[asterisk-users] Very strange registration problem
Hello list, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's not a firewall problem as all register to port 5060 and the range 5060 -- 5064 is open. It's just very strange that some can register and other not. Any input/suggestions ?! 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] one for your filters
On Wed, 23 Jun 2010, Steve Edwards wrote: On Wed, 23 Jun 2010, Gordon Henderson wrote: Ouch. 82.0.0.0/8 is on my block list, available at: http://www.sedwards.com/class-a-block-list If you don't need to receive packets from far away places, it's a great start. I'd like to have a look, but can't - I think there may be issues with your registrar for your domain - from where I am, there are no glue records for the nameservers, therefore I can't look it up... Looks like it was last edited just over 4 weeks ago, so maybe some caches are starting to time-out... From whois: Domain servers in listed order: DOMAIN0.SEDWARDS.COM DOMAIN1.SEDWARDS.COM You need to supply the IP address of the nameservers (the glue records) if they're inside your own domain... I think I have the name servers configured correctly. I think you were having difficulty because I was blocking everything from 195.0.0.0/8 Please try again. I have and get the same results. DNS glue records are held by the registrar on the gTLD name servers, not your own servers - so (even though I can't access them), I should be able to see the IP addresses for your 2 name servers (DOMAIN[01].SEDWARDS.COM). The output of 'whois' should provide me with those IP addresses, but it's not. See: http://en.wikipedia.org/wiki/Domain_Name_System#Circular_dependencies_and_glue_records E.g. do a whois on my domain, drogon.net and you'll see ns1.drogon.net195.10.225.68 which indicates the glue record is in-place for ns1.drogon.net - the glue is needed because otherwise no-one would be able to find ns1.drogon.net unless they already knew it's IP address - which they won't without the glue in the gTLD servers. Same for your nameservers - no-one can find domain0.sedwards.com unless they know it's IP address, and they can't find that IP address because they don't know the IP address of your nameservers - a circular dependancy that can only be broken by providing the IP address as glue in the gTLD server. This are probably working for some people right now because of caching going on - I suspect you made a change just over 4 weeks ago and that's a typical cache-time out for a lot of systems. Your site is going to drop off the Internet fairly soon unless you get the glue records in-place. And I wasn't accessing from 195/8, but from 81/8. (Although I've tried from both places) Your filtering is far to wide-spread - you can't invite people to view things when you're blocking off a third of the Internet - including most of Europe. Well, you can, but then people are just going to whinge. That's as bad as what Earthlink or was it Verizon did a while back when they decided to reject all email from Europe on the flawed basis that more spam comes from Europe than the US. (It doesn't) 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] I look ARI (Asterisk Recording Interface)
Hello Bruce, This module is not reliable on FreePBX? You know if there is a open source web-voicemail for Asterisk? Best regards, Mickael. 2010/6/23 bruce bruce bruceb...@gmail.com It's one of the bad modules that goes with FreePBX anyhow. The moment you go over 3000 recordings you are already in trouble. It's about time someone come up with a better moduel. On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur mickael.monsi...@gmail.com wrote: Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, 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 -- _ -- 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] realtime queues membername problem
I'm just about to start experimenting with realtime queues, so can't
offer anything from my own experience, but what happens if instead of
updating the existing row, you delete it and insert a new one for the new
user ?
ie. DELETE FROM queue_member_table WHERE id=1;
INSERT INTO
queue_member_table (membername,queue_name,interface,penalty,paused) VALUES
('OtherUser','in_pruebas','SIP/1336','1','0');
Rob
On Wed, 23 Jun 2010
12:37:47 -0600, Jean Chassoul wrote: On Wed, Jun 23, 2010 at 1:57 AM,
Tiago Geada wrote:
to re-read peers from realtime db try: sip prune
realtime all
Hi, I don't have a problem with peers or realtime sip! the
problem is with realtime queues I obviously have a queue_member_table for
INSERT/UPDATE/DELETE users to any queue on my callcenter system, I have for
example this row in queue members: | id | membername | queue_name |
interface | penalty | paused | | 1 | chassoul | in_pruebas | SIP/1336 | 1 |
0 | I can see everything is fine on in_pruebas queue with the *CLI
*CLI queue show in_pruebas in_pruebas has 0 calls (max unlimited) in
'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s
Members: chassoul (SIP/1336) with penalty 1 (realtime) (Not in use) has
taken no calls yet No CallersThe problem is when i need to disconect a
user and connect another one with the same interface or simple update the
membername field on any row, for example: mysql update
queue_member_table set membername = 'OtherUser' where membername =
'chassoul'; Query OK, 1 row affected (0.00 sec) Rows matched: 1 Changed: 1
Warnings: 0 mysql select * from queue_member_table where queue_name =
'in_pruebas'; | id | membername | queue_name | interface | penalty |
paused | | 1 | OtherUser | in_pruebas | SIP/1336 | 1 | 0 |
+--+++---+-++ 1 row in
set (0.00 sec) Asterisk don't read this change and don't update the new
information on the queue, I still got the old member instead of OtherUser
member. :( *CLI queue show in_pruebas in_pruebas has 0 calls (max
unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0%
within 0s Members: chassoul (SIP/1336) with penalty 1 (realtime) (Not in
use) has taken no calls yet No Callers Any ideas?
Links:
--
[1] mailto:tiago.ge...@gmail.com
--
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Re: [asterisk-users] Asterisk + E1 card
On Wed, Jun 23, 2010 at 04:53:27PM -0400, Zeeshan Zakaria wrote: Zaptel and dahdi is the same thing, except the later one is weirdly named to make it harder to pronounce. Don't worry to upgrade to dahdi. But it is not plug and play and you'll need to configure /etc/zaptel.conf and /etc/asterisk/zapata.conf according to your requirement. They are not exactly the same. Consider DAHDI the latest version of Zaptel. That is: some very new devices may actually no longer be supported with it. Some bugs may actually not get fixed there. There are also a number of fixes that made it into the SVN branch Zaptel 1.4, but not into the last release (1.4.12.1). -- 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] Hidden memory leak
On Wed, Jun 23, 2010 at 04:27:20PM -0500, Tilghman Lesher wrote: On Wednesday 23 June 2010 15:45:05 Miguel Molina wrote: Hi all, Anyone know why this happens? Mem:524288k total, 508120k used,16168k free,0k buffers Swap:0k total,0k used,0k free,0k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init 7398 root 18 0 10172 2904 2312 S 0.0 0.6 0:00.21 sshd 9856 root 15 0 4756 1528 1232 S 0.0 0.3 0:00.06 bash 11316 root 15 0 3332 1112 572 S 0.0 0.2 0:01.14 crond 16282 root 25 0 4756 1008 820 S 0.0 0.2 0:00.00 safe_asterisk 22514 root 25 0 494m *445m* 6612 S 0.0 *87.0* 663:08.66 asterisk [snip] Anoyone knows why the memory leak is not shown in the asterisk malloc debug, and how can I figure what's causing it? The asterisk version is 1.6.2.9. If it's not listed in the internal debug, then it's probably a memory leak in one or more of the external libraries linked into Asterisk. That could be anything from openssl to libxml2 (neither likely) to something specifically related to a second or third party module that you've loaded. If so, it should appear in the memory map of the process under that library. Try: pmap 22514 Also try running that command a bit later on and see what has changed. -- 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] Very strange registration problem
Jonas, have you checked the log files? Giorgio Jonas Kellens wrote: Hello list, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's not a firewall problem as all register to port 5060 and the range 5060 -- 5064 is open. It's just very strange that some can register and other not. Any input/suggestions ?! 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] PRI span problem - no D channel
On Tuesday 22 Jun 2010, Mike wrote: Hi, I have the following happen to me after the restart of one of my servers: out of my 3 PRIs (all configured with the same technical settings), the last one isn't coming back. It's underutilized (chances it didn't get a call since my reboot), if it makes a difference . The PRI goes from provisioned to unprovisioned, and I get this regularly: [Jun 22 09:03:48] WARNING[30723]: chan_dahdi.c:2790 pri_find_dchan: No D-channels available! Using Primary channel 72 as D-channel anyway! I was getting something similar and it turned out to be because a *different* span was configured in chan_dahdi.conf, but not connected to anything. Make sure, if you're only using 3 of the 4 spans, that all lines relating to the unused one are commented out in /etc/asterisk/chan_dahdi.conf (and maybe in /etc/dahdi/system.conf as well). I'm guessing this is due to a change in default behaviour between libpri or DAHDI versions (since an older install worked just fine with broken configs). -- AJS -- _ -- 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] Astersik can not detect DTMF key
Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- 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 + E1 card
True, what I meant was they serve the same purpose, i.e. drivers for the non-SIP hardware. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 4:52 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Jun 23, 2010 at 04:53:27PM -0400, Zeeshan Zakaria wrote: Zaptel and dahdi is the same thin... They are not exactly the same. Consider DAHDI the latest version of Zaptel. That is: some very new devices may actually no longer be supported with it. Some bugs may actually not get fixed there. There are also a number of fixes that made it into the SVN branch Zaptel 1.4, but not into the last release (1.4.12.1). -- 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 Pr... -- _ -- 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] Very strange registration problem
Without submitting some logs or your sip settings, how could somebody on this list help you? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:28 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: Jonas, have you checked the log files? Giorgio Jonas Kellens wrote: Hello list, using asterisk 1.4.30 I have the strangest problem that... -- _ -- 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] Very strange registration problem
Giorgio, there is just no registration coming in. SIP debug shows nothing on the SIP peers that do not register. TCPdump shows nothing on incoming registrations. Firewall is down. I'm using SIP realtime and I'm starting to think it is a problem with the MySQL-DB of the MySQL-driver or something... The thing that made it all work again (after reloading, restarting asterisk and the whole CentOS-server) was recompiling asterisk 1.4.30 and addons 1.4.11... I'm still looking for an explanation so I can act quicker the next time this should re-appear. According to you, what logfiles could tell me some more ?? Jonas. On 06/24/2010 12:18 PM, Giorgio Incantalupo wrote: Jonas, have you checked the log files? Giorgio -- _ -- 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] Astersik can not detect DTMF key
What type of phone you are using? It is possible that # is used by this phone as one of its internal functions. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:41 AM, huu giang huugiang...@yahoo.com wrote: Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- 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] Very strange registration problem
If tcpdump not showing any incoming packets from the phones, then the phones are not communicating with the server. There could be no other reason other than iptables blocking that communication. Did you check if iptables is blocking anything? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:55 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Giorgio, there is just no registration coming in. SIP debug shows nothing on the SIP peers that do not register. TCPdump shows nothing on incoming registrations. Firewall is down. I'm using SIP realtime and I'm starting to think it is a problem with the MySQL-DB of the MySQL-driver or something... The thing that made it all work again (after reloading, restarting asterisk and the whole CentOS-server) was recompiling asterisk 1.4.30 and addons 1.4.11... I'm still looking for an explanation so I can act quicker the next time this should re-appear. According to you, what logfiles could tell me some more ?? Jonas. On 06/24/2010 12:18 PM, Giorgio Incantalupo wrote: Jonas, have you checked the log files? ... -- _ -- 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] Friday at 1PM: SIPVicious has a new tool: svcrash
Hi, Got some great news a few days ago from Sandro Gauci (@SandroGauci) and we'll be talking about this with him this Friday at 1PM. SIPVicious, the free security tools for SIP scanning, now include a new tool: svcrash. It is aimed at helping system administrators stop bandwidth consuming scans making use of svwar and svcrack. Here is the announcement on SIPViscious blog: http://blog.sipvicious.org/2010/06/how-to-crash-sipvicious-introducing.html Video demo: http://vimeo.com/12744376 Download link: http://code.google.com/p/sipvicious FAQ for svcrash: http://code.google.com/p/sipvicious/wiki/SvcrashFrequentlyAskedQuestions Any other questions or comments, join us live from 12 noon EDT: http://vuc.me SIP:200...@login.zipdx.com See you there. /r -- _ -- 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] Friday at 1PM: SIPVicious has a new tool: svcrash
Awesome! -- _ -- 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] Very strange registration problem
As I said earlier, the firewall was down. I'm using LFDCSF. If it had anything to do with the firewall, how can the problem be resolved by recompiling asterisk and asterisk-addons ? Yesterday everything went well, this night something happened I guess, and this morning on some locations some phones (of all types of brand) showed a failed registration and rebooting the phones did not help (neither did reloading/restarting asterisk or the entire server)... To make an example : some Snom M3 phones were registered, some Snom 320 were not, on my own Cisco SPA942 2 out of 4 SIP-accounts were registered (the other two would not register after several reboots). It seems as if some SIPaccounts could register and others could not. I don't think a firewall distinguishes between phone brands or SIP accounts. Jonas. On 06/24/2010 01:00 PM, Zeeshan Zakaria wrote: If tcpdump not showing any incoming packets from the phones, then the phones are not communicating with the server. There could be no other reason other than iptables blocking that communication. Did you check if iptables is blocking anything? Zeeshan A Zakaria -- _ -- 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] Very strange registration problem
On 24 Jun 2010, at 12:49, Jonas Kellens wrote: It seems as if some SIPaccounts could register and others could not. I don't think a firewall distinguishes between phone brands or SIP accounts. Alas 'stabbing in the dark' is all we can do until you actually provide some information for us. SIP traces, sip.conf, log extracts etc. Your theory about the MySQL stuff is probably wrong. You'd still see the SIP packets coming to you would you not? 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] PRI span problem - no D channel
I checked, it made sense. But it isnt it :-) The fourth span is commented out, but the third one isnt (and shouldn't be since it's active). Thanks for the tip though, could have easily been this. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Thursday, June 24, 2010 6:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI span problem - no D channel On Tuesday 22 Jun 2010, Mike wrote: Hi, I have the following happen to me after the restart of one of my servers: out of my 3 PRIs (all configured with the same technical settings), the last one isn't coming back. It's underutilized (chances it didn't get a call since my reboot), if it makes a difference . The PRI goes from provisioned to unprovisioned, and I get this regularly: [Jun 22 09:03:48] WARNING[30723]: chan_dahdi.c:2790 pri_find_dchan: No D-channels available! Using Primary channel 72 as D-channel anyway! I was getting something similar and it turned out to be because a *different* span was configured in chan_dahdi.conf, but not connected to anything. Make sure, if you're only using 3 of the 4 spans, that all lines relating to the unused one are commented out in /etc/asterisk/chan_dahdi.conf (and maybe in /etc/dahdi/system.conf as well). I'm guessing this is due to a change in default behaviour between libpri or DAHDI versions (since an older install worked just fine with broken configs). -- AJS -- _ -- 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] Friday at 1PM: SIPVicious has a new tool: svcrash
Maybe we can easily extend the tool to crash Asterisk too (using some exploits non-up2date Asterisk installations) ;-) Am 24.06.2010 13:36, schrieb Randy R: Hi, Got some great news a few days ago from Sandro Gauci (@SandroGauci) and we'll be talking about this with him this Friday at 1PM. SIPVicious, the free security tools for SIP scanning, now include a new tool: svcrash. It is aimed at helping system administrators stop bandwidth consuming scans making use of svwar and svcrack. Here is the announcement on SIPViscious blog: http://blog.sipvicious.org/2010/06/how-to-crash-sipvicious-introducing.html Video demo: http://vimeo.com/12744376 Download link: http://code.google.com/p/sipvicious FAQ for svcrash: http://code.google.com/p/sipvicious/wiki/SvcrashFrequentlyAskedQuestions Any other questions or comments, join us live from 12 noon EDT: http://vuc.me SIP:200...@login.zipdx.com See you there. /r -- _ -- 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] Hidden memory leak
El 24/06/10 05:05, Tzafrir Cohen escribió: On Wed, Jun 23, 2010 at 04:27:20PM -0500, Tilghman Lesher wrote: On Wednesday 23 June 2010 15:45:05 Miguel Molina wrote: Hi all, Anyone know why this happens? Mem:524288k total, 508120k used,16168k free,0k buffers Swap:0k total,0k used,0k free,0k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init 7398 root 18 0 10172 2904 2312 S 0.0 0.6 0:00.21 sshd 9856 root 15 0 4756 1528 1232 S 0.0 0.3 0:00.06 bash 11316 root 15 0 3332 1112 572 S 0.0 0.2 0:01.14 crond 16282 root 25 0 4756 1008 820 S 0.0 0.2 0:00.00 safe_asterisk 22514 root 25 0 494m *445m* 6612 S 0.0 *87.0* 663:08.66 asterisk [snip] Anoyone knows why the memory leak is not shown in the asterisk malloc debug, and how can I figure what's causing it? The asterisk version is 1.6.2.9. If it's not listed in the internal debug, then it's probably a memory leak in one or more of the external libraries linked into Asterisk. That could be anything from openssl to libxml2 (neither likely) to something specifically related to a second or third party module that you've loaded. If so, it should appear in the memory map of the process under that library. Try: pmap 22514 Also try running that command a bit later on and see what has changed. Thanks a lot, I found this process but it shows as anonymous: 08a6a000 340776K rw---[ anon ] And a lot of more anons that spend betwen 200K and 2000K... Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Using SetVar with System() is it possible?
Steve Edwards wrote:
On Sat, 19 Jun 2010, bruce bruce wrote:
Is it possible to harvest the output of system into a SetVar(variable)?
exten = s,n,SetVar(var=system(asterisk -rx sip show channels | grep
-c (ulaw))
??? any problem with the syntax?
) Your parentheses don't match.
) You didn't read the documentation that says system() returns FAILURE or
SUCCESS.
) You didn't notice that setvar() is deprecated.
) You didn't read the documentation that says set[var]() sets a name to a
value -- no mention of evaluating an application.
And you can't return the output of a dialplan application within another
application -- that's what functions are for.
With 1.6.0 and later, you have the SHELL() function which allows you to do what
you're trying to do:
exten = s,n,Set(RESULT=${SHELL(...)})
As mentioned by Steve, it would be trivial to find out what was wrong with your
syntax if you had tested it in your development environment.
Leif.
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Re: [asterisk-users] Asterisk 1.6 + Jabber crashes
Michael wrote: I am attempting to setup Asterisk to work with Gtalk. I am using the following versions: Slackware Linux 12.0 Asterisk 1.6.2.9 GNU TLS 2.8.6 Iksemel (svn v25) OpenSSL 0.9.8o It all compiles however about 10 seconds after starting Asterisk it crashes. If there is any other information needed please advise. http://svn.asterisk.org/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt Leif. -- _ -- 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] dialplan reload 1.4.33
When I execute a dialplan reload I am getting color commands (that I never got before) with the echoed Dialplan reloaded. The sip reload works as it always has with no color codes echoed. Its not detrimental just messes up my screen some. jerry -- _ -- 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] dialplan reload 1.4.33
Jerry Geis wrote: When I execute a dialplan reload I am getting color commands (that I never got before) with the echoed Dialplan reloaded. The sip reload works as it always has with no color codes echoed. Its not detrimental just messes up my screen some. jerry Sorry I meant to add - asterisk -rx dialplan reload -- _ -- 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] 'NO ANSWER' with cdr duration of 0
I have a system running Asterisk 1.6.2.6 that generates about 80k calls/day. Calls are fired from Asterisk Manager (async originate -- 60 second timeout). I am capturing 100% of the originate responses (recorded in DB). About 5% of the calls result in a reason code of '3' (Remote End is Ringing). CDR Disposition of these calls is 'NO ANSWER', and billsec is 0. All good there. What's confusing me is that in *all* these cases duration is also recorded as 0. IIRC, duration should be time spent in the system, so should be roughly equal to my timeout (60 sec). Is this a bug, or am I not understanding correctly? I know there was a bug where all calls were recorded as 'NO ANSWER', I was plagued with that until updating to 1.6.2.6, this does not seem to be related. Any feedback is appreciated. Thanks, Josh McAllister -- _ -- 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] Dialplan for conference
Hello, I wanted to add the functionality of 3-way conference to my asterisk pbx using meetme or confbridge. During a call the user should be able to put the other party on hold and dial another number, then on dialing some key sequence all three of them enters into a conference. Is it possible to do this? Can someone please point me to any dialplan examples to do this. Thanks -- _ -- 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] A lot of : doing dnsmgr_lookup for - Asterisk installed from YUM
Hi Guys, Asterisk 1.6.2.7 install from Yum Repository shows a lot of : doing dnsmgr_lookup for sip.provider.com Google searches show it was fixed in some version. Is this to be ignored? Thanks -- _ -- 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] realtime queues membername problem
On Thu, Jun 24, 2010 at 2:42 AM, Rob Coward r...@jive-videos.net wrote:
I'm just about to start experimenting with realtime queues, so can't offer
anything from my own experience, but what happens if instead of updating the
existing row, you delete it and insert a new one for the new user ?
ie. DELETE FROM queue_member_table WHERE id=1;
INSERT INTO queue_member_table
(membername,queue_name,interface,penalty,paused) VALUES
('OtherUser','in_pruebas','SIP/1336','1','0');
Hi, the result of delete or update is the same if I only change the
membername field :(
Asterisk CLI:
*CLI queue show in_pruebas
in_pruebas has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime),
W:0, C:0, A:0, SL:0.0% within 0s
Members:
(SIP/1336) with penalty 1 (realtime) (Not in use) has taken no
calls yet
No Callers
SQL:
mysql select * from queue_member_table where queue_name = 'in_pruebas';
+--+++---+-++
| id | membername | queue_name | interface | penalty | paused |
+--+++---+-++
| 4528 | | in_pruebas | SIP/1336 | 1 | 0 |
+--+++---+-++
1 row in set (0.00 sec)
mysql delete from queue_member_table where id = '4528';
Query OK, 1 row affected (0.00 sec)
mysql select * from queue_member_table where queue_name = 'in_pruebas';
Empty set (0.00 sec)
mysql insert into queue_member_table
(membername,queue_name,interface,penalty,paused) VALUES
('chassoul','in_pruebas','SIP/1336','1','0');
Query OK, 1 row affected (0.00 sec)
mysql select * from queue_member_table where queue_name = 'in_pruebas';
+--+++---+-++
| id | membername | queue_name | interface | penalty | paused |
+--+++---+-++
| 7261 | chassoul | in_pruebas | SIP/1336 | 1 | 0 |
+--+++---+-++
1 row in set (0.00 sec)
Asterisk CLI:
*CLI queue show in_pruebas
in_pruebas has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime),
W:0, C:0, A:0, SL:0.0% within 0s
Members:
(SIP/1336) with penalty 1 (realtime) (Not in use) has taken no
calls yet
No Callers
as you can see the problem is the same, I really don't know why or what to
do with this :( please help!
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[asterisk-users] SPA8000 outbound CID problem
Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a Use Auth ID checkbox. It's running 6.1.3 firmware, which looks to be the latest, and supports SIP trunking, though even if I don't use trunking, I have the same obstacle if I configure it per-line instead of per-trunk. Inbound CID works fine. When VOIP calls come in via the provider or Asterisk, the SPA generates CID on it's analog ports. The problem is that the outbound caller ID number seems to come from the SIP User ID setting, which is also the SIP authentication name. If I instead put the SIP account id into the Auth ID field and check the Use Auth ID box, Asterisk reports: Registration from 'John Smith sip:jsm...@our.sip.gateway.com' failed for '1.2.3.4' - Username/auth name mismatch. Sure, I can overide the CID number on our Asterisk server, but I don't have that ability with the VOIP provider's Asterisk server. The outbound caller ID always looks like John Smith jsmith instead of John Smith 211212 no matter how I try to set these fields. I take it the SIP username and auth name need to match, so that leaves me with the question of how to configure a CID number that doesn't necessarily match the SIP user/auth name. Is this a limitation of this device, or is there some other option I'm overlooking? Mark -- Mark G. Thomas (m...@misty.com) Web: http://mgtinternet.com/ Tel: +1-215-512-0112 US: 877-512-0112 -- _ -- 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] Dialplan for conference
Its possible but not easy. Search for n-way conferencing on voip-info.org, it has all the details on how to do it. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 2:01 PM, Deepesh D deep.d2...@gmail.com wrote: Hello, I wanted to add the functionality of 3-way conference to my asterisk pbx using meetme or confbridge. During a call the user should be able to put the other party on hold and dial another number, then on dialing some key sequence all three of them enters into a conference. Is it possible to do this? Can someone please point me to any dialplan examples to do this. Thanks -- _ -- 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] T.38 on a MAX/Lucent/Ascend TNT
Hello folks, I've been trying to get T.38 over SIP working with calls terminated by a MAX/Lucent/Ascent TNT. As far as I can tell, SIP and T.38 are actually working perfectly; however, I can't get the TNT to properly terminate a FAX call. Does anyone have a working configuration for SIP and T.38 for calls from a TNT or APX? Here's a brief description/diagram of my test setup: Laptop --RS232-- Modem --POTS-- Channel bank (ADIT 600) --CAS T1-- TNT --Ethernet/SIP-- Asterisk --SIP-- t38modem. The TNT is running TAOS 11.0.4 with both the SIP and realtime fax features licensed. I've tried using several modems, and with each I can establish a data call without any problems (although at a maximum of 31200bps/3429 baud using a 33.6 modem, for reasons I haven't dug into yet.) Whenever I try to send a fax from the laptop, however, the call always seems to fail in the first HDLC phase (phase B) with either a timeout or error 23 (COMREC invalid command received.) The modem is connected directly to the channel bank and the channel bank is connected directly to the TNT in an attempt to reduce the number of variables. With my current configuration, the call to Asterisk will come up as a voice call, then be dumped into t38modem when Asterisk receives the 1100Hz CNG tone from the sending modem. At that point, the call is dumped into t38modem which re-invites the TNT with the T.38 options, and the TNT usually sends a T.38 t30-ind of 0x00 (no-signal) or 0x3a (???), although I do occasionally see more promising messages like v29-7200-training or t30-data/hdlc-fcs-ok. Shortly thereafter, the dialing modem will give up and terminate the call. If I try dumping the same call into iaxmodem instead of t38modem, the call actually progresses further -- the real modem receives and decodes the HDLC CSI/DIS from iaxmodem, but the high-speed trainup always fails for calls coming from the TNT. Does anyone have any advice or suggestions? Has anyone actually made T.38 work with one of these devices running ANY TAOS version? Thanks, -- Ben Winslow winsl...@pa.net -- _ -- 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] OT: Bandwidth calculations
Hi, I know some of you are very experienced as to the working of networks. I wondered whether there is some accepted way of determining bandwidth needs based on the network traffic over time. For example, looking at the figures for the network traffic through the server interface, we have hourly, daily and monthly figures. If everything were linear, taking the hourly figure and dividing it by 3600 (or the daily by divided 3600*24) would give us the required bps, but this average is pretty meaningless. Those of you who have experience and education in this area, where can I look for guidance (links?) and do you have any rules of thumb you'd care to share? I'm actually looking for this for a web server not VoiP, but any info is welcome. It seems obvious to me that taking the per second average and multiplying it by some kind of seat of the pants number must give a decent idea? WHat is that magic number? Thanks in advance for any ideas. /r -- _ -- 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] OT: Bandwidth calculations
ISP 10% rule is what you are asking about expected that average usage is 10% of total subscribers with bursts higher ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Jun 24, 2010 at 3:46 PM, Randy R randulo2...@gmail.com wrote: Hi, I know some of you are very experienced as to the working of networks. I wondered whether there is some accepted way of determining bandwidth needs based on the network traffic over time. For example, looking at the figures for the network traffic through the server interface, we have hourly, daily and monthly figures. If everything were linear, taking the hourly figure and dividing it by 3600 (or the daily by divided 3600*24) would give us the required bps, but this average is pretty meaningless. Those of you who have experience and education in this area, where can I look for guidance (links?) and do you have any rules of thumb you'd care to share? I'm actually looking for this for a web server not VoiP, but any info is welcome. It seems obvious to me that taking the per second average and multiplying it by some kind of seat of the pants number must give a decent idea? WHat is that magic number? Thanks in advance for any ideas. /r -- _ -- 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] OT: Bandwidth calculations
http://www.asteriskguru.com/tools/bandwidth_calculator.php ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Jun 24, 2010 at 3:46 PM, Randy R randulo2...@gmail.com wrote: Hi, I know some of you are very experienced as to the working of networks. I wondered whether there is some accepted way of determining bandwidth needs based on the network traffic over time. For example, looking at the figures for the network traffic through the server interface, we have hourly, daily and monthly figures. If everything were linear, taking the hourly figure and dividing it by 3600 (or the daily by divided 3600*24) would give us the required bps, but this average is pretty meaningless. Those of you who have experience and education in this area, where can I look for guidance (links?) and do you have any rules of thumb you'd care to share? I'm actually looking for this for a web server not VoiP, but any info is welcome. It seems obvious to me that taking the per second average and multiplying it by some kind of seat of the pants number must give a decent idea? WHat is that magic number? Thanks in advance for any ideas. /r -- _ -- 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] Dialplan for conference
On Thu, Jun 24, 2010 at 2:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Its possible but not easy. Search for n-way conferencing on voip-info.org, it has all the details on how to do it. Or you could post the direct link: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO -- 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] Dialplan for conference
I could, but it depends upon from where I am replying. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 4:28 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Jun 24, 2010 at 2:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Its possible but not ... Or you could post the direct link: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] Dialplan for conference
I know on my polycom phones, I just press the conf button, dial, and then hit join, and all done, no special programming required on dialplan. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Thursday, June 24, 2010 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialplan for conference On Thu, Jun 24, 2010 at 2:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Its possible but not easy. Search for n-way conferencing on voip-info.org, it has all the details on how to do it. Or you could post the direct link: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO -- 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 -- _ -- 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 extension dialing error
Dear, just a short question: If I use G.711a and G.711b codecs between the Portech GSM Gateway and Asterisk 1.4.23, what DTMF mode is better to use in both sides if a mobile phone call the GSM Gateway in order to contact an internal IP extension (Mobile to LAN scenario): RFC2238 Inband SIP INFO What are the requisites to choose among them Thanks in advance Alejandro 2010/6/18 Danny Nicholas da...@debsinc.com: I would definitely change the prompt from 1 to 0. It is not an advisable practice to have an IVR selection that can be misinterpreted like this. Assuming that all of your extensions are in 1000-1999, 2 for the operator would be just as good; the important thing is that you don't have a single digit extension 1. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Friday, June 18, 2010 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error Hi, I tell you I've made some calls from a land-phone to my IVR in order to avoid the possible poor quality of cell phone's DTMF, and when I called extension 1003 I was connected to extension 1000 againthe same error. My IVR says dial 1 to connect to operator or dial the extension in case you know.and my extension ranges is 1000-1999, so I think it could be a problem that extensions and IVR option start with the same digit: 1. When I'll be at work I'm thinking in modify the IVR speech in order to say dial 0 to connect to operator., and not dial 1 to connect to operator, so IVR option and extensions will not start with the same digit. Do you think this may be the problem ??? Thanks a lot and sorry for my interruption. Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: According to this link http://www.smallnetbuilder.com/content/view/30469/82/1/2/ You probably want to make 80 be 120. This is a millisecond delay value, so the 500 value is a give it up proposition; 200 might be doable for your outliers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Thursday, June 17, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error OK, now I understand..but just one more question...In the DTMF settings tab from the GSM gateway manager I have this line: Mobile DTMF debounce: 80 (range: 40 ~ 500, default: 80 ) step: 10ms What does this setting really mean and do I have to modify the current value ??? Final thanks :) 2010/6/17 Zeeshan Zakaria zisha...@gmail.com: I once setup a callback system for someone and we had these DTMF issues on constant basis, and all the complains were from cell phone users. At that time I found out that even my own cellphone would not DTMF correctly from certain locations, including my home, but would work perfectly fine from my work location. Probably times of the day matters too, but yes, calling from cell phones does result in DTMF issues, and the reason is that it is just the audio signals, which get distorted based on various factors like the signal strength, cell tower transmission quality, transcodings, etc. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-17 11:25 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Danny, so you say it's a problem of the cell phone and not the Astreisk or GSM Gateway ??? OK, in this case if I call from a fixed phone (not a cell phone) to the IVR, the DTMF quality problem will not be presentthis may be a good test, isn't it ??? Or do you suggest another test I can implement ??? Thanks again Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: The physical location of the phone (access to towers) can vastly affect the quality of DTMF pass... -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocati... -- _ -- 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
Re: [asterisk-users] Astersik can not detect DTMF key
Hi Almost phones I used meet this problem, they are - Nokia 1200 - Nokia 6210 - Nokia E72. When I used Softphone to test on IP (SIP), some softphones meet similar problem (twinkle), some don't meet (Xlite, Kapanga). Very Thanks Giang From: Zeeshan Zakaria zisha...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, June 24, 2010 5:48:39 PM Subject: Re: [asterisk-users] Astersik can not detect DTMF key What type of phone you are using? It is possible that # is used by this phone as one of its internal functions. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:41 AM, huu giang huugiang...@yahoo.com wrote: Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- 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] Astersik can not detect DTMF key
Hi, When I require user enter a code and end wich # key, for example 1234#, Asterisk can detect # key and detect the code people just enter. Thanks From: Zeeshan Zakaria zisha...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, June 24, 2010 5:48:39 PM Subject: Re: [asterisk-users] Astersik can not detect DTMF key What type of phone you are using? It is possible that # is used by this phone as one of its internal functions. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:41 AM, huu giang huugiang...@yahoo.com wrote: Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- 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] Astersik can not detect DTMF key
Nokia cell phones is a totally different case altogether because it used inband dtmf. I can't make any guess on this. As for the sip soft phones, for twinkle, there must be something in its settings which is using # for something else. You mentioned Xlite works fine. This shows that nothing is wrong on Asterisk end as far as the out-of-band dtmf is concerned. -- Zeeshan On Thu, Jun 24, 2010 at 10:14 PM, huu giang huugiang...@yahoo.com wrote: Hi, When I require user enter a code and end wich # key, for example 1234#, Asterisk can detect # key and detect the code people just enter. Thanks -- *From:* Zeeshan Zakaria zisha...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Thu, June 24, 2010 5:48:39 PM *Subject:* Re: [asterisk-users] Astersik can not detect DTMF key What type of phone you are using? It is possible that # is used by this phone as one of its internal functions. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:41 AM, huu giang huugiang...@yahoo.com wrote: Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- 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 -- Zeeshan A Zakaria -- _ -- 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] Big time system
We are an asterisk user... small time system 50-100 users or so. But, we have an opportunity to get into a big time telecom activity. It would have 2000 to 30,000 user lines per city, and we would like to have those brought back to a central location for control and because transport can be more economical than remote site rentals, maintenance and personnel. We could take the local lines into concentrators (TNTs or equivalent) and bring back IP to a central site, or put servers at the remote cities. Our object is to serve as a central office switch for subscribers on standard telco service loops. This isn't a How many lines can I handle using a Belchfire 2600 processor? type question but a request for pointers to big time systems. There would be no IP path to the end user, just copper. Thank you Cary Fitch -- _ -- 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
