[Asterisk-Users] can a sip.conf stanza be shared by several phones?
Hi, If several phones register to the same sip.conf section what will happen with a Dial SIP/shared in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial'ing multiple SIP devices impossible when forward activated
Hi, When I Dial(SIP/1SIP/2SIP/3) if any of these phones has a forward to another destination (302: moved response) then the simultaneous ring stops immediately and the incoming call goes to wherever the forward points to. We are using simultaneous ringing as a fallback when the receptionist doesn't anwser after a while and such a call should never be forwarded. Is there a way to tell * to ignore any forward on certain calls? Thanks for your help, -- Fast Food: Corporate America in your body Television: Corporate America in your mind. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dial'ing multiple SIP devices impossible when forward activated
On Fri, Apr 01, 2005 at 04:12:05PM +0200, Julius Vindex wrote: When I Dial(SIP/1SIP/2SIP/3) if any of these phones has a forward to another destination (302: moved response) then the simultaneous ring stops immediately and the incoming call goes to wherever the forward points to. We are using simultaneous ringing as a fallback when the receptionist doesn't anwser after a while and such a call should never be forwarded. Is there a way to tell * to ignore any forward on certain calls? Following up to myself. Thanks to ManxPower I found out that the s,1,Anwser I had in my macro was playing foul by allowing the forwarded call to reenter the macro and become Answer'd, which stopped all other devices from ringing. Now the second part of my problem remains: I'd like to disable any forwarding when ringing multiple SIP devices. i.e. I'd like to ignore any 302: moved messages sent by SIP devices. Is that feasible? -- Save the whales. Feed the hungry. Free the mallocs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] distinctive ringing in a queue?
Hi, Is it possible to have distinctive ringing in a queue? I've tried: exten = s,2,SetVar(ALERT_INFO=Custom 1) exten = s,3,Queue(standard|r) without success. However the SetVar(...) works fine when just dialing a SIP device. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] patch to add distinctive ringing to queues
Please find attached a patch I made to app_queue.c to add distinctive
ringing support. So the following works:
exten = 2131,1,SetVar(ALERT_INFO=Internal)
exten = 2131,2,Queue(standard|r)
I took code in app_dial.c and lightly adapted it.
I hope this gets included in * as it is really useful. I faxed my
disclaimer earlier.
Cheers,
PS: I am unable to create an account on the BTS (no password is ever
sent and no, my spam filter didn't eat it) or login to my previoulsy
created account.
--
If you're happy, you're successful.
--- ./asterisk-1.0.7.dfsg.1/apps/app_queue.c2005-04-07 17:08:17.0
+0200
+++ ./asterisk-1.0.7.dfsg.1+ldm/apps/app_queue.c2005-04-06
17:10:42.0 +0200
@@ -548,6 +548,8 @@
static int ring_entry(struct queue_ent *qe, struct localuser *tmp, int *busies)
{
int res;
+ struct ast_var_t *current, *newvar;
+ struct varshead *headp, *newheadp;
if (qe-parent-wrapuptime (time(NULL) - tmp-lastcall
qe-parent-wrapuptime)) {
ast_log(LOG_DEBUG, Wrapuptime not yet expired for %s/%s\n,
tmp-tech, tmp-numsubst);
if (qe-chan-cdr)
@@ -568,6 +570,26 @@
(*busies)++;
return 0;
}
+ /* If creating a SIP channel, look for a variable called */
+ /* VXML_URL in the calling channel and copy it to the*/
+ /* new channel. */
+
+ /* Check for ALERT_INFO in the SetVar list. This is for */
+ /* SIP distinctive ring as per the RFC. For Cisco 7960s, */
+ /* SetVar(ALERT_INFO=x) where x is an integer value 1-5. */
+ /* However, the RFC says it should be a URL. -km- */
+ headp=qe-chan-varshead;
+ AST_LIST_TRAVERSE(headp,current,entries) {
+ if (!strcasecmp(ast_var_name(current),VXML_URL) ||
+ !strcasecmp(ast_var_name(current), ALERT_INFO) ||
+ !strcasecmp(ast_var_name(current), OSPTOKEN) ||
+ !strcasecmp(ast_var_name(current), OSPHANDLE))
+ {
+
newvar=ast_var_assign(ast_var_name(current),ast_var_value(current));
+ newheadp=tmp-chan-varshead;
+ AST_LIST_INSERT_HEAD(newheadp,newvar,entries);
+ }
+ }
tmp-chan-appl = AppQueue;
tmp-chan-data = (Outgoing Line);
tmp-chan-whentohangup = 0;
___
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[Asterisk-Users] Polycom IP600 stuck at Running App = sip.ld (was: Re: Polycom 1.4.1 firmware for IP500/IP600)
On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote: If you have it, can I get a copy please, or possibly can you send it to the keeper of http://www.freedomphones.net/polycom/files/ I am looking for the latest boot image too. 1) I have the 1.4.1 firmware. To whom should I send the files? There is no contact info in this web site. 2) Now I am having a problem with my IP600 test unit: While performing tests on the Polycom IP600 I changed a configuration item and during reboot the phone stopped at the Running App = sip.ld stage and seems stuck there. I reinitialized all configuration files to their defaults from the zip files you sent me, to no avail. Plugging/unplugging the phone does not help as it starts and then stops booting at the same stage, while the message waiting indicator stays solid red (whereas previously it would flash continuously until full startup). Bootrom version: 2.6.1 Sip.ld version: 1.4.1.0040 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: broken message waiting indicator on Polycom IP600?
On Tue, Feb 01, 2005 at 09:41:25AM -0500, Noah Miller wrote: I faithfully followed the instructions from: http://www.voip-info.org/wiki- Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk but still the message waiting indicator doesn't flash when a message is waiting. There is a brief intermittent chirp but nothing more. Using latest firmware 1.4.1 My only suggestion would be to make sure you've put the correct VM context in the phone's sip.conf section - i.e. mailbox=[EMAIL PROTECTED]. It sounds like you must have done this, though, since you're getting the MWI chirp, just not the light. Do you get the stutter on the dialtone? Actually it goes this way: - leave message, - MWI flashes for ~ 10 minutes then stops, - stutter tone and enveloppe on line label still there, - 5 minutes later there is a single chirp and flash, - nothing more, Strange behaviour, isn't it? -- Field Artillery lends dignity to what would otherwise be a vulgar brawl. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
Hi, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? Thanks, -- Lord, protect me from your followers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
On Tue, Feb 15, 2005 at 07:56:10PM +1100, Adam Goryachev wrote: On Tue, 2005-02-15 at 09:38 +0100, Louis-David Mitterrand wrote: Hi, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? It could quite possibly be related to the log files the polycom uploads to the FTP server. I found this quite a pain, so disabled all the logging in the config files. If that isn't it, then you will need to find out *what* activity the polycom is doing, hint tcpdump -tn -A -s 16384 port 21 might help or else see your ftp server log files. You are right, this activity is related to logging. After consulting the admin manual I am unsure as to what settings related to logging are safe to change (some are marked as don't modify without consulting Polycom). Do you remember which settings you changed to disable logging? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why does the Polycom IP600 check FTP every 60 seconds...
On Tue, Feb 15, 2005 at 08:26:42PM +1100, Adam Goryachev wrote: On Tue, 2005-02-15 at 10:14 +0100, Louis-David Mitterrand wrote: You are right, this activity is related to logging. After consulting the admin manual I am unsure as to what settings related to logging are safe to change (some are marked as don't modify without consulting Polycom). Do you remember which settings you changed to disable logging? I changed the settings that it told me not to... Basically, I think I changed the various log levels to don't log anything... Hope that helps, if not, let me know and I can send you the polycom file off-list... OK, I changed only the following settings: log.render.realtime=0 log.render.stdout=0 log.render.file=0 and now proftpd is quiet. Thanks again, cheers, -- If you're not having fun right now, you're wasting your time. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail auth failure
When I access voicemail remotely, from a gsm phone say, some extra characters get inserted in my dtmf tones: when I type , * understands 88f8f8 (it always seems to be 'f'): -- Incorrect password '88f8f8' for user '2130' (context = any) And the 'f' always starts after the second digit. Might it be related to this warning message? Feb 4 16:40:59 WARNING[622613]: res_adsi.c:234__adsi_transmit_messages: Unknown ADSI response 'f' This is on debian unstable with 7.1 packages. -- Every day is a gift, that's why the present is so named ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to access the underlying channel of Local?
Hi, I am in the process of setting up call forwarding through capiECT with the 7960's CFwdAll button. When the phone redirects the call to an outside number (through a 302 SIP redirect) then the CAPI[contr1/xxx] channel becomes Local/[EMAIL PROTECTED] as the call is reinjected into the dialplan. To perform the capiECT I need to test if the call is originally from the CAPI interface, however I can find no way to access that info. All I get is Local/xxx. Is there a way to access the original channel, behind Local? Thanks, -- What? You haven't picked up on the new legal system in the US? If it annoys a large company, it's illegal. -- (thudfactor on Slashdot) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] picking up a ringing extension
Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 SIP problem when calling from outside of LAN
Hi, I am testing a 7960 in this context: [SIP] --- VPN --- [*] --- [ANY] (ANY == any type of phone: isdn, SIP, IAX, etc.) the call goes through and is dropped after 5 seconds with this message in the log: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call IP address for seqno 101 when calling from the LAN with the exact same phone: [SIP] --- LAN --- [*] --- [ANY] it works fine, what could be wrong? I am using the asterisk packages from debian. Thanks in advance for any help, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bad clicking sounds with Diva+capi+asterisk
Hello, We have been using a Diva 4BRI with our Asterisk PBX through the capi interface for almost a year now with good results. However, recently we started to hear heavy clicking sounds in our phones when two simultaneous incoming calls are processed by the card. The clicking does not originate with the phones as it happens also in voicemail left directly on the server and happens with different phone models. Theses interferences almost prevent us from hearing the remote party and they don't always happen on all simultaneous calls. For instance if we hang up a corrupted call and call back then the line is clean. The clicking sometimes sounds like crosstalk between both lines. After talking on irc with chan_capi's developper (kapejod) it seems the problem might originate from the Diva card. We are using a vanilla 2.6.7 kernel on debian unstable. We recently updated to use the very latest Diva firwmare and divactrl-2.1 without any effect on the problem. I suspect the problem might be related to the Diva 4BRI drivers in kernel 2.6.x as the problem seems to have started when switching from 2.4.x. Has someone had a similar experience? Is there any solution? Thanks in advance for your insight, -- Jesus is coming! Everyone look busy! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remotely picked-up extension keeps ringing
Hello, As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco 7960 with *8 but the extension then keeps ringing indefinitely, even though I picked up the call. Is this a known issue? Thanks, -- There are no Debian developers in any part of Hell, because the good karma incurred by being one takes you straight to the pearly gates. Of course, the frequent flame wars you put up with on the Debian lists make up for this on Earth. - Seth Cohn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] disconnect when 7960 far from * (was Re: Pointer to upgrade 7960sip beyond v3.2.0?)
On Thu, Sep 04, 2003 at 10:56:10PM -0700, Andrew Gillham wrote: Unless you're hoping to load Linux or some pirate image in the future, there is no reason to stay with the old code. At least I have not experienced any new issues I can attribute to the update to 5.3 code. Hello, I bought my 7960 phones used with the 4.4 sip image and suffer from disconnections after 3/5 seconds if the phone is connected to a remote asterisk, for example at the remote end of a VPN (when the 7960 is on the same LAN as asterisk all is well). Do you think upgrading to 5.x series images would solve that issue? Thanks, -- [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Callgroup, Pickupgroup and SIP
On Tue, Sep 09, 2003 at 08:23:49AM +, WipeOut . wrote: OK you are correct.. *8 picks up the call..I wonder why *8# does not work?? I also had the same problem that the phone that I collected the call from did not stop ringing.. I have the same problem. Mark Spencer is working on the bug, he logged into my machine yesterday to see it first hand. -- I had no wish to arrive, but I had to do my utmost, in order to arrive. -- Samuel Beckett, The Unnamable ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP LD carrier
On Tue, Sep 09, 2003 at 07:57:20PM -0400, Jeremy McNamara wrote: Travis Johnson wrote: I've called NuFone and was not impressed by their voicemail answering system (choppy) and was unable to even leave a message before the phone call was disconnected (in the middle of the recording). So your going to judge our system by making one phone call into my home asterisk system that runs on a fully saturated ADSL connection. Wait... of course people are going to judge you by that! If putting your company's answering machine on your (saturated) dsl connection is an indication of your other technical choices it reflects poorly on your company. Also your website is almost totally empty of details on your services and pricing but heavy on doubtful eye candy. If people on this mailing list had not time and againg recommended the excellence of your service I wonder how you would find your customers. I for one sent an enquiry at [EMAIL PROTECTED] 20 hours ago and still haven't received an answer. How hard is it to install a generic e-mail answerer saying here is our price list, please be patient I will contact you? Now I can understand that when developping a technically ambitious platform such as yours, communication and marketing is not your priority, but there is no need to bash your prospective customers with insulting remarks such as this one below: Typical, ^^^ -- Marijuana is nature's way of saying, Hi!. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] running * on a VPN gateway
If like me you run * on a VPN (or multihomed) gateway and want to serve remote SIP clients, make sure you have bindaddr = 192.168.0.1 ; or whatever is your box's private IP otherwise * might bind to its public IP and send it as return address in the SIP call setup, which will (should) be rejected by your firewall. To * experts: might this setting interfer with NATed SIP clients? -- I feel naked outside of Vim. -- Ted Knab ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 Firmware Upgrade
On Mon, Sep 15, 2003 at 10:48:00PM -0400, [EMAIL PROTECTED] wrote: I upgraded my 7960 firmware to ver 4.4. I now can't make any calls and I get errors (retrans_packet) on call on the console maximum retries exceeded. And ideas? Check that the bindaddr in sip.conf is set to a reachable address if your * box is on a mutli-homed gateway. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream Source in the EU?
On Mon, Sep 15, 2003 at 10:27:15PM +0200, Dave Cotton wrote: On Mon, 2003-09-15 at 22:11, Tom (UnitedLayer) wrote: Anyone have a good source for BT-101 phones? Yes. But it may not work for you because I've no idea on which of the 5 continents you are. I am looking for Grandstream phones (BT-102) in France, EU. Importing them from the US seems really a waste of money since one would pay double duty (.cn(?) - .us and .us - .fr). Is there a GS distributor in the EU or is it possible to order them direct from the manufacturer? Thanks for any info, -- HIPPOLYTE: Argos nous tend les bras, et Sparte nous appelle. A nos amis communs portons nos justes cris ; (Phèdre, J-B Racine, acte 5, scène 1) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ISDN BRI active adapters with NT mode - any alternatives ?
On Tue, Sep 16, 2003 at 11:10:33AM +0200, Jean-Marc V. Liotier wrote: On Mon, 2003-09-15 at 11:52, Klaus-Peter Junghanns wrote: i dont think that the Eicon Diva Server 4BRI's NT mode feature will work with linux/capi. I think the feature in the driver is for their PRI cards (where everything is always P2P). i may be wrong, though. I just had a chat with Eicon's French representatives and they confirm that the Eicon Diva Server 4BRI supports NT mode with Eicon's binary drivers that Eicon supports for Red Hat and Suse. They have no idea what the capabilities of the standard kernel drivers are : they do not support software libre. I took a look at the source and there may be something relevant at line 341 of drivers/isdn/eicon/bri.c but actually understanding what this code is supposed to do is far beyond my technical abilities. It is quite strange that Eicon's representatives have no clue about the standard kernel driver's capabilities : drivers/isdn/eicon/bri.c is copyright Eicon. But maybe the people who actually know are at Eicon's headquarters, not in France. Anyone here knows who the relevant contacts may be ? I am using the Diva 4BRI daily with our * and indeed it does support NT mode on a port by port basis: when you configure the card initially you are specifically asked whether you want ports in TE or NT mode. And this is with the open-source Melware drivers from http://mmm.melware.de. Now I have no idea if * supports plugging ISDN phones in the Diva. AFAIK it's not supported by chan_capi, but that may change. -- In linux-dominated parallel world, all children's names begin with g, k or x ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] failed to load chan_zap
Suddenly after recompiling my 2.4.22 kernel I can no longer load chan_zap: Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span status: Inappropriate ioctl for device Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 6638 (load_module): Unable to register channel '1' Sep 22 21:25:08 WARNING[16384]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 Sep 22 21:25:08 WARNING[16384]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! The wcfxo module loaded without warning: Sep 22 21:23:44 zenon kernel: Zapata Telephony Interface Registered on major 196 Sep 22 21:23:44 zenon kernel: Found a Wildcard FXO: Wildcard X101P Sep 22 21:23:44 zenon kernel: Registered tone zone 2 (France) What could be wrong? -- (remember when we spent millions coming up with a pen that would write in zero-G and the Russians just used pencils?) -- limekiller4 on /. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] failed to load chan_zap
On Mon, Sep 22, 2003 at 03:41:43PM -0400, Jeremy McNamara wrote: cvs update the zaptel source and make clean install it. That did it, thanks a lot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Anyone looking for IP Phones?
On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote: My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of service. They were deployed for about 6 months. These include the AC power adapter and station license. We also have some other related equipment. If someone is reading this and is interested, shoot me an email [EMAIL PROTECTED] Man, 500 phones at ~ $450 a pop decommissioned after a mere 6 months? Did they not fulfil your needs? Were you disappointed with them? As I'm thinking about deploying cisco 79xx phones, I'm just curious about your experience. -- Logiciels libres : nourris au code source sans farine animale. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Tested 7905G
On Mon, Oct 20, 2003 at 09:21:45AM +0200, Michael Devenijn wrote: Justy to let you all know that i tested 7905G phone with a SIP image (latest download) and it works great ! for a reasonable price but with a good quality and a brand ... which inspires trust and helps selling better The only minus : Missing a microphone to work handsfree (or i didn't find it.) but strange enough their is a speaker ... Yeah, that's a real bummer. Cisco calls that feature Monitor mode, ie: you can listen but not speak. Oh so useful! I'm sure including a microphone would have threatened their whole product line price structure... -- May the Legos (TM) always be swept from your path in the night. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to escape #
Hi, This morning I found myself stumped when a remote interactive system asked me to enter some identification followed by the # key, and my local Asterisk interrupted with Transfer?. Is there a way to escape the pound key, short of disabling transfers? Cheers, -- Make it idiot proof, and somebody will make a better idiot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to escape #
On Mon, Oct 20, 2003 at 02:55:18PM +0100, WipeOut wrote: This morning I found myself stumped when a remote interactive system asked me to enter some identification followed by the # key, and my local Asterisk interrupted with Transfer?. Is there a way to escape the pound key, short of disabling transfers? Don't put the T or t options on your outbound dialing string.. So basiccally one has to choose between being able to transfer calls or sending # to remote IVR systems? I'd call that a bug -- Of course Australia was marked for glory, for its people had been chosen by the finest judges in England. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Tested 7905G
On Mon, Oct 20, 2003 at 08:49:42AM -0700, John Todd wrote: At 2:54 PM +0200 10/20/03, Louis-David Mitterrand wrote: Missing a microphone to work handsfree (or i didn't find it.) but strange enough their is a speaker ... Yeah, that's a real bummer. Cisco calls that feature Monitor mode, ie: you can listen but not speak. Oh so useful! I'm sure including a microphone would have threatened their whole product line price structure... -- May the Legos (TM) always be swept from your path in the night. The 7905 with a microphone is called the 7912, and is more expensive. Go figure. Unfortunately the 7912 is just a 7905 with a switch (two ethernet ports), nothing less, nothing more, and still no microphone. If you found a mike on yours please let me know. -- Hand, n.: A singular instrument worn at the end of a human arm and commonly thrust into somebody's pocket. -- Ambrose Bierce, The Devil's Dictionary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Setting up an IAX2 trunk
On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote: Also trunking requires that some sort of timing device (digium card or ztdummy) be in place for trunking. Otherwise trunking is disabled. What does ztdummy require to work? kernel compile options? Does it work on SMP systems? Typically if you have trunking enabled in your config and neither timing device present a log message will be generated to inidicate that you need a timing device for trunking to be enabled. On both ends ? (of course, but just to make sure) Thanks for the info, -- There is no operating system but linux and linus is its kernel maintainer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Setting up an IAX2 trunk
On Thu, Oct 23, 2003 at 07:55:09PM +0200, Olle E. Johansson wrote: Louis-David Mitterrand wrote: On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote: Also trunking requires that some sort of timing device (digium card or ztdummy) be in place for trunking. Otherwise trunking is disabled. What does ztdummy require to work? kernel compile options? Does it work on SMP systems? http://www.voip-info.org/wiki-Asterisk+timer So if I understand correctly: on SMP systems without UHCI USB support the _only_ way to have a zaptel interface is to purchase a card from Digium? In other words, SMP systems will always have exclusive use of /dev/rtc? Thanks, -- Disclaimer - These opiini^H^H damn! ^H^H ^Q ^[ :w :q :wq :wq! ^d X exit X Q ^C ^? :quitbye CtrlAltDel ~~q :~q logout save/quit :!QUIT ^[zz ^[ZZ ^H man vi ^@ ^L ^[c ^# ^E ^X ^I ^T ? help helpquit ^D mhelp ^C ^c help exit ?Quit ?q CtrlShftDel Hey, what does this button d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: QoS What to do?
On Mon, Nov 03, 2003 at 11:32:11AM +0100, Roy Sigurd Karlsbakk wrote: If your DSL link is the bottleneck, rather than earlier hops back through the providers network, the provider could also prioritize VOIP packets going up the DSL line. That requires a cooperating provider, of course. You may also setup a linux box (or another QoS supporting router) on the inside and tune the communication with queueing there. Read the LARTC howto for more info. FIAIF at http://www.fiaif.net/ is an excellent piece of software which integrates firewall and QOS (based on lartc.org's wondershaper) functions. -- Poor girl looks as confused as a blind lesbian in a fish market. - Simon R. Green ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX midget packets!?
Hi, At the * console I periodically get these messages: Dec 9 10:58:11 WARNING[-1248765008]: chan_iax2.c:5021 socket_read: midget packet received (1 of 4 min) Which seem pretty inocuous. Google say (almost) nothing about that subjet. What does it mean? -- Field Artillery lends dignity to what would otherwise be a vulgar brawl. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Thomson ST2030 firmware upgrade
Hello, I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42 firmware to the latest version (1.56) through tftp. The phone loads the .inf file, then the correct firmware file (as stated in the ST2030S.inf), then it reboots and loops doing these same things again and again. The firmware version on the phone stays at 1.42. Is there a special intermediate firmware version to use before going to the latest? Something special to include in the .inf file? I looked everwhere on the Net (including voip-info). Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX2 trunking on one side only.
On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote: Hello, I have searched google, read everything on the mailing list, read /usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on the IRC channel and I cannot figure out what is wrong with my IAX2 trunk. Only asterisk2 of an ASTERISK1--LAN--ASTERISK2--PSTN will use IAX2 trunking. If I do an iax2 show trunk on asterisk1 it says 0 calls on trunk Do you have a zaptel device on each side? AFAIR zaptel timing is required for trunking to work. -- If Galileo is the spark that lights up the gas giant Jupiter, turning it into a second sun, that will be the last straw. We will then have no choice but to make safety the number one priority at NASA. -- falsification on /. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX/IAX2 encryption?
On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote: I second that, and I think I remember hearing Mark talking about it too. But. What type of encryption can you do that does not introduce latency? That said, I would like it to support hardware encryption cards. I have done work with FreeS/WAN and it works, and yes it adds about 30-100ms of latency depending on what else is going on. I think it has something to do with keying. Ipsec with Freeswan does _not_ add 30-100ms of latency, try a handful of ms: styx:/# ping 192.168.0.3 PING 192.168.0.3 (192.168.0.3): 56 data bytes 64 bytes from 192.168.0.3: icmp_seq=0 ttl=62 time=60.5 ms 64 bytes from 192.168.0.3: icmp_seq=1 ttl=62 time=64.2 ms 64 bytes from 192.168.0.3: icmp_seq=2 ttl=62 time=63.8 ms 64 bytes from 192.168.0.3: icmp_seq=3 ttl=62 time=62.2 ms 64 bytes from 192.168.0.3: icmp_seq=4 ttl=62 time=60.7 ms 64 bytes from 192.168.0.3: icmp_seq=5 ttl=62 time=73.0 ms styx:/# ping my.ipsec.gateway.com PING my.ipsec.gateway.com (85.89.188.89): 56 data bytes 64 bytes from 85.89.188.89: icmp_seq=0 ttl=57 time=57.5 ms 64 bytes from 85.89.188.89: icmp_seq=1 ttl=57 time=60.4 ms 64 bytes from 85.89.188.89: icmp_seq=2 ttl=57 time=57.5 ms 64 bytes from 85.89.188.89: icmp_seq=3 ttl=57 time=60.1 ms 64 bytes from 85.89.188.89: icmp_seq=4 ttl=57 time=59.2 ms 64 bytes from 85.89.188.89: icmp_seq=5 ttl=57 time=59.1 ms The first ping goes through a remote Ipsec gateway to reach an internal host (192.168.0.3) and the second one is directly to that Ipsec gateway's public IP. So latency is clearly not the issue. The main problem with ipsec packets is the lack of TOS support: data and voice traffic are agregated in one stream which is opaque to external routers. On further reflexion, either with separate IP addresses or ipsec nat traversal, a specialized voice ipsec tunnel could be setup with packets marked with the low-latency bit. That should work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: * Party in Paris
On Sat, Nov 29, 2003 at 11:28:56PM -0600, Mark Spencer wrote: I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Hi Mark, Nice to hear you are coming to Paris! I am based in Paris and will certainly be around during these times. You can count me in for any meeting, presentation, event, drink, orgywhatever takes place. My company already sells some * integration services in France and would like to go one step further, especially with regard to Digium hardware. Please let me know if I can help with your logistics and planning. Cheers, -- Linux: The Ultimate NT Service Pack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 lockups - any experiences?
On Mon, Dec 15, 2003 at 09:25:05AM -0500, John Todd wrote: Paul - Yes, your description is correct. - moving the phone (no ethernet passthrough) results in no symptoms You might have a virus on that XP box that totally saturates the poor 7960 switch with bogus IP packets. -- May the Legos (TM) always be swept from your path in the night. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom videoconferencing with asterisk?
Hello, Has anyone used Polycom's VSX line of videoconferencing equipment with Asterisk? It seems some of their models, namely the newer VSX 5000, supports SIP. -- The Internet used to be a lot of smart people sitting at dumb terminals, but now its a lot of dumb people sitting at smart terminals! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7920 wi-phone firmware
Hello, I just acquired a used Cisco 7920 wi-phone and it mostly works with the newest asterisk and chan_sccp, but it reboots after most calls. Would a kind soul send me the latest firmware for that phone? Thanks in advance, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quickfix for building zaptel with 2.6.24?
Hi, I am trying to build zaptel 1.4.8 with kernel 2.6.24 on debian/sid: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop. Is there a quickfix out there? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickfix for building zaptel with 2.6.24?
On Thu, Feb 28, 2008 at 11:10:37AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop. Is there a quickfix out there? Yes, use Zaptel 1.4.9.1 or wait for the release of 1.4.10 later today or first thing tomorrow. If you decide to use 1.4.9.1, please note that if you are using analog cards with FXO modules, there is a known bug in DTMF generation that will affect your ability to dial out on those ports. That has been fixed in Subversion (see issue 11855 on bugs.digium.com) and will be in the next release. Thanks for your answer Kevin, but I need the debian'ized bristuff'ed version to be able to package and deploy it. I'll just patiently wait for Tzafrir (thanks for your work!) to release them for debian. Cheers, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] discrepancy between CDR clid and Polycom IP601 clid
Hi, Returning to my office I find two missed calls (from autodialers) that my IP601 displays as originating from 011. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src:0172752780 dst:2131 dcontext: default channel:Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:VoiceMail lastdata: 2131|su duration: 55 disposition:ANSWERED amaflags: 3 accountcode: uniqueid: asterisk-4208-1207311496.129 userfield: How can the phone display a different clid than the CDR database? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] wireless headphone that can answer a call?
Hello and sorry for the OT, Is it possible for a wireless headset of which the base is connected to a Polycom IP601 to remotely answer a call? In the same way as a bluetooth headset. thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mismatched callerid on phone and CDR ?
Hi, Using asterisk 1.4.21.2. For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mismatched callerid on phone and CDR ?
On Wed, Oct 15, 2008 at 11:30:49AM -0500, Tilghman Lesher wrote: On Wednesday 15 October 2008 10:26:50 Louis-David Mitterrand wrote: For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? The ANI and CallerID do not necessarily have to match; they just generally do. The src field reflects the ANI, if set, with a fallback to CallerID number, if not. Thanks for your explanation. Is it then possible to record both informations in the CDR as well? And is there a way to display both fields on my phone's display? -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
On Mon, Apr 16, 2007 at 12:25:55PM +0200, Bas van der Veen wrote: Hello, Did you find anything while testing the LAN? Also, can you confirm that switching the switch, cabling, etc. did NOT solve the problem? It did not. We finally changed the server itself and reinstalled from a known-working installation at another of our sites. We also removed a 4BRI card percieved to be flaky (not needed on this 100% voip site). No more reboots since. I have spontaneous reboots with IP600's. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bad case of buzzing
On Wed, Apr 18, 2007 at 01:04:31PM +0200, Tim Koehler wrote: Hi, are you using PoE or power supplies? As power supllies usually are not grounded it could be that it's comming from the power source. We are using PoE You could try using a grounded PoE switch or probably a power backup to test if this is the case. The problem was solved by changing the server and installing a fresh OS image on it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.2.x - 1.4.x upgrade: dialplan block no longer works
Hi,
a block of my extensions.conf no longer works after upgrading from
1.2.17 to 1.4.4. I have:
[macro-dialout]
exten = s,1,Gosub(s-${ARG1},1)
exten = s,n,Congestion
;; default
exten = _s-!,1,Gosub(s-NET,1)
When calling that macro whith no argument ($ARG1 empty):
exten = _0[1-9],1,Macro(dialcapi)
The call is not routed. Apparently _s-! does not match s-:
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/0146472130-0821fe08,
dialcapi) in new stack
-- Executing [EMAIL PROTECTED]:5] Gosub(SIP/0146472130-0821fe08,
s-|1) in new stack
== Auto fallthrough, channel 'SIP/0146472130-0821fe08' status is
'UNKNOWN'
Any idea why?
___
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[asterisk-users] preventing voicemail pickup after SIP redirect ?
Hello, I'm using the classic [stdexten-macro] in extensions.conf whereby a call is picked up by voicemail after a certain ringing time. When programming a SIP phone to redirect calls (SIP 302 redirect) to another extension I'd like to avoid that voicemail pickup so that the call goes into the new destination's voicemail (if applicable). How can I detect that a call has been redirected and should no longer be intercepted by vm? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: preventing voicemail pickup after SIP redirect ?
On Tue, Mar 06, 2007 at 07:18:08AM -0600, Eric ManxPower Wieling wrote: How can I detect that a call has been redirected and should no longer be intercepted by vm? That should happen by default. The call should get sent to the new place and it should act like the call was directly dialed to that extension. Actually no. When a call coming in through Zap, Capi or mISDN is redirected by a SIP phone with a 302, then asterisk creates a Local/xx channel to the new destination, while the original channel is still open. So after $RINGTIME is reached, [stdexten-macro] answers the original call and sends it to the original extension's vm. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom random reboots
Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
On Wed, Mar 21, 2007 at 07:07:00AM -0400, joe a. wrote: Did you swap the power module as well? If POE, did you swap the patch cord? If the power module plugs into a power strip did you change that? or at least the position in the strip? Thanks for the tought, but the IP430 has no external power strip or module, it's fully integrated like the IP601. We changed the cable, the wall socket and the switch (was due for an upgrade). Now on to testing the LAN. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
On Wed, Mar 21, 2007 at 04:07:34AM -0700, Henry Cobb wrote: On 3/21/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that? Our Polycom 3s and 5s ship with flaky power supplies and tend to reboot all of the time (especially in India...), so we found replacement non-Polycom power supplies and they are much more stable. I should have added that we use POE with a 3com PWR-class switch. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote: Yes, I recently saw this with a 501, in my case the network drop was the problem. If you have a good tester then run it on the connection. I had another drop near by and just swicthed to it. What kind of test tool would you suggest? Usually we rely on the cabling guys for that but that entails a delay and I'd be interested in knowing how to do it myself. Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote: Yes, I recently saw this with a 501, in my case the network drop was the problem. If you have a good tester then run it on the connection. I had another drop near by and just swicthed to it. Was that phone using POE ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no incoming dad with mISDN 1.1.1 and asterisk?
Hello, After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I no longer match any extension. Apparently the dad is empty. However I can see the number just before it (146472130): P[ 4] I IND :SETUP oad:!?145201798p ¡146472130 dad: ¡146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 Mar 23 09:35:28 WARNING[6725]: chan_misdn.c:4750 chan_misdn_log: Extension can never match, so disconnecting P[ 4] I SEND:RELEASE oad:!?145201798p ¡146472130 dad: ¡146472130 pid:2 P[ 4] -- bc_state:BCHAN_CLEANED P[ 4] I IND :RELEASE_COMPLETE oad: dad: pid:2 state:EXTCANTMATCH P[ 4] hangup_chan P[ 4] - hangup P[ 4] * IND : HANGUPpid:2 ctx:default dad: ¡146472130 oad:!?145201798p ¡146472130 State:EXTCANTMATCH P[ 4] -- cause:2 P[ 4] -- out_cause:2 P[ 4] -- state:EXTCANTMATCH P[ 4] Channel: mISDN/4-u0 hanguped new state:CLEANING P[ 4] release_chan: bc with l3id: 40001 With mISDN-1.0.4 and the same asterisk it works fine: P[ 4] I IND :SETUP oad:145201798 dad:146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 P[ 4] I SEND:PROCEEDING oad:0145201798 dad:0146472130 pid:2 P[ 4] -- bc_state:BCHAN_CLEANED -- Executing Goto(mISDN/4-1, 2130|1) in new stack -- Goto (default,2130,1) -- Executing NoOp(mISDN/4-1, ) in new stack -- Executing Macro(mISDN/4-1, queue) in new stack -- Executing NoOp(mISDN/4-1, 0145201798) in new stack -- Executing Monitor(mISDN/4-1, gsm|20070323-093814-0145201798-2130|mb) in new stack -- Executing Queue(mISDN/4-1, 2130|rntT|||10) in new stack P[ 4] * IND : Indication [3] from s P[ 4] -- * IND : ringing pid:2 P[ 4] I SEND:ALERTING oad:0145201798 dad:0146472130 pid:2 P[ 4] -- bc_state:BCHAN_CLEANED P[ 4] -- * SEND: State Ring pid:2 P[ 4] -- incoming_early_audio off -- Called SIP/0146472130 -- Called SIP/ekiga -- SIP/0146472130-08199d18 is ringing I didn't touch to the mISDN installation other than upgrade the kernel and its modules (compiled on another machine). Should I also upgrade mISDNuser to 1.1.1 on that server? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bad case of buzzing
Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using SIP to our central PRI-equipped asterisk. Phones are Polycom 430, 601, Cisco 7960, 7912 all to the latest firmware. We tried everything: changing the switch, network cards, auditing every network drop with fluke, re-certifying our wan, swapping some phones to no effect. Has anyone gone through that ordeal? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] crashes after upgrade from 1.2.16 to 1.4.21.2
Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, unfinished ... +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached On a second sister-machine with a mirror install we have the same problem. So it doesn't seem to be a hardware problem. This is with a TE410P card. Any idea? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tired of midget packet received warnings
Hi, When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Is there a way to avoid these warnings? Or at least turn them off when at the console in non-verbose mode? Thanks, -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Hi, Is POKE a generic udp thing or specific to iax? In the former case I'll probably be able to submit a patch to wmnetmon (great dockable applet I'm using). Thanks, -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. Granted, the monitoring app is simple minded: it only checks if a port is open. In that respect is does a hell of a good job: I hear a beeping alarm as soon as an asterisk instance goes south. So what you are saying is that all monitoring apps should speak native iax, else they are bad? Simply checking if a port is open means it's misconfigured or badly written? I wouldn't go so far. Small generic port-monitoring apps should be allowed to check on asterisk without raising such spurious warnings. You know what happens when crying wolf to often, no one listens after a while. A midget packet is not corrupted, I do have a stateful firewall (fiaif) to intercept those. rant AFAIK the onus is on asterisk to adapat: I've suffered too long of the infamous iax2 port-clogging bug that would and render a server 'unreachable' for no good reason. So much so that I went off iax2 entirely and use SIP exclusively for inter-asterisk communication. So much for the muched touted new and advanced pbx communication protocol the iax2 was sold for! This deal-breaker bug went unfixed for years until recently, despite numerous asterisk users reporting iax2 anomalies month after month. A I bitter? yes. Do I trust Digium folks to know their stuff about what is correct or not in networking protocols? I'll let you guess the answer. /rant I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. T. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote: Tzafrir Cohen wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? Absolutely it does. Warnings of malformed packets are often (as mentioned above) symptomatic of network problems. Fix the network problem, fix the warning. C'mon, even firewalls give you the option of _not_ logging malformed packets! fiaif does. Else your logfile would be the weak point of your system. And what if you can't fix the source of these packets? And what if friendly peers outside of your realm (likely to iax-call you, so can't block them) sends these packets? There are holes in your logic. So asterisk has to be puritan of the lot? Holier than thou? Pro-life with malformed packets? I see where this is going and I don't like it one bit. -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] crashes after upgrade from 1.2.16 to 1.4.21.2
On Thu, Nov 06, 2008 at 11:46:48AM +0100, Louis-David Mitterrand wrote: Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, unfinished ... +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached No other crash yet but an asterisk instance eating all of our resources: top - 09:48:32 up 4 days, 12:38, 7 users, load average: 18.06, 16.75, 14.46 Tasks: 142 total, 6 running, 136 sleeping, 0 stopped, 0 zombie Cpu(s): 44.7%us, 50.0%sy, 0.0%ni, 5.1%id, 0.2%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 4057264k total, 3994416k used,62848k free, 279500k buffers Swap:2k total,0k used,2k free, 3044220k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 6537 asterisk -11 0 610m 22m 8760 S 379 0.6 110:33.91 asterisk -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE410P alarms stay RED with 1.4.22
Hi, I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by zap show channels. I tried adding dahdichanname = no to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel 1.4.12.1. -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P alarms stay RED with 1.4.22
On Tue, Nov 11, 2008 at 09:49:14AM +0100, Louis-David Mitterrand wrote: Hi, I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by zap show channels. I tried adding dahdichanname = no to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. OK false alarm here: we use an isdnguard device that needs an additional res_watchdog.c file (bristuff patch). Once added it works. Let's hope 1.4.22 will solve our random crashes and system resources hogging... -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] integration with Microsoft CRM?
Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integration with Microsoft CRM?
On Wed, Jan 21, 2009 at 09:02:51AM -0200, David fire wrote: how hard is to integrate whit a virus? sorry ok i read MS CRM but... did you tried VTiger? www.vtiger.com the next release (5.1) will be integrated whit asterisk not only click to dial and popups on incoming calls a queue monitor system too. (Thanks to Wolfgang) I wasn't aware of VTiger. It looks pretty good. Do you know when 5.1 is supposed to be released? What version of asterisk is required for integration with VTiger? Thanks, -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integration with Microsoft CRM?
On Wed, Jan 21, 2009 at 12:58:51PM -, Andrew Thomas wrote: Try http://forums.vtiger.com/viewtopic.php?t=14314 Thanks, this is a really interesting link. -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integration with Microsoft CRM?
On Wed, Jan 21, 2009 at 09:00:41AM -0500, Jon Weisman wrote: ok what about people that have no choice but to use MS CRM? That's also my concern, as MS CRM is my customer's choice, not ours, and I may or may not succeed in steering them toward an open-source solution such as vTiger. They already looked at (and dismissed) SugarCRM. I am assuming that MS CRM uses TAPI to interface with a third party PBX. In that cas the TAPI page on voip-info.org gives a few (mostly commercial) solutions. In any cas I'd still welcome any pointers or ideas on Microsoft CRM with asterisk. Thanks, -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use? .conf or AEL?
On Tue, Feb 10, 2009 at 01:56:16PM -0800, Mik Cheez wrote: I use them both; my legacy dialplan is all .conf and new stuff is .ael. I find AEL to be the better option when jumping around, but that's just my opinion. But isn't AEL just converted into .conf language anyway? Or has this evolved with 1.4.x ? -- http://www.critikart.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] advice on OrderlyStats (or other cc software)
Hi, Is anyone here using OrderlyStats with asterisk in a call center setting? If so what what is your experience with it? Is that software really free for asterisk users? Or is there a better option out there? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] advice on OrderlyStats (or other cc software)
On Mon, May 04, 2009 at 10:04:53PM +1000, Rob Hillis wrote: Louis-David Mitterrand wrote: Hi, Is anyone here using OrderlyStats with asterisk in a call center setting? If so what what is your experience with it? Is that software really free for asterisk users? Or is there a better option out there? The short answer is OrderlyStats isn't really free for Asterisk. The long answer is that OrderlyStats is free for Asterisk systems with two or less agents. That's really only applicable for the tiniest of call centres. I haven't used OrderlyStats, so I can't speak for the relative merits of it. However, I have used QueueMetrics (which incidentally is /also/ free for call centres of two or less simultaneous agents) and am fairly happy with it. It's not spectacularly pretty - only the latest version has begun to introduce graphs and charts, but it's functional. The price is similar to that of OrderlyStats and the licence you purchase for both of them is time limited - 4 years in the case of QueueMetrics, 5 for OrderlyStats. QueueMetrics will offer a 50% discount for non-profit organisations - I don't know whether OrderlyStats offers the same thing or not. Thank you Rob for the detailed and informative answser. Much appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hfcpci with 1.6 ?
Hi, Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ? What drivers are available? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hfcpci with 1.6 ?
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote: Louis-David Mitterrand schrieb: Hi, Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ? What drivers are available? Digium's BRI cards are also based on Cologne Chip - thus you could try Digiums BRI drivers. http://lists.digium.com/pipermail/asterisk-users/2008-April/208806.html You mean the vzaphfc module included in dahdi ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hfcpci with 1.6 ?
On Tue, Jun 09, 2009 at 04:04:29PM +0300, Tzafrir Cohen wrote: On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote: Louis-David Mitterrand schrieb: Hi, Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ? What drivers are available? mISDN 1.1? mISDN2 (chan_lcr)? chan_dahdi? I tried chan_lcr and it works fine. Just one small problem: callerid's arrive without the 'national' prefix (0). How can I fix that? Digium's BRI cards are also based on Cologne Chip - thus you could try Digiums BRI drivers. http://lists.digium.com/pipermail/asterisk-users/2008-April/208806.html The Digium driver is for a slightly different chip: HFC-4S. mISDN (the various versions) include drivers for it. zaphfc should work with Zaptel. zaphfc has been ported to DAHDI and is reported to crash Asterisk successfully (http://bugs.debian.org/532345 ). Does the digium driver also work for Beronet's (or Junghanns) 4BRI and 8BRI cards? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] optimising asterisk sounds for g722
Hi, After upgrading to 1.6.x and hdvoice (g722) polycome phones I am wondering how to optimize asterisk sounds and music on hold to take advantage of that codec. I often listen to a special music extension on my headset: /usr/bin/wget -q -O - http://music.example.com | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 - I tried doubling the frequency to 16000 but this slows down the music. What should I do to get better music quality while retaining backwards compatibility to g711? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gap between Playback and Queue
Hi,
I have a 2/3 second gap between the end of a welcome message played with
Playback and the start of the Queue music. Here is the dialplan:
exten = ${EXTEN},1,NoOp($EXTEN)
exten = ${EXTEN},n,SIPAddHeader(Alert-Info: Ring_CCC)
exten = ${EXTEN},n,Set(CALLERID(name)=${MYCID})
exten = ${EXTEN},n,Answer()
exten = ${EXTEN},n,Wait,1
exten = ${EXTEN},n,Playback(/usr/local/share/asterisk/sounds/welcome)
;; slight gap (silence) here -
exten = ${EXTEN},n,Queue(ccc|t|||${QUEUEWAITTIME})
The welcome sound does end correctly after the last word.
Any idea?
Thanks,
___
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Re: [asterisk-users] gap between Playback and Queue
On Wed, Jun 17, 2009 at 01:08:33PM -0500, Danny Nicholas wrote: If this is a recorded sound, you might want to truncate it with lame or audacity. It is quite common in my shop as we record using the phones. Thanks for this suggestion. The problem was indeed a silence at the beginning of my musiconhold tracks. Audacity did a fine job and fixed my problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agent login status visual clue on Polycom?
Hi, Is there a way on Polycom phones to show an agent whether he is logged in or not? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP430 sound level too low?
Hello, Has anyone noticed that the Polycom IP430 has a low incoming/outgoing sound level? Is it a firmware issue or should I adjust my zap's tx/rxgain? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] corrupt faxes
Hello, Since our telco messed with our PRI in some way, we get corrupt faxes like these: http://zenon.apartia.fr/stuff/corrupt_fax.pdf We use the lastest asterisk with a TE410P and spandsp. (for some strange reason, our neighbour company has a traditional pbx fed by 7 BRI's and sees the same problem) Now the telco is trying to racket us with some audit to solve the problem. They are claiming our pbx clockrate might be responsible. What could interefere with faxing in such a way? Could the telco have enabled some echo cancellation on their side? Thanks in advance for any insight, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] importance of crc4 in zaptel.conf?
Hello, We have a TE410P connected to an EuroISDN E1 with these span definitions: span=1,1,0,ccs,hdb3 span=2,1,0,ccs,hdb3 span=3,1,0,ccs,hdb3 span=4,1,0,ccs,hdb3 Why should we add crc4 to these definitions? What does it do? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bristuff problem?
Hi Kape, With latest asterisk 1.2.12.1, zaptel 1.2.9.1 and bristuff 0.3.1s after a while calls become stuck: either the caller or callee can't hear the other party, or heavy static is heard. An asterisk restart fixes it for a short while only. This doesn't happen with our older installs (asterisk 1.2.9, zaptel 1.2.7, bristuff 0.3.1q). Are you aware of that problem? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cisco 7960 not registering after * restart
Hello, When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to re-register themselves with asterisk, even though I put timer_register_expires: 60 in SIPDefault.cnf Is there a way to have these phones register themselves every 60 seconds? Alternatively, can asterisk be made to remember its dynamic sip hosts' registration after a restart? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote: That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. Thanks for your helpful answer, What is the cisco part number for the appropriate smartnet contract required to obtain 79XX firmware? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [bristuff] returning a Busy to the telco?
Hi Kape, Life is generally good with bristuff and the quadBRI cards. However I've got a concern: how does one return a busy signal to the telco when all B channels are busy? Right now, when all channels are in use, the remote caller is kept waiting until the telco times out and finally get a busy signal (on a land line) or error tone (mobile). On the asterisk side here is the output: -- Ignoring callwaiting SETUP on channel 0/0 span 4 0 Jul 18 10:21:20 WARNING[27332]: chan_iax2.c:5063 socket_read: midget packet received (1 of 4 min) Jul 18 10:21:22 WARNING[27325]: chan_zap.c:7534 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! going on ... -- Ignoring callwaiting SETUP on channel 0/0 span 4 0 Thanks in advance for your help, Best regards, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ignoring callwaiting?
Hello, I have the exact same question as you. Did you find an answer? We are using asterisk at the office and the incoming line is an ISDN (HFC-PCI card with zap_hfc driver from bristuff 0.2.0 RC3a). And I have a problem, when both ISDN B channels are in use (i.e. 2 calls in progress) it seems that anyone that calls in gets no answer at all, and after 20 seconds or so a voice from the telephone company that says (translated) It is not possible to connect your call at the moment. In the asterisk console I can se the message Ignoring callwaiting SETUP on channel 0/0 span 2 0. I think it would be much better if the caller would get a busy tone... Is there any way to accomplish that? Or is this simply a bug? /Ola -- Only half the people in the world are above average intelligence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 600 one-touch message access?
Hello, With the 1.5.2 firmware, have you managed to get one-touch message access when pressing the Messages button? It worked for me with 1.4.1 but no longer with 1.5.2: I have to go through the message count screen first. In phone.cfg I have: msg msg.bypassInstantMessage=1 and in sip.cfg: user_preferences up.headsetMode=0 up.useDirectoryNames=0 up.oneTouchVoiceMail=1 up.welcomeSoundEnabled=1 up.welcomeSoundOnWarmBootEnabled=0 up.localClockEnabled=1/ Have I forgotten a setting? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom 600 one-touch message access?
On Mon, Jul 25, 2005 at 09:38:29AM -0400, Noah Miller wrote: With the 1.5.2 firmware, have you managed to get one-touch message access when pressing the Messages button? It worked for me with 1.4.1 but no longer with 1.5.2: I have to go through the message count screen first. In phone.cfg I have: msg msg.bypassInstantMessage=1 In the phone.cfg file under the above line, make sure you also have: mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi. 1.callBack=Your-VM-Exten ... Yes, I have that setup too (no change from 1.4.1) Are you saying one-touch voicemail works for you with 1.5.2 ? (meaning no message count summary screen when pressing Messages) -- You can't shake the Devil's hand and say you're only kidding. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] probing a SIP device for redirection information?
Hello, I'd like to find a way to probe a SIP phone for forwarding information before I actually Dial() it. For instance, if an absent user entered a forwarding number in his (Cisco or Polycom) phone, it will anwser a Dial() with a REDIRECT and asterisk will comply if the context allows. However I'd like to intercept that REDIRECT, get the number and use the telco's call deflection service to deflect the call without answering it. That way, once the deflection is done, no telco lines are used. Whereas with asterisk's default behavior, two lines would be used: one for the incoming call and one for the forwarded call. I've looked in the docs without finding, for a way to get a SIP device's status: will it forward the call? if so to what number? Thanks for your insights, -- Brains x Beauty x Availability = Constant. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7920 boot causes 7940 to release DHCP lease
On Thu, Aug 11, 2005 at 07:12:32PM +0100, Mark Thorpe wrote: I have been trying to solve a problem wherby when I boot a cisco 7920 my 7940 seeks a new IP and the dhcpd log shows it released its existing IP. In searching for the solution I notice there were 2 messages on this list in Aug Sep 2004 which raised the problem, but I could not find any answer was posted. I've noticed the same issue. Haven't tried it with 79[46]0's 7.5 firmware. A workaround would be to give a static addres to your 7920. Let me know if you find a solution. -- I have no special talents. I am only passionately curious. --Albert Einstein ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Paul Hales wrote: I battled with chan_capi during the week, and it was not fun. Since I'm working on chan_capi, I would like to know what problems exist. Can you please be more specific on what problems you have encountered? It's good to see a capi expert working on that module! Could you elaborate on your plans for chan_capi? Cheers, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
On Fri, May 13, 2005 at 12:33:10PM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Louis-David Mitterrand wrote: On Fri, May 13, 2005 at 09:55:58AM +0200, Armin Schindler wrote: On Fri, 13 May 2005, Paul Hales wrote: I battled with chan_capi during the week, and it was not fun. Since I'm working on chan_capi, I would like to know what problems exist. Can you please be more specific on what problems you have encountered? It's good to see a capi expert working on that module! Could you elaborate on your plans for chan_capi? Currently I have three topics: - fix the problem with kernel 2.6 (looks like a CAPI problem) You mean the loud cracking, poping and line crosstalk on SMP-enabled kernels I notified you about? - cleanup in chan_capi.c (I noticed some errors) - add native bridging using CAPI Line-Interconnect Nice. Then I hope to receive some reports on what is buggy/not working, wishlist and hopefully also some reports on what works well. ECT (explicit call transfer) seems broken (last time I tried it). I also was thinking about an application for receiving fax over CAPI, but I'm not yet familiar with the current asterisk fax support, so I need to learn more here. Maybe some else can inlight me here... Faxing in asterisk is Steve Underwood's ([EMAIL PROTECTED]) specialty. -- Slight disorientation after prolonged system uptime is normal for new Linux users. Please do not adjust your browser. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 4 port BRI options ?
On Fri, Jun 03, 2005 at 02:39:48PM +0100, Gavin Hamill wrote: On Friday 03 June 2005 14:28, Nardis Dome wrote: --- Brett, Gary [EMAIL PROTECTED] wrote: Is the Eicon that much better ? sorry, i have only experience with Eicon... maybe someone else is able to give a feedback... Aside from paying for a recognised brand name, with Eicon you get on-board DSPs and firmware, as well as software to make the card appear as a series of /dev/tty devices (and/ or CAPI) under Linux. With kernel 2.6 you only get ISDN tty devices through the capidrv module, no more analog modem login support with the obsoleted diva2i4l module. The Junghanns BRI boards have no such on-board logic, they are pure telephone interfaces driven by the host - and as such they are /perfect/ for Asterisk. You get benefits from the zaptel interface (zapscan, zapbarge, etc.), closer to the metal, less latency. We use an Eicon Diva Server 4BRI for our fax server because of the TTY interface for Hylafax - the DSPs themselves do all the fax negotiation / compression and it works extremely well. It works, but when ISDN channels are busy faxing, asterisk has no way to know which outgoing channel is free: users get a congestion when trying to call out (unless you program some channel testing logic/loop inside your dialplan). This is a major bummer in production. However, we're also running * on that machine via CAPI, and the board gives us the flexibility to share 8 channels between * and Hylafax really easily :) There is no sharing, they contend for the channels, even thought they both use the capi layer. Go figure. My experience with both cards is that the Diva is a waste of money. It's better to go with the Junghanns card and do faxing from inside asterisk, which works very well in the latest versions. -- They can have my jeans, as soon as they pry them off my dead, cold ass. -- (Spackler on /. about corporate dress code becoming formal again) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] localize ${VM_DATE} ?
Hello,
I looked everywhere in the docs and in google but couldn't find an
answer.
Is it possible to localize the output of ${VM_DATE} (say, in french) ?
--
Only half the people in the world are above average intelligence.
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[Asterisk-Users] Re: Console ALSA Sound
On Fri, Jun 17, 2005 at 10:34:25PM +0200, Conrad Beckert wrote: ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference FM is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. It's not muting your speakers, it's locking your sound device. Two solutions: - noload chan_alsa, - provide a virtual mixer device in alsa.conf instead of a real hardware device, see http://alsa.opensrc.org/index.php?page=DmixPlugin for details, -- Turn off your computer and go outside ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oneTouchVoicemail issue with Polycom 1.5.2
Hi, After upgrading to 1.5.2 I no longer can directly access to my voicemail by pressing the Message button, I have to go through the urgent,new,old report first. The oneTouchVoicemail parameter is set to 1 but not taken into account apparently. Anyone noticed that problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
