Re: [asterisk-users] Does 'file' command work with asterisk genereted alaw file
On Sun, 2010-04-25 at 23:39 -0500, Tilghman Lesher wrote: On Sunday 25 April 2010 23:22:21 Pham Quy wrote: I record an alaw file by asterisk's record monitor command, and i use linux's file command to check it information. file command recognized the alaw file as DATA, is it correct? The file command works by recognizing certain header data in a file. As the alaw format has no header, that command won't recognize the data as any particular format, so it tells you DATA as its fallback. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org I've just tried to config my softphone (twinkle) with different codecs. i found out that if I remove all the active codecs but G.711 A-law, then the .alaw-output file can be recognized by 'file' command as following: #file 983006584-20100426-142120.alaw 983006584-20100426-142120.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz - but after i changed the active codec to the others, the output is recognize as DATA again. Does the .alaw-output internal codec (or whatever it is) depends on the codec used by softphone? and if i choose different softphone codec, i will get different output file? Thanks in advance Quyps -- _ -- 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] Detect if a Number is up or not
Thank you Zhang Shukun, I was wondering if it is possible to make one or ring and then stop the call. But i don't find a way for that. So i am doing it like .. make a call on accept wait and then hangup. On Tue, Apr 20, 2010 at 12:34 PM, Zhang Shukun bit...@gmail.com wrote: Dial() will return Dialstatus , if the number dialed is busy or off now. use this application you can detect a number is busy or not in several seconds. i use this method in my dialplan. 2010/4/19 ABBAS SHAKEEL shakeel.abbas@gmail.com: Hello Community, I Want to detect if a cell number is ON or OFF... for that matter i can generate call to it using PSTN lines (configured with asterisk). The problem is that i only want to see if the cell number can receive a ring or not. If ring is recieved at called number end then mark it as ON in database... Dial application or Originate action might not be that helpful. Do you have any idea regarding this.. -- 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 -- Best regards, Sucan -- _ -- 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 -- 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
[asterisk-users] play a sound from the callee before putting it in connection.
Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/11, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -- _ -- 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] play a sound from the callee before putting it in connection.
Look at option A(x) on this page:- A(x): Play an announcement (x.gsm) to the called party. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Dial(SIP/11,mA(soundfile)) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mickael MONSIEUR Sent: 26 April 2010 11:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: t...@zilok.com Subject: [asterisk-users] play a sound from the callee before putting it in connection. Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/11, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -- _ -- 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] play a sound from the callee before putting it in connection.
Perfect! Thank you! Dan Journo a écrit : Look at option A(x) on this page:- *A(*/x/*)*: Play an announcement (/x/.gsm) to the called party. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Dial(SIP/11,mA(soundfile)) *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mickael MONSIEUR *Sent:* 26 April 2010 11:22 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* t...@zilok.com *Subject:* [asterisk-users] play a sound from the callee before putting it in connection. Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/11, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -- _ -- 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] Detect if a Number is up or not
AMI writes event Ringing..., you can catch it and (via the same AMI) send a soft hangup request. On Mon, Apr 26, 2010 at 12:54 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Thank you Zhang Shukun, I was wondering if it is possible to make one or ring and then stop the call. But i don't find a way for that. So i am doing it like .. make a call on accept wait and then hangup. On Tue, Apr 20, 2010 at 12:34 PM, Zhang Shukun bit...@gmail.com wrote: Dial() will return Dialstatus , if the number dialed is busy or off now. use this application you can detect a number is busy or not in several seconds. i use this method in my dialplan. 2010/4/19 ABBAS SHAKEEL shakeel.abbas@gmail.com: Hello Community, I Want to detect if a cell number is ON or OFF... for that matter i can generate call to it using PSTN lines (configured with asterisk). The problem is that i only want to see if the cell number can receive a ring or not. If ring is recieved at called number end then mark it as ON in database... Dial application or Originate action might not be that helpful. Do you have any idea regarding this.. -- 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 -- Best regards, Sucan -- _ -- 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 -- 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 -- _ -- 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] 1.6.1.18 : app_voicemail is calling sendmail without any argument
Hi, I'm banging my head on this : chmod +x /etc/asterisk/mysendmail.sh cat /etc/asterisk/mysendmail.sh #!/bin/sh logger Entering $0 with arguments $* logger $(whoami) exit 0 cd /usr/sbin ln -s /etc/asterisk/mysendmail.sh sendmail tail /etc/asterisk/voicemail.conf ... attach=yes ... [default] 7790 = 1234,FooBar,f...@example.com Whenever a voicemail is received, I can see in /var/log/syslog: Apr 26 13:59:00 foo-dev logger: Entering /usr/sbin/sendmail with arguments -t Apr 26 13:59:00 foo-dev logger: asterisk My understanding is that asterisk should have passed at least 2 values to /usr/sbin/sendmail : - one naming email's recipient (here f...@example.com) - one naming the attached file So I think I should have seen something like : Entering /usr/sbin/sendmail with arguments -t f...@example.com -a msg001.WAV Is this correct or am I missing something obvious ? 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] hardware clock drift and CDR
--- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? 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] Detect if a Number is up or not
Thanks Motiejus Jakstsys Thank you for the value able info i will give it a try. 2010/4/26 Motiejus Jakštys desired@gmail.com AMI writes event Ringing..., you can catch it and (via the same AMI) send a soft hangup request. On Mon, Apr 26, 2010 at 12:54 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Thank you Zhang Shukun, I was wondering if it is possible to make one or ring and then stop the call. But i don't find a way for that. So i am doing it like .. make a call on accept wait and then hangup. On Tue, Apr 20, 2010 at 12:34 PM, Zhang Shukun bit...@gmail.com wrote: Dial() will return Dialstatus , if the number dialed is busy or off now. use this application you can detect a number is busy or not in several seconds. i use this method in my dialplan. 2010/4/19 ABBAS SHAKEEL shakeel.abbas@gmail.com: Hello Community, I Want to detect if a cell number is ON or OFF... for that matter i can generate call to it using PSTN lines (configured with asterisk). The problem is that i only want to see if the cell number can receive a ring or not. If ring is recieved at called number end then mark it as ON in database... Dial application or Originate action might not be that helpful. Do you have any idea regarding this.. -- 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 -- Best regards, Sucan -- _ -- 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 -- 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 -- _ -- 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 -- 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] hardware clock drift and CDR
On 4/26/2010 7:33 AM, Vieri wrote: --- On Sun, 4/25/10, Gordon Hendersongordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? Vieri If it is NTP that you are worried about, you can see what your servers look like by doing an ntpq -p which should show you the clocks in the pool, which ones it is using etc. Example: remote refid st t when poll reach delay offset jitter == *clock.trit.net 192.12.19.20 2 u 385 512 377 50.2203.094 0.558 +blue.nonexiste. 91.189.94.4 3 u 339 512 377 49.154 -16.663 4.596 +216.45.57.38216.218.254.202 2 u 155 512 377 50.2381.419 0.481 With my system synchronized to clock.trit.net. That is off my master clock, and everything else is synced to it by +/- 1 second. To fix this the easiest way, that I have seen at least, stop ntpd, do an ntpdate to your primary chosen clock (ntpdate clock.trit.net in my example) and restart ntpd and verify that your clock is sync'ed accurately. Also verify that it isn't hitting your hardware dummy clock in ntpd.conf, and if it is, and you can't force it out, you can remove it temporarily. Your CDR's will be screwy in terms of timestamps based on the system time constantly changing, as well as your log files being slightly off, and if you are doing anything remote to another box in terms of logging or database, it will be even more screwy. ~Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] 1.6.1.18 : app_voicemail is calling sendmail without any argument
On Monday 26 April 2010 07:22:33 Olivier wrote: My understanding is that asterisk should have passed at least 2 values to /usr/sbin/sendmail : - one naming email's recipient (here f...@example.com) - one naming the attached file So I think I should have seen something like : Entering /usr/sbin/sendmail with arguments -t f...@example.com -a msg001.WAV Is this correct or am I missing something obvious ? Your understanding is incorrect. The voicemail application encodes the entire email message and invokes sendmail to send the message. There is no re-encoding expected from the sendmail app. -- 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
[asterisk-users] [PATCH] Make Queue announcements more consistent (1.4.26.2)
Hi,
After playing around with queues a bunch on 1.4.26.2, I noticed a few things,
which the patch below addresses. It addresses:
- Callers in position 0 will hear periodic/position announcements at a
very different rate than all other callers.
-- Announcements while in position 0 could be delayed up to
timeout+retry seconds.
-- This patch reduces that possible delay to only timeout seconds
- The say_position and periodic_announcement times are in elapsed time
that _includes_ the
time of the announcement.
-- This patch changes those times to be the time _between_ playing
of those announcements
Thanks.
-- James
--- asterisk-1.4.26.2/apps/app_queue.c 2009-08-10 13:14:34.0 -0700
+++ asterisk-1.4.26.2.new/apps/app_queue.c 2010-04-25 22:25:08.0
-0700
@@ -345,6 +345,7 @@
time_t last_periodic_announce_time; /*! The last time we played a
periodic announcement */
int last_periodic_announce_sound; /*! The last periodic
announcement we made */
time_t last_pos;/*! Last time we told the user
their position */
+ time_t last_ring_time; /*! Last time we tried to ring
the agents */
int opos; /*! Where we started in the queue
*/
int handled;/*! Whether our call was handled */
int pending;/*! Non-zero if we are
attempting to call a member */
@@ -1653,6 +1654,7 @@
res = 0;
/* Set our last_pos indicators */
+ time(now);
qe-last_pos = now;
qe-last_pos_said = qe-pos;
@@ -2131,6 +2133,8 @@
if (!res)
ast_moh_start(qe-chan, qe-moh, NULL);
+ /* Refresh now so that frequency is time _between_ recordings */
+ time(now);
/* update last_periodic_announce_time */
qe-last_periodic_announce_time = now;
@@ -3292,7 +3296,8 @@
static int wait_a_bit(struct queue_ent *qe)
{
/* Don't need to hold the lock while we setup the outgoing calls */
- int retrywait = qe-parent-retry * 1000;
+ //int retrywait = qe-parent-retry * 1000;
+ int retrywait = RECHECK * 1000;
int res = ast_waitfordigit(qe-chan, retrywait);
if (res 0 !valid_exit(qe, res))
@@ -4003,6 +4008,7 @@
qe.max_penalty = max_penalty;
qe.last_pos_said = 0;
qe.last_pos = 0;
+ qe.last_ring_time = 0;
qe.last_periodic_announce_time = time(NULL);
qe.last_periodic_announce_sound = 0;
qe.valid_digits = 0;
@@ -4074,9 +4080,12 @@
break;
}
/* Try calling all queue members for 'timeout' seconds
*/
- res = try_calling(qe, args.options,
args.announceoverride,
args.url, tries, noption, args.agi);
- if (res)
- goto stop;
+ if ((time(NULL) - qe.last_ring_time)
qe.parent-retry) {
+ res = try_calling(qe, args.options,
args.announceoverride,
args.url, tries, noption, args.agi);
+ qe.last_ring_time = time(NULL);
+ if (res)
+ goto stop;
+ }
stat = get_member_status(qe.parent, qe.max_penalty);
@@ -4125,7 +4134,7 @@
/* If using dynamic realtime members, we should
regenerate the
member list for this queue */
update_realtime_members(qe.parent);
- /* OK, we didn't get anybody; wait for 'retry' seconds;
may get a
digit to exit with */
+ /* OK, we didn't get anybody; poll our retry */
res = wait_a_bit(qe);
if (res)
goto stop;
--
_
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[asterisk-users] misdn accountcode?
Hello asterisk users, I am having quite a problem finding, in the misdn.conf file, the accountcode variable. In sip and dahdi the variable name is account code, is there any kind of variable to set this property in misdn. Thanks in advance, Alex -- _ -- 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 - Pickup and SIP Replaces header
Hello,
I'm using Thomson/Technicolor ST2030S hardphones with Asterisk 1.6.
Changing from 1.6.1.18 to 1.6.2.6, I can see a change in Pickup's behaviour
and I'm a bit confused about it.
With 1.6.2.6, when extension 7791 is calling extension 7792, I can see
INVITE messages coming in and out Asterisk.
I can also see a NOTIFY message advertising this call to subscriber 7793,
for instance.
Here is an extract of this message :
NOTIFY sip:7...@192.168.101.102:5060;user=phone SIP/2.0
snip
Call-ID: 7019-c0a80101-...@192.168.101.102
snip
Content-Length: 212
From then, if BLF 7792 on extension 7793 is pressed, then an INVITE message
is send with :
INVITE sip:*87...@192.168.101.240:5060;user=phone SIP/2.0
snip
Replaces: pickup-9582-c0a80101-...@192.168.101.102
snip
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
This INVITE fails with :
snip
chan_sip.c: Trying to pick up 7...@subs
snip
app_directed_pickup.c: No target channel found for 7792.
If I'm dialing *87792 instead of using BLF, then I'm entering the dialplan
part in which there is Pickup(${EXTEN:2...@pickupmark) and the call is
correctly pickup.
So my understanding is :
when upgrading from 1.6.1 to 1.6.2, Asterisk must somehow advertise a newly
supported SIP capability which is now used by ST2030S hardphones to build
Pickup requests.
My question is :
- is my understanding correct ?
- if positive, is there a way to tame asterisk to behave appropropriately ?
Regards
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Re: [asterisk-users] 1.6.2 - Pickup and SIP Replaces header
2010/4/26 Olivier oza_4...@yahoo.fr
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
I took a look at chan_sip.c and read this :
/* RFC3891: Replaces: header for transfer */
{ SIP_OPT_REPLACES, SUPPORTED, replaces },
Should voip-info.org be updated accordingly ?
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[asterisk-users] How to disable dialog-info based call pickups (Was: Re: 1.6.2 - Pickup and SIP Replaces header)
Hello, I searched this list archives and couldn't find any practical way to disable newly introduced dialog-info based call pickups (see CHANGES file). 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] hardware clock drift and CDR
On Mon, 26 Apr 2010, Vieri wrote: --- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? Yes. ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). ntp binds to the ntp port (123) and prevents anything else binding to it, or listening on it - which ntpdate needs to do. Example here: Desktop is running ntpd: yakko:/home/gordon# ps ax | fgrep ntp 22064 ?Ss 0:14 /usr/sbin/ntpd -p /var/run/ntpd.pid -u 106:107 -g 30340 pts/29 R+ 0:00 fgrep ntp I try to run ntpdate: yakko:/home/gordon# ntpdate essen.drogon.net 26 Apr 14:20:47 ntpdate[30341]: the NTP socket is in use, exiting Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. Using pool and your LAN server would be the best way forward - there are pool server avalable for most countries too, so us.pool.ntp.org, uk.pool.ntp.org, and so on. Your /etc/ntp.conf file can be very simple indeed - my workstation one is nothing more than: server essen.drogon.net server uk.pool.ntp.org You can check your servers ntp daemon with: ntpq -c peers and ntpq -c rl The key thing to look for in the 'rl' command is 'stratum'. If it's 16 then it's not synchronised and anything less than 16 is good. yakko:/home/gordon# ntpq -c rl | fgrep stratum processor=i686, system=Linux/2.6.29.2, leap=00, stratum=4, Don't get too hung-up on how close to zero the stratum is. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. Might be easier to read the code ;-) My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? If ntpd can't keep the kernel time in-sync then it will step abput every 900 seconds - which is what appears to be happening here. (the intervals are typically much longer than 20 seconds - e.g. 13:06:30 to 12:21:24 is ~15 minutes - 900 seconds. I don't think I've ever had a server a bad as that before, so have never looked further... Still, it's 2 seconds in 900 seconds, not 2 in 20 as you thought. Which I think is odd - the Linux clock is software derived based on a hardware interrupt - it only consults the hardware battery-backed clock at boot time (and is supposed to write the current time to it at shutdown time) so I wonder if your server is missing interrupts, or otherwise mis-behaving. Is there anything else odd in the log-files? 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
[asterisk-users] SIP authentication
Hi, For IAX there is a fairly clear description of the authentication process for inbound calls. A similar SIP document used to exist on the voip-info wiki, but since 1.6.2 has a number of changes, I was wondering how different (if at-all) 1.6 authentication might be in SIP over 1.2. or 1.4 versions? Can I also assume that for an outbound SIP call, defaultuser is now used instead of username if there is an auth-challenge from the remote party? Or am I completely missing the point? Thank for any pointers. Regards, Steve -- _ -- 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 of registration with Asterisk using exosip2
Hi Everybody,
I try to register to the Asterisk server using exosip2, this is my code :
*TRACE_INITIALIZE (6, stdout);
if (eXosip_init ()) {
printf(eXosip_init failed\n);
exit (1);
}
i = eXosip_listen_addr (IPPROTO_UDP,192.168.14.35, port, AF_INET, 0);
if (i!=0) {
eXosip_quit();
printf(could not initialize transport layer\n);
exit (1);
}
osip_message_t *reg = NULL;
int id;
int j;
eXosip_lock ();
id = eXosip_register_build_initial_register
(sip:3...@192.168.14.35sip%3a3...@192.168.14.35,
sip:192.168.14.36, sip:3...@192.168.15.35:5060, 1800, reg);
if (id 0)
{
eXosip_unlock ();
return -1;
}
j = eXosip_register_send_register (id, reg);
printf(%d\n,j);
eXosip_unlock ();
return j;
*
Unfortunately, the Asterisk server send to me this message :
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.35:5060
;branch=z9hG4bK1366493399;received=192.168.14.35;rport=5060
From: sip:3...@192.168.14.35 sip%3a3...@192.168.14.35;tag=1127882803
To: sip:3...@192.168.14.35 sip%3a3...@192.168.14.35
Call-ID: 796455...@192.168.14.35
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:3...@192.168.14.36 sip%3a3...@192.168.14.36
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.14.35:5060
;branch=z9hG4bK1366493399;received=192.168.14.35;rport=5060
From: sip:3...@192.168.14.35 sip%3a3...@192.168.14.35;tag=1127882803
To: sip:3...@192.168.14.35 sip%3a3...@192.168.14.35;tag=as0f0ad599
Call-ID: 796455...@192.168.14.35
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1929e732
Content-Length: 0
What is wrong in my code ?
Big thanks for your help.
--
IDRISS GHODHBANE
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[asterisk-users] Taqua users out there?
Are there any other Taqua users out there? We have a trunk to a Taqua switch through our ITSP and all outbound calls have the ANI of the primary number on the trunk regardless of what outbound caller-id we generate. This is more than a little annoying, as it interferes with single-number calling, find-me/follow-me, and other features we're using with Asterisk 1.6. Is there anyone with expertise in what tweaking needs to be done on the Taqua side? And is there an open Taqua forum? Thanks, -Philip -- _ -- 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] hardware clock drift and CDR
--- On Mon, 4/26/10, Gordon Henderson gordon+aster...@drogon.net wrote: --- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? Yes. ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). ntp binds to the ntp port (123) and prevents anything else binding to it, or listening on it - which ntpdate needs to do. Example here: Desktop is running ntpd: yakko:/home/gordon# ps ax | fgrep ntp 22064 ? Ss 0:14 /usr/sbin/ntpd -p /var/run/ntpd.pid -u 106:107 -g 30340 pts/29 R+ 0:00 fgrep ntp I try to run ntpdate: yakko:/home/gordon# ntpdate essen.drogon.net 26 Apr 14:20:47 ntpdate[30341]: the NTP socket is in use, exiting Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. Using pool and your LAN server would be the best way forward - there are pool server avalable for most countries too, so us.pool.ntp.org, uk.pool.ntp.org, and so on. Your /etc/ntp.conf file can be very simple indeed - my workstation one is nothing more than: server essen.drogon.net server uk.pool.ntp.org You can check your servers ntp daemon with: ntpq -c peers and ntpq -c rl The key thing to look for in the 'rl' command is 'stratum'. If it's 16 then it's not synchronised and anything less than 16 is good. yakko:/home/gordon# ntpq -c rl | fgrep stratum processor=i686, system=Linux/2.6.29.2, leap=00, stratum=4, Don't get too hung-up on how close to zero the stratum is. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. Might be easier to read the code ;-) My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? If ntpd can't keep the kernel time in-sync then it will step abput every 900 seconds - which is what appears to be happening here. (the intervals are typically much longer than 20 seconds - e.g. 13:06:30 to 12:21:24 is ~15 minutes - 900 seconds. I don't think I've ever had a server a bad as that before, so have never looked further... Still, it's 2 seconds in 900 seconds, not 2 in 20 as you thought. Which I think is odd - the Linux clock is software derived based on a hardware interrupt - it only consults the hardware battery-backed clock at boot time (and is supposed to write the current time to it at shutdown time) so I wonder if your server is missing interrupts, or otherwise mis-behaving. Is there anything else odd in the log-files? I ran the following and it supposedly updated my system time
[asterisk-users] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit
Been trying to get this to go but nongo :-). I'm asking for some guidance especially if I should not be doing this on an RT kernel. I've installed what is supposed to be all of the requred deps. Some factors that may be adding to my problem are: 1. this is only a test.. it's a 32bit guest OS running in VMware under a 64 bit windows host.. (although I'be compiled it on other distros in this config without issues. 2. This is ubuntu Studio which uses an RT (realtime kernel).. There seems to be very little aout there regarding running asterisk on RT linux... one woudl think this would have some benefits.. Big benefits.. I've always wondered. But moreso in a nn-virtual machine environment. Asterisk builds just fine and works. Kernel is : 2.6.31-9-rt (bui...@palmer) (gcc version 4.4.1 (Ubuntu 4.4.1-4ubuntu8) ) #152-Ubuntu SMP PREEMPT RT Thu Oct 15 05:01:14 UTC 2009 Install is Ubuntu Studio 9.10 (Karmic) 32bit and up to date. I have not yet tried it on a stand-alone machine.. I don't think that will be the fix but I will try that soon. -- r...@ubuntu:/usr/src/dahdi-linux-2.3.0# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.3.0/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.3.0/drivers/dahdi/firmware' make -C /lib/modules/2.6.31-9-rt/build SUBDIRS=/usr/src/dahdi-linux-2.3.0/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.3.0/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-headers-2.6.31-9-rt' CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi-base.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o SHIPPED /usr/src/dahdi-linux-2.3.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o LD [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_dynamic.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_dynamic_loc.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_dynamic_eth.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_dynamic_ethmf.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_transcode.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/wctdm.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/wct1xxp.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/wcte11xp.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/wcfxo.o gcc -o /usr/src/dahdi-linux-2.3.0/drivers/dahdi/makefw /usr/src/dahdi-linux-2.3.0/drivers/dahdi/makefw.c /usr/src/dahdi-linux-2.3.0/drivers/dahdi/makefw /usr/src/dahdi-linux-2.3.0/drivers/dahdi/tormenta2.rbt tor2fw /usr/src/dahdi-linux-2.3.0/drivers/dahdi/tor2fw.h Loaded 69900 bytes from file CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/tor2.o /usr/src/dahdi-linux-2.3.0/drivers/dahdi/makefw /usr/src/dahdi-linux-2.3.0/drivers/dahdi/pciradio.rbt radfw /usr/src/dahdi-linux-2.3.0/drivers/dahdi/radfw.h Loaded 42096 bytes from file CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/pciradio.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_echocan_jpah.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_echocan_sec.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_echocan_sec2.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_echocan_kb1.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_echocan_mg2.o LD [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/dahdi_vpmadt032_loader.o CC [M] /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.o /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.c: In function âvoicebus_stopâ: /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.c:980: warning: type defaults to âintâ in declaration of âDECLARE_MUTEXâ /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.c:980: warning: parameter names (without types) in function declaration /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.c:980: error: invalid storage class for function âDECLARE_MUTEXâ /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.c:982: error: âstopâ undeclared (first use in this function) /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.c:982: error: (Each undeclared identifier is reported only once /usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.c:982: error: for each function it appears in.) make[3]: *** [/usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus/voicebus.o] Error 1 make[2]: *** [/usr/src/dahdi-linux-2.3.0/drivers/dahdi/voicebus] Error 2 make[1]: *** [_module_/usr/src/dahdi-linux-2.3.0/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-2.6.31-9-rt' make: *** [modules] Error 2 r...@ubuntu:/usr/src/dahdi-linux-2.3.0# -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. --
Re: [asterisk-users] hardware clock drift and CDR
On Mon, 26 Apr 2010, Vieri wrote: I ran the following and it supposedly updated my system time while ntpd was running: # ps ax | fgrep ntp 1256 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -u ntp:ntp 1623 pts/14 S+ 0:00 fgrep ntp # ntpdate -b -u pool.ntp.org 26 Apr 19:41:18 ntpdate[2791]: step time server 163.117.131.239 offset 0.142263 sec From the ntpdate man page: -u Direct ntpdate to use an unprivileged port for outgoing packets. This is most useful when behind a firewall that blocks incoming traffic to privileged ports, and you want to synchronise with hosts beyond the firewall. Note that the -d option always uses unprivileged ports. So ntpdate does not try and use 123 -- which is in use by ntpd. Does: sudo netstat -a -n -p | grep ntpd show something like: udp0 0 192.168.0.xx:1230.0.0.0:* 1693/ntpd udp0 0 127.0.0.1:123 0.0.0.0:* 1693/ntpd udp0 0 0.0.0.0:123 0.0.0.0:* 1693/ntpd udp6 0 0 fe80::222:68ff:fe36:123 :::* 1693/ntpd udp6 0 0 ::1:123 :::* 1693/ntpd udp6 0 0 :::123 :::* 1693/ntpd unix 2 [ ] DGRAM6635 1693/ntpd -- 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
[asterisk-users] Inbound route question
Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? Thanks a lot. Alejandro -- _ -- 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] Inbound route question
I must be missing something because this sounds REAL simple - just dial 1000, 1001 or 1002 from dialplan or do a Goto to the IVR context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, April 26, 2010 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inbound route question Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? Thanks a lot. Alejandro -- _ -- 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] Inbound route question
But suppose the cell phones DID number is: 11654321 and the GSM Gateway extension has DID number: Which is the DID number I have to use in the inbound route I create to point to the IVR ??? Thanks again. 2010/4/26 Danny Nicholas da...@debsinc.com: I must be missing something because this sounds REAL simple - just dial 1000, 1001 or 1002 from dialplan or do a Goto to the IVR context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, April 26, 2010 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inbound route question Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? Thanks a lot. Alejandro -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] hardware clock drift and CDR
On Mon, 26 Apr 2010, Vieri wrote: I ran the following and it supposedly updated my system time while ntpd was running: # ps ax | fgrep ntp 1256 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -u ntp:ntp 1623 pts/14 S+ 0:00 fgrep ntp # ntpdate -b -u pool.ntp.org 26 Apr 19:41:18 ntpdate[2791]: step time server 163.117.131.239 offset 0.142263 sec Steves posted the reason - the -u flag causes it to bypass the normal ports, and so does work in this instance. By the way, as a side question, on another server I see this: # ntpq -c peers remote refid st t when poll reach delay offset jitter == inf-srv1.hospit .LOCL. 1 u 56 64 3770.314 21755.8 7.634 Not sure what LOCL means but I'll refer to the NTP docs (inf-srv1 is my LAN Windoze time server). It may mean that it's using it's internal clock as the master source. If so, then it's trust it as far as I could spit a rat... Try this: ntpq host inf-srv1 (or it's IP addresS) peers and find out what peers it's using. It's just possible that your server is actually more accurate that your LAN server... Give your server a few more peers and find out - just list pool.ntp.org in the /etc/ntp.conf file a few times (and restart ntpd) Anyway, back to the faulty new server (which reports a stratum of 3 after ntpd has been running for a while and sync'ing to pool.ntp.org): The stratus is just how far it is away from stratum 1 - which is deemed to be synchronised to true time - usually derived from GPS, local atomic clock or MSF type radio. (I used to run an MSF clock synced NTP server for a while) So a host synchronised to a stratum 1 server will be at stratum 2, and hosts synchronised to a stratum 2 server will be at stratum 3. If you synchronise to a mixture, then your host will be somewhere in the range, depending on how good it reckons the other are... it's supposed to be a good motherboard (Asus) but I'm running a relatively old kernel (2.6.23). Googling around suggests me to try to boot with noapic if I keep seeing my clock drift so much. # more /proc/interrupts CPU0 CPU1 CPU2 CPU3 0:103 0 0 1 IO-APIC-edge timer 1: 2151 0 0 9 IO-APIC-edge i8042 4: 12772543 1321793296030647661766 IO-APIC-edge serial That's a rather high number of serial interrupts... Do you have a serial console, or using the serial link with Linux HA? In-general, I like ASUS motherboards though and use them a lot myself. 8: 1 0 1 0 IO-APIC-edge rtc 9: 0 0 0 1 IO-APIC-fasteoi acpi 12: 0 0 0 4 IO-APIC-edge i8042 14: 2234 73664 0 2470 IO-APIC-edge ide0 16: 28322780 51914617 40744985 39615361 IO-APIC-fasteoi eth0 17: 63242610 42157366 43790794 48255583 IO-APIC-fasteoi eth1 18:1348544 0 0 1 IO-APIC-fasteoi eth2 20:9006839824429560765954923525 IO-APIC-fasteoi ahci 21: 162750903 140985080 176469550 166839225 IO-APIC-fasteoi wcte12xp0 22: 16662710 18210608 12053147 12739782 IO-APIC-fasteoi HFC-multi NMI: 0 0 0 0 LOC: 64546905 64546897 64546897 64546897 ERR: 0 MIS: 0 I have 3 PCI cards: 1 PRI, 1 quad BRI, 1 dual ethernet. Could booting with noapic help? Doubt it, but iy's worth a try. Personally, I'd try more NTP hosts first. (Especially knowing you're syncing to a windoze host ;-) What about my PCI devices? Will they be stable even with noapic? The reason I got this new mobo is that the previous hardware froze the system with a kernel crash. In fact, I rsync'ed to this new hardware (so identical system software) and it has been running flawlessly for more than a week now, while it used to crash/freeze once a day (another Asus board, by the way). My only problem now is with the d...@!mned clock... As far as syslog messages, I don't see anything wrong. No errors whatsoever. Thanks for your time. I'll try to boot with noapic and cross my fingers. Good luck.. What may also help is compiling a custom kernel for your hardware - it's what I do by default, but I appreciate that's not for everyone, however it is the best way to make sure you have the kernel tuned exactly to your hardware needs. 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:
[asterisk-users] DTMF from SIP phone to FXS/FXO
Hello, I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my FXS/FXO lines. I am running Asterisk 1.4.21.1 In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly from SIP to FXS, but in the SIP phone I only hear a continuous noise. However, when I press any digit in the pone (FXS) I hear the DTMF tone fine in the SIP phone (the noise goes away for as long as I am pressing the key and I can hear the tone correctly). If I call from SIP phone to SIP phone everything works fine. When I set dtfmode to any other option, I can hear voice in both sides (SIP phone and analog line) but DTMF is not transfered. I remember in the past (about 5 months ago) it worked for me with dtmfmode=inband (even dialing through an FXO line. I have been working on other things and just now came to realize that there is a problem there). I have tried other options in sip.conf and rtp.conf (relaxdtmf, directrtpsetup, dtmftimeout) but none seem to make a difference. Any help is greatly appreciated. ANDRES-- _ -- 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] Inbound route question
Did a little reading on this - looks like your GSM gateway is configured to call Asterisk with second dialtone instead of direct dial to operator. Don't know if changing that would get the DID passed through (beyond my pay grade) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, April 26, 2010 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inbound route question But suppose the cell phones DID number is: 11654321 and the GSM Gateway extension has DID number: Which is the DID number I have to use in the inbound route I create to point to the IVR ??? Thanks again. 2010/4/26 Danny Nicholas da...@debsinc.com: I must be missing something because this sounds REAL simple - just dial 1000, 1001 or 1002 from dialplan or do a Goto to the IVR context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Monday, April 26, 2010 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inbound route question Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? Thanks a lot. Alejandro -- _ -- 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 -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- 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] Building Asterisk-RPM for 1.4.24.1
Hi everybody, quite frequently I build customized RPMs with asterisk-1.4.20.1 including some special patches for it, to install the on CentOS 5. Now I was looking to upgrade to asterisk-1.4.24.1, but the RPM-build is not working anymore with my build environement. In version 1.4.22 the Makefile was modified and all the RPM-stuff was removed, same for the asterisc.spec-file in the redhat-directory. Why the developers removed this and what is the correct way to reenable this for newer versions (including 1.4.24.1 and 1.4.30)? Thanks a lot, -- Chau y hasta luego, Thorolf -- _ -- 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] Building Asterisk-RPM for 1.4.24.1
Probably (JIMO) had something to do with the Zaptel-to-DAHDI switch at 1.4.22.X -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorolf Godawa Sent: Monday, April 26, 2010 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Building Asterisk-RPM for 1.4.24.1 Hi everybody, quite frequently I build customized RPMs with asterisk-1.4.20.1 including some special patches for it, to install the on CentOS 5. Now I was looking to upgrade to asterisk-1.4.24.1, but the RPM-build is not working anymore with my build environement. In version 1.4.22 the Makefile was modified and all the RPM-stuff was removed, same for the asterisc.spec-file in the redhat-directory. Why the developers removed this and what is the correct way to reenable this for newer versions (including 1.4.24.1 and 1.4.30)? Thanks a lot, -- Chau y hasta luego, Thorolf -- _ -- 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] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit
Steve Gladden wrote: 2. This is ubuntu Studio which uses an RT (realtime kernel).. There seems to be very little aout there regarding running asterisk on RT linux... one woudl think this would have some benefits.. Big benefits.. I've always wondered. But moreso in a nn-virtual machine environment. RT kernels don't have traditional mutexes, which are used in various places in DAHDI for Linux. To my knowledge nobody has done the work to update the drivers to be able to use the RT kernel replacement synchronization mechanisms when compiled against an RT kernel. -- 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] [PATCH] Make Queue announcements more consistent (1.4.26.2)
On 27/04/10 2:21 AM, James Lamanna wrote: Hi, After playing around with queues a bunch on 1.4.26.2, I noticed a few things, which the patch below addresses. It addresses: - Callers in position 0 will hear periodic/position announcements at a very different rate than all other callers. -- Announcements while in position 0 could be delayed up to timeout+retry seconds. -- This patch reduces that possible delay to only timeout seconds - The say_position and periodic_announcement times are in elapsed time that _includes_ the time of the announcement. -- This patch changes those times to be the time _between_ playing of those announcements Please post this to issues.asterisk.org. Unfortunately developers are unable to look at or add patches without knowing the license. When you create an account on issues.asterisk.org you can file a disclaimer for the code and then the patch can be added to the base Asterisk install (assuming it meets coding guidelines etc). -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Supporting addressing formats and unsolicited Notify
Hi All, I am using Asterisk as my pbx talking to a main proxy server. The main Proxy server is sending unsolicited Notify Messages to the clients after a call is established. Is there a setting that I can tell Astersik to forward any NTY received from Proxy to be forwarded to the End users? 2) Also Can I have the translation from u...@domain to u...@domain. ( right Now I am having dail plan as numbers) Now I want to support soft clients also whose User names are not numbers only but also Alpha numeric character, I am looking something like: b...@asterisk.com to be mapped to b...@external.com what is the config changes :-) TIA. -- _ -- 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 will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit
Thanks for responding.. So that explains why it won't compile eh? And wow Kevin... I'm curious how much work would it be and would it be worth it? I've always imagined RT kernels would be excellent for asterisk. I've also wondered why it appears not to have been done 'out there' Or discussed very much. -Steve Steve Gladden wrote: 2. This is ubuntu Studio which uses an RT (realtime kernel).. There seems to be very little aout there regarding running asterisk on RT linux... one woudl think this would have some benefits.. Big benefits.. I've always wondered. But moreso in a nn-virtual machine environment. RT kernels don't have traditional mutexes, which are used in various places in DAHDI for Linux. To my knowledge nobody has done the work to update the drivers to be able to use the RT kernel replacement synchronization mechanisms when compiled against an RT kernel. -- 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- 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] 1.6.2 - Pickup and SIP Replaces header
On Mon, Apr 26, 2010 at 11:33 AM, Olivier oza_4...@yahoo.fr wrote:
2010/4/26 Olivier oza_4...@yahoo.fr
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
I took a look at chan_sip.c and read this :
/* RFC3891: Replaces: header for transfer */
{ SIP_OPT_REPLACES, SUPPORTED, replaces },
Should voip-info.org be updated accordingly ?
Go for it. It's easy to make an account there and make the change. I
started the documentation for ConfBridge() a few months ago when I
first found it interesting.
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Re: [asterisk-users] Jitter Buffer and MeetMe.
On Sun, Apr 25, 2010 at 7:13 AM, russian qwerty russian.qwe...@gmail.com wrote: Hello, David. Thank you for reply. But my problem is certainly in the size of JitterBuffer of chan_local. I realy need to know how to change the size of JB (reduce). BTW: 1. The file /etc/asterisk/dsp.conf doesn't exist in my Asterisk 1.6.0.6 (something wrong?). It will take defaults if it does not exist. Since you want to over-ride the defaults, you should make one and put in the suggested values. If you want to see the sample one, you can download source and do 'make samples', but I don't recommend doing that with your production system. 2. VAD is already disable for all trunks. Good, but you should also disable vad within asterisk. 3. And 'talker optimization' future is already disable by patch for app_meetme.c. Good, but make sure also not invoking it with any special flags to meetme. Original 1.6.0 series was forced on, then it became optional, and possible to invoke with flags. Also, 1.6.0.6 is rather old at this point. I no longer remember the improvements, if any, that may affect your situation, but you could check out meetme in changelog for latest 1.6.0 series. -- _ -- 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] Redone setup, bizare problems
Hi ! Sorry if this is a long post... I had this setup for about a year without problems : Network A - wrv200 - internet - wrv200 - net b The 2 networks are linked with an ipsec vpn. The 2 internet connections are with the same cable company to minimize latency, both separates /24 subnets. On network A, I got 2 computers, a single sip phone (aastra 9112i). On network B, I got a linux box acting as smb domain controler and many other thins, another linux box that only do an asterisk server, 4 computers, 3 sip phones (2 aastra 9112i and one 480i ct), 1 analog ATA for the alarm system, an internet point of sale terminal, that's about it. The asterisk server is connected to the PSTN with a VOIP provider trough IAX. All of this was working fine. Since the last couple of months, we had some qos problems with the exterior. The sound was choping up randomly. With what I can do with the linksys wrv200, I was limited. I also had some filesystem issues that made the asterisk server locks up from time to time. So I wanted to reformat the server and re-install it on a ssd drive. So for the time of the re-format, I installed asterisk on my smb server, and re-copied /etc/asterisk, /var/spool/asterisk and /var/lib/asterisk to it. When I arrived at home, the externel power supply of the little shuttle box refused to power up. So I ended up building a new computer from scratch. This new server will be acting as asterisk server and router, replacing the wrv200 from network b (demoting it to wireless access point and switch). Last sunday, I installed the server, re-copying the 3 folders of asterisk in the same maner. I then had a hard time making the phones register to the server. I always had no service with the mwi light steady on. I finaly got the phones to register, I'm still not sure exactly what I did to make it work. And the phone on network A didn't work wither. I'll get to that one later. The linksys pap2t ata didn't had the problem, it registered right as I started the server. On the phones on network B, since then I get the MWI light come on momentarely, with the no service on the display, and then all comes back to normal. But all phones can make and receive calls. For the remote phone, for those familiar with ipsec vpns, it's a net to net connection. So, the gateway cannot reach the computers on the network on the other side. So I had to add a static route on my router/asterisk server to be able to reach the phone on the other side. It was able to register for some time, but the next morning it was no service, and I wasn't able to make it work again. I ended up connecting the phone trough the externel interface of the router/asterisk server. So, my questions : Why is the phones are constatly showing no service with the light flashing ? Is there a way for the remote phone from net A to connect properly trough the vpn ? Thanks for any hints, Niolas -- _ -- 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
