is fine. Plus the window and the text doesn't hurt your eyes. It has One click SFTP as well. So, good bye to WinSCP. I think I found what I need. I just downloaded it's version 3.0 beta and I already love it. Thanks for the suggestion Michael. -Bruce On Tue, Jul 6, 2010 at 8:26 AM, Matt Watson m...@mattgwatson.ca wrote: On Tue, Jun 29, 2010 at 10:39 AM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: I use SecureCRT+FX , and use ansi graphics. Putty is nice w/WinSCP as well. I'll +1 this - SecureCRT+FX is the first thing I got my employer to buy a license of for me when I had to start using Windows on my desktop for other reasons instead of a Linux Distro. I do keep a copy of PuTTY handy on too though, the great thing about putty is that it doesn;t require it to be installed on a desktop, so you can just keep a copy of the executable on a USB flash drive or Windows share that you can then run from any desktop if you happen to be at a computer other than your own. PuTTY works well, but there are some things in it that just drive me absolutely crazy, like right-click in the window is an automatic paste... i have pasted into a putty window by accident more times than I can count, most of the time its from doing a right-click expecting a context menu to get a 'copy' action and then i just end up pasting what i actually wanted to copy :| -- Matt -- Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update the LCD with the callee's name after dialing
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)
})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone.
Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.
Ryan
--
Just to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his
client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something
like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
Max-Forwards: 70
Contact:
sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
To: Callee Name sip:2...@192.168.1.10:5060
From: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
Yes that is what the patch addresses. The phones will only display the
name of the called extension if Remote-Party-ID or P-Asserted-Identity
is set.
Ryan
--
But if the Remote-Party-ID is set or not can only be checked by sniffing with
Wireshark or another sniffer.
It can not be checked by using sip set debug on in Asterisk. Correct?
Because there I cannot see anything added.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with wct4xxp - cannot make calls
Hi, I'm having problems with a TE420P card, in which I cannot make calls using spans 2 through 4. After a couple of days of working correctly, spans 2, 3 and 4 start failing (can not make calls). The system is configured to work with SS7. After the ACM message goes out, immediately a REL message is returned. I searched for error messages in /var/log/messages and could not find a clue about what is going on. I also asked the provider and they tell me that everything looks OK on their side. Then I found out that if I disconnected the lines, the spans started to work correctly. The error persists a system reboot. Asterisk also does not indicate what the trouble is. When I enabled debug for wct4xxp, the following messages started to show: Jul 6 11:15:03 server kernel: wct4xxp: LOF/LFA detected on span 2 but debouncing for 2500 ms Jul 6 11:15:03 server kernel: wct4xxp: LOF/LFA detected on span 4 but debouncing for 2500 ms Jul 6 11:15:03 server kernel: wct4xxp: LOF/LFA detected on span 3 but debouncing for 2500 ms Jul 6 11:15:03 server kernel: wct4xxp: LOF/LFA detected on span 2 but debouncing for 2500 ms Jul 6 11:15:03 server kernel: wct4xxp: LOF/LFA detected on span 2 but debouncing for 2500 ms Jul 6 11:15:03 server kernel: wct4xxp: LOF/LFA detected on span 4 but debouncing for 2500 ms Jul 6 11:15:04 server kernel: wct4xxp: LOF/LFA detected on span 3 but debouncing for 2500 ms Jul 6 11:15:04 server kernel: wct4xxp: LOF/LFA detected on span 2 but debouncing for 2500 ms Jul 6 11:15:04 server kernel: wct4xxp: LOF/LFA detected on span 2 but debouncing for 2500 ms The systems calls a number, then calls another and then does a bridge. A symptom that leads me to the sync problem is that when the call goes into the bridge with spans 2, 3 or 4, I start getting clicks and the call quality goes down. Also of note is that span 1 has always worked correctly, even when bridging is between channels in the same span 1. Another thing that I find odd is the output of dadhi_scan, that indicates that syncsrc=0 for all spans. Is that correct? [1] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 1 name=TE4/0/1 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) (VPMOCT128) location=Board ID Switch 0 basechan=1 totchans=31 irq=177 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) (VPMOCT128) location=Board ID Switch 0 basechan=32 totchans=31 irq=177 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS [3] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 3 name=TE4/0/3 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) (VPMOCT128) location=Board ID Switch 0 basechan=63 totchans=31 irq=177 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS [4] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 4 name=TE4/0/4 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) (VPMOCT128) location=Board ID Switch 0 basechan=94 totchans=31 irq=177 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS dahdi_tool indicates that Sync Source is Internally clocked for all spans. I have installed Rev 8852 of dahdi-complete without success. The system is CentOS 2.6.18-164.15.1.el5PAE. Here is the output of wct4xxp startup and dahdi_cfg: Jul 5 18:00:09 server kernel: dahdi: Telephony Interface Registered on major 196 Jul 5 18:00:09 server kernel: dahdi: Version: 2.3.0.1 Jul 5 18:00:09 server kernel: Found TE4XXP at base address fb7ffc00, remapped to f898ac00 Jul 5 18:00:09 server kernel: DMA memory base of size 2048 at e7eeb000. Read: e7eeb400 and Write e7eeb000 Jul 5 18:00:09 server kernel: TE4XXP version c01a016c, burst ON Jul 5 18:00:09 server kernel: Octasic optimized! Jul 5 18:00:09 server kernel: card 0: FALC framer is v2.1 or earlier. Jul 5 18:00:09 server kernel: FALC version: 0005, Board ID: 00 Jul 5 18:00:09 server kernel: Reg 0: 0x27eeb400 Jul 5 18:00:09 server kernel: Reg 1: 0x27eeb000 Jul 5 18:00:09 server kernel: Reg 2: 0x Jul 5 18:00:09 server kernel: Reg 3: 0x Jul 5 18:00:09 server kernel: Reg 4: 0x0101 Jul 5 18:00:09 server kernel: Reg 5: 0x Jul 5 18:00:09 server kernel: Reg 6: 0xc01a016c Jul 5 18:00:09 server kernel: Reg 7: 0x1f00 Jul 5 18:00:09 server kernel: Reg 8: 0x010200ff Jul 5 18:00:09 server kernel: Reg 9: 0x00fd0001 Jul 5 18:00:10 server kernel: Reg 10: 0x004a Jul 5 18:00:10 server kernel: wct4xxp :06:08.0: Enabled 1sec error counter interrupt Jul 5 18:00:10 server kernel: wct4xxp :06:08.0: Enabled errored second interrupt Jul 5 18:00:10 server kernel: wct4xxp
Re: [asterisk-users] Dahdi - Which process to swap from Octo to QuadBRI ?
On Tue, Jul 06, 2010 at 05:03:04PM +0200, Olivier wrote: Hi, I'll soon replace a Junghanns OctoBRI with a Junghanns QuadBRI. As both use wcb4xxp driver (dahdi 2.3.0, libpri 1.4.10.2 and asterisk 1.6.1), I'm planning to proceed this way : 1. Edit 2 versions of files /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf (one for each card). 2. Link current system.conf and dahdi-channels.conf to QuadBRI files. 3. Power PC off 4. Swap cards 5. Power PC on Will this succeed ? It should. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk: call failed error 408 timeout
Hello every one: I have instaled asterisk in a open suse 10.2 operative system and i try to probe two softphone inside my LAN but i alway receive the same error call failed:error 408 timeout and i don`t have any error in a /var/log/asterisk and any in the CLI i hope your can help me Bye and good look -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update the LCD with the callee's name after dialing
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.
exten =
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)
})
Ryan
If you installed Asterisk from source you just need to patch and
recompile / install.
cd asterisk-version
patch -p1 ../asterisk-verson-called-
rpid.patch
make install
Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.
Ryan
Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.
Ryan
--
Version 1.6.1.20
But it was my individual problem. Installing from scratch solved the
patching issue.
Now the application SIPCalledRPID is active and gets executed but i still
don't get the name of the called person
on my display. Maybe this is client dependent? I am using 3CX Softphone.
Or
is somethins else missing?
The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.
Ryan
--
Just to make sure that we are talking about the same issue.
What I want is that when two users are registered at the same peer that
when user A calls user B user A gets the name of user B displayed on his
client.
Is this what you are trying to fix with the patch?
Because from my understanding as an absolute newbie to SIP and Asterisk, the
header
should already contain the let's call it displayname and look something
like
INVITE sip:2...@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP
192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
Max-Forwards: 70
Contact:
sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
To: Callee Name sip:2...@192.168.1.10:5060
From: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
Yes that is what the patch addresses. The phones will only display the
name of the called extension if Remote-Party-ID or P-Asserted-Identity
is set.
Ryan
--
But if the Remote-Party-ID is set or not can only be checked by sniffing with
Wireshark or another sniffer.
It can not be checked by using sip set debug on in Asterisk. Correct?
Because there I cannot see anything added.
--
I am sorry, my fault. It is added and I can see it in Asterisk sip debug.
But comparing the Remote-Party-ID Header of a (displayed) caller and a (not
displayed) callee looks a bit different.
Remote-Party-ID: Callee
sip:2...@192.168.1.10:5060;party=called;id-type=subscriber;screen=yes
Remote-Party-ID: Caller sip:1...@192.168.1.10;privacy=off;screen=yes
Could maybe this be the reason why it does not work for me?
Sorry if I ask stupid questions but this feature is quite important for me.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 482 Loop Detected
- Original Message -
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients. We run a multi-tenant
server
and have set sip.conf to forward calls to a public context based
on
incoming domain name. This was all working before but not it is
complaining of a loop back as the source and target server are
the
same.
Any ideas on how to overcome this problem as we dial our clients
based
on their email address.
Grabbing a SIP debug I see:
--- Transmitting (no NAT) to 10.172.120.5:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u
To: sip:us...@seconddomain.com
Call-ID: 66b3314cc6d1-jxu0nhluv4zt
CSeq: 2 INVITE
Server: secret
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:us...@172.30.14.8
Content-Length: 0
And am guessing that as the source from IP matches the Contact:
address Asterisk sees that as a loop ?
I don't know these things, but you should probably post more of a
SIP
trace. Maybe turn on full sip debug to a file for long enough to see
what the SIP conversation looks like that asterisk 1.6.2.9 is having
with itself.
From what I have read hairpin calls are not supported by asterisk;
so am guessing something has been fixed in the 1.6.2.X branch that
should have not worked in 1.6.1.X anyway :) While I continue the
research have implemented using a workaround via the AstDB and the
following changes to the uri-dial plan:
exten =
_[a-zA-Z0-9].,n,GotoIf(${DB_EXISTS(URI/${ext...@${sipdomain})}?inturi:exturi)
exten =
_[a-zA-Z0-9].,n(inturi),Goto(${DB(URI/${ext...@${sipdomain})})
exten = _[a-zA-Z0-9].,n(exturi),Macro(uridial,${ext...@${sipdomain})
This is a bit of pain as we have to make sure we update the DB when a
new inbound URI is added; though it works and means we can stick with
the 1.6.2.X branch.
Would be interested to hear from a dev though as to whether they think
it should work as we originally had it configured ?
Do you think this should be raised as a issue in bugtraq or at least brought up
on the asterisk-dev mailing list ?
--
Thanks, Phil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Y-cords - What are they ?
Good Afternoon, Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Y-cords - What are they ?
Tuesday, July 6, 2010, 8:11:46 PM, bruce wrote: Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? I think Y-cords only for PSTN. Or there're Y-cords for twisted pair ethetnet too, but that not a good idea. Usualy VoIP phones includes a mini 2 port switch to use one switch port for a phone and a PC. -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Y-cords - What are they ?
I believe people use this for headsets, to have a superviror listen in on a call with the agent (for training purposes). You can therefore plug in two headsets on the same phone. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Gergo Csibra Sent: Tuesday, July 06, 2010 15:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Y-cords - What are they ? Tuesday, July 6, 2010, 8:11:46 PM, bruce wrote: Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? I think Y-cords only for PSTN. Or there're Y-cords for twisted pair ethetnet too, but that not a good idea. Usualy VoIP phones includes a mini 2 port switch to use one switch port for a phone and a PC. -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't dial out through AMI
SIP user = Asterisk 1.6 server = SIP Trunk = external destination:
works
AMI script = Asterisk 1.6 server = SIP Trunk = external destination:
Failed to authenticate on INVITE to 'asterisk
sip:asterisk@(ipaddr);tag=alphanumeric'
I¹ve tried doing things like ³include = contextwithtrunk in various
places, googling, re-reading relevant portions of the largish O'Reilly
Asterisk book, no avail.
The call will go through to a registered SIP user just fine, but won't seem
to go out off the trunk.
Here's the basic set of commands being sent through AMI:
Action: Originate
Channel: SIP/ShoreTel
Variable: Data=teletubbie-murder
Context: accept
priority: 1
Number: (external number reachable from regular SIP user account)
Here's the accept context:
[accept]
include = incoming
include = outbound-pbx
exten = s,1,Answer
exten = s,n,Playback(custom/msg1)
exten = s,n,Background(custom/how-to-ack)
exten = s,n,WaitExten(5,m)
exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))
exten = 1,n,Background(${Data})
exten = 1,n,Background(discon-or-out-of-service)
exten = 1,n,WaitExten(5,m)
exten = 1,n,Hangup
exten = 2,1,Background(de-activated)
exten = 2,n,ForkCDR(v,s(reject=${Data}))
exten = 2,n,Hangup
exten = 3,1,Goto(accept,1,2)
exten = *,1,Goto(accept,s,1)
exten = i,1,Goto(accept,s,1)
exten = t,1,Goto(accept,s,1)
Obviously, I'm playing around with the context a bit but for now just want
to get the outbound call working.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Y-cords - What are they ?
We deal with Y-cords all the time for Ethernet and BRIs. They are just normal cords, making use of the fact that both Cat5 networks and BRI ports don't use all the 8 pins, so why not use extra wires in the cable for something useful instead of wasting them. It has nothing to do with the performance, and the cables are provided by reputable manufacturers like Aculab and Sangoma, because some of their equipment have no choice but to use these cables. For example Sangoma's BRI cards use two BRI channels per one physical port, so you need one end of the cable with 8 pins and split it into two on the other end with 4 pins each. Same is the case on Ehernet ports on the Aculab's Groomer II equipment. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-06 4:00 PM, Gergo Csibra csi...@gmail.com wrote: Tuesday, July 6, 2010, 8:11:46 PM, bruce wrote: Can someone please explain what Y-cords are avail... I think Y-cords only for PSTN. Or there're Y-cords for twisted pair ethetnet too, but that not a good idea. Usualy VoIP phones includes a mini 2 port switch to use one switch port for a phone and a PC. -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf User vs Username
Hi In sip.conf, you generally have something like [name] .. username= secret= What is the difference between the name specified in brackets and the username key ? What the sip client should provide ? What do we use in dialplan when trying to reach this client ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't dial out through AMI
On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely mike...@amyskitchen.net wrote: Obviously, I'm playing around with the context a bit but for now just want to get the outbound call working. debug log would be helpful: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf User vs Username
On Tue, Jul 6, 2010 at 7:11 PM, Ruddy Gbaguidi plugwo...@micnes.com wrote: What is the difference between the name specified in brackets and the username key ? Context and username. What the sip client should provide ? The client will tell you their settings What do we use in dialplan when trying to reach this client ? Dial(SIP/Context) This is all documented in sip.conf, otherwise the book (http://astbook.asteriskdocs.org/). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't dial out through AMI
Log attached. It looks like the call is trying to do an invite to the sip trunk and fails there - it never actually tries to send the destination to the ShoreTel system on the other end of the trunk. Here's the ShoreTel context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no On 7/6/10 4:21 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely mike...@amyskitchen.net wrote: Obviously, I'm playing around with the context a bit but for now just want to get the outbound call working. debug log would be helpful: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.t xt nosiptrunk.txt Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't dial out through AMI
On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote: Log attached. --- SIP read from UDP:10.10.10.16:5060 --- SIP/2.0 401 Unauthorized context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no Your context is not setup properly for outbound, you have no credentials defined. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't dial out through AMI
-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Paul Belanger Sent: Tue 7/6/2010 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] Can't dial out through AMI On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote: Log attached. --- SIP read from UDP:10.10.10.16:5060 --- SIP/2.0 401 Unauthorized context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no Your context is not setup properly for outbound, you have no credentials defined. None needed on the ShoreTel side and as I mentioned before regular SIP users can dial out through the Asterisk box using this trunk. Keep in mind, this is a development system on a tightly-controlled network, and I'm trying to start with the simplest case possible, which includes no digest auth for the trunk connection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Externnotify on pollmailboxes=yes
Not if a voicemail is left by another system, but if a voicemail is deleted by an external system (ie. web interface). But externnotify is only run upon voicemail() or voicemailmain()? What is the purpose of pollmailboxes=yes then? -Eric From: tles...@digium.com To: asterisk-users@lists.digium.com Date: Mon, 5 Jul 2010 23:31:01 -0500 Subject: Re: [asterisk-users] Externnotify on pollmailboxes=yes On Monday 05 July 2010 19:17:00 Eric Hiller wrote: Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks for any help, The externnotify script is only run when voicemail is left through the Voicemail application. I'm not sure if you're leaving voicemail messages through an external app or if you're expecting the script to be run when the count changes, but it's only run in that single case. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to work Asterisk with Video Conference
Hi, How do I configure Asterisk as a Video Conference purpose. What package I need to configure and what steps I need to follow to configure in dial-plan to authenticate user. Regards, Hiren Mistry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
