Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks!
G. ----- Original Message ----- From: <[EMAIL PROTECTED]> To: <asterisk-users@lists.digium.com> Sent: Monday, July 18, 2005 2:00 PM Subject: Asterisk-Users Digest, Vol 12, Issue 117 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Asterisk Comedian Web page login (Kurt Pasewaldt) 2. Asterisk @ Home incoming CID ([EMAIL PROTECTED]) 3. massive outbound calling... (Goolsby, Daniel S (Daniel)) 4. Re: SpanDSP+astfax with multiple fax pages (Lee Howard) 5. Re: System Jsut hangs Up (sylvain garcia) 6. Re: Iaxy and Echo (Adam Goryachev) 7. Re: Asterisk/Hylafax <=> Receive/Send faxes (Lee Howard) 8. Re: Teliax to VoIPJet (Andrew Latham) 9. IP Trunking for LD? (Matthew S. Krawitz) 10. Re: swissvoice (Doug Lytle) 11. Re: Iaxy and Echo (Aaron with Morad) 12. long pause on dialing.. (Goolsby, Daniel S (Daniel)) 13. Comments on Areski Calling Card Solution plz (Arnd Vehling) 14. IAX register confusion (David Cook) 15. Transcoding problems (Martin Sutherland) 16. Re: [EMAIL PROTECTED] not accepting IAX calls from outside (Mark Phillips) 17. Codecs and bandwidth (Tim Pushor) 18. RE: Teliax to VoIPJet (Wiley Siler) 19. Re: long pause on dialing.. (Randy Williams) 20. RE: swissvoice (Florian Overkamp) 21. Re: long pause on dialing.. (Giorgio Incantalupo) 22. Re: Memory leak in asterisk CVS (Erik Espinoza) 23. Re: SoftPhones: Bad, or just bad QoS? (Time Bandit) 24. Re: long pause on dialing.. (Randy Williams) ---------------------------------------------------------------------- Message: 1 Date: Mon, 18 Jul 2005 11:29:23 -0400 From: Kurt Pasewaldt <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Asterisk Comedian Web page login To: Asterisk <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 When I try to login into voicemail through the web interface It states incorrect login. In my voicemail.conf I have all voicemail boxes set under local. I changed the symbolic link to reflect the new directory under /var/spool/asterisk. Am I missing something? My vm link = /var/spool/asterisk/voicemail/local. Kurt ------------------------------ Message: 2 Date: Mon, 18 Jul 2005 15:38:12 +0000 From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk @ Home incoming CID To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED] .com> OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can receive and place calls with no issues, however, when I receive a call, the CID only shows "Analog Line" on the Grandstream 2000XP phone. Does anyone have any ideas even where to look to change this?? Is it a setting in the phone, Asterisk, or both?? Thanks, Marc ------------------------------ Message: 3 Date: Mon, 18 Jul 2005 10:40:25 -0500 From: "Goolsby, Daniel S (Daniel)" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] massive outbound calling... To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii Does anyone know what kind of limitations asterisk has when it comes to massive outbound dialing.. i.e. how many sip/iax phones could be dialed at the same time-- and if someone answered, play a .wav file? Or outbound throughput on zaptel devices? Say if I had a dual xeon with 2 quad t1 cards, hooked up to a 100mbit lan. Anyone know how many it could actually sustain w/o the voice file being distorted on playback? Daniel ------------------------------ Message: 4 Date: Mon, 18 Jul 2005 08:38:19 -0700 From: Lee Howard <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] SpanDSP+astfax with multiple fax pages To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Paul van Brouwershaven wrote: > HylaFAX can (we ar doing this now), but not with E1 or T1. So you can > only send with a maximum 2 channels. (with two default analog modems) HylaFAX can use E1 and T1 fax modems just fine (24 and 30 channels each). Furthermore, HylaFAX also supports multiport modems (usually up to 8 ports each). But, yes, if you are only going to use internal ISA or external modems connected to the motherboard's serial ports, then yes, you're limited to two modems. But that's completely ignoring the possibilities of using PCI and USB modems. Lee. ------------------------------ Message: 5 Date: Mon, 18 Jul 2005 17:39:45 +0200 From: sylvain garcia <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] System Jsut hangs Up To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Tim King a écrit : > I took care of my earlier problem. But now if I call in it just says > goodbye, And on my extension no matter what I do it seems to just hang > up on me immediately. It's a slackware 10.1 box with Digium 22b card. > I am running AMP so its mysql driven. I'm not seeing any errors. It > just hangs up. > > > > Tim King > > Network Engineer > > Computer & Network Solutions LLC > > 1331 Plainfield Ave > > Grand Rapids MI 49505 > > > > Phone: 800-669-3290 > > > >------------------------------------------------------------------------ > >_______________________________________________ >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 > Could you describe your problem with your extensions.conf And send me your email please sorry for my english i'm french -------------- next part -------------- Skipped content of type multipart/related ------------------------------ Message: 6 Date: Tue, 19 Jul 2005 01:46:09 +1000 From: Adam Goryachev <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Iaxy and Echo To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain On Mon, 2005-07-18 at 09:25 -0600, Aaron with Morad wrote: > I have been searching for a while and can't find anything specific > like this. > > Here's is my setup: > > IAXy -- broadband network -- Asterisk -- TE110P -- Channel > Bank -- POTS lines (FXO) > > Everything works fine except for the echo at the IAXy. There is no > echo on the POTS end, so Asterisk is doing a good job of echo > canceling. Is there any provisioning in the IAXy to do echo > canceling? If you get echo on the IAXy end, then asterisk is NOT doing it's echo cancellation function fully. The POTS user will never get echo if you don't generate any, or their echo cancellation is functioning correctly. So, you need to tune the echo cancellation at your asterisk box (or perhaps you can do that in your channel bank?? I dunno how clever those things are)... Regards, Adam ------------------------------ Message: 7 Date: Mon, 18 Jul 2005 08:47:37 -0700 From: Lee Howard <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Asterisk/Hylafax <=> Receive/Send faxes To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Jian Hong GUAN wrote: > Can you tell me how to configure Hylafax + Asterisk in order to be > able to receive/send faxes. If you have an incoming T1/E1 line: telco --> T1/E1 --> TE405P/TE410P --> Asterisk --> TE405P/TE410P (another port) --> T1/E1 fax modem --> HylaFAX or: telco --> T1/E1 --> TE405P/TE410P --> Asterisk --> TE405P/TE410P (another port) --> channel bank --> analog fax modem --> HylaFAX If you don't have an incoming T1/E1 line (you've just got analog lines coming in) then just get yourself another analog line for your analog fax modem and bypass Asterisk altogether. Look up the archives for "TDM" and "fax" to get a synopsis as to why you don't want to run fax through Asterisk on analog channels. Lee. ------------------------------ Message: 8 Date: Mon, 18 Jul 2005 10:50:34 -0500 From: Andrew Latham <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Teliax to VoIPJet To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 HUH? Why? If you are having Cellphones dialed for the user its one thing but what is the goal.... On 7/18/05, code select <[EMAIL PROTECTED]> wrote: > I'm trying to setup asterisk to accept call from Teliax, request the > 10 digit number from user, then dial it thru the VoIPJet. If I'm not > wrong I will be charged by both providers because both connection is > active during conversation. So my question is can I set the things so > that I pay only to VoIPJet? Specific configuration snippets will be > greatly appeciated. > > Thank you. > _______________________________________________ > 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 > -- <sig> Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! </sig> ------------------------------ Message: 9 Date: Mon, 18 Jul 2005 11:57:55 -0400 From: "Matthew S. Krawitz" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] IP Trunking for LD? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" I'm sure this topic has been discussed to death, but I haven't found a comprehensive answer yet... I have a very large installation of Cisco Call Managers connecting directly local and LD T1's for service. I would like to replace some of my LD T1's with IP trunks (or something like that). I would need low-cost domestic and international LD... but quality and reliability is our top priority, so we're not simply looking for low-bid. I assume IAX2 is the protocol I should use, but who are the major players in providing this type of service? Thanks! - matthewk (MSK2) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050718/74d939 f0/attachment-0001.htm ------------------------------ Message: 10 Date: Mon, 18 Jul 2005 12:04:49 -0400 From: Doug Lytle <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] swissvoice To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed thomas DEILLON wrote: >Hello, > >I have swissvoice phones and when i use one, a have in asterisk lines like: >Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp >-13691.-232125 > >have a idea ? > > > > Yes, Kevin said this earlier today: 2 wrote: > i get lots of the below from friday 15.7.05 cvs as well > > ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... I will be looking into this issue later today. Doug ------------------------------ Message: 11 Date: Mon, 18 Jul 2005 10:04:23 -0600 From: "Aaron with Morad" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Iaxy and Echo To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original Thanks Adam. My channel banks are pretty old (NEC ND4's) so they don't do anything for echo. I'll have to try tweaking Asterisk. Aaron ----- Original Message ----- From: "Adam Goryachev" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, July 18, 2005 9:46 AM Subject: Re: [Asterisk-Users] Iaxy and Echo > On Mon, 2005-07-18 at 09:25 -0600, Aaron with Morad wrote: >> I have been searching for a while and can't find anything specific >> like this. >> >> Here's is my setup: >> >> IAXy -- broadband network -- Asterisk -- TE110P -- Channel >> Bank -- POTS lines (FXO) >> >> Everything works fine except for the echo at the IAXy. There is no >> echo on the POTS end, so Asterisk is doing a good job of echo >> canceling. Is there any provisioning in the IAXy to do echo >> canceling? > > If you get echo on the IAXy end, then asterisk is NOT doing it's echo > cancellation function fully. The POTS user will never get echo if you > don't generate any, or their echo cancellation is functioning correctly. > > So, you need to tune the echo cancellation at your asterisk box (or > perhaps you can do that in your channel bank?? I dunno how clever those > things are)... > > Regards, > Adam > > > _______________________________________________ > 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 > ------------------------------ Message: 12 Date: Mon, 18 Jul 2005 11:08:19 -0500 From: "Goolsby, Daniel S (Daniel)" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] long pause on dialing.. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii I have an Asterisk setup with AMP installed. I have phone extensions from 7000 to 7010. I experience long delays when dialing a 9 digit number as opposed to a 10-digit number. How do you get around not having to press the # key to speed up the dialing process? For any length phone number for that matter-- like dialing another extension. If I dial 7005, I'll have to wait a while.. but it's instant when I press the # key. Daniel ------------------------------ Message: 13 Date: Mon, 18 Jul 2005 18:17:37 +0200 From: Arnd Vehling <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Comments on Areski Calling Card Solution plz To: Asterisk Users <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd ------------------------------ Message: 14 Date: Mon, 18 Jul 2005 12:03:28 -0400 From: David Cook <[EMAIL PROTECTED]> Subject: [Asterisk-Users] IAX register confusion To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 I have been unable to understand the connection between an IAX registration for dynamic IP assignment and and the host definition. I have signed up with an ITSP for a DID. My ip is dynamic and although I have a dynamic DNS name, we are registering and outbound works fine. I'm at a loss to understand the relationship between the registration and the [section] definition in iax.conf that will allow me to specify my context for inbound calls and deal with the inbound DID. For example: register => myuser:[EMAIL PROTECTED] ;OK. This part works fine. My dial statement calls ; exten => _NXXNXXXXXX,3,Dial,IAX2/myuser:[EMAIL PROTECTED]/${EXTEN},45,tr) ; ; VoIP Local service from myitsp ;[something] ??? [LO_TRNK_MYSWITCH] type=peer host=dynamic context=from-myitsp secret=mypasswd qualify=3000 ; How do I construct this entry? I would _like_ the entry to be labelled ; LO_TRNK_MYSWITCH so I can maintain a naming convention that makes ; sense. ; How do I associate this with the inbound itsp so the calls come into ; the "s" extension in a particular context so I can deal with the DID? I simply don't see how I associate the inbound stream with my section heading? Thanks, dbc. -- David Cook ------------------------------ Message: 15 Date: Mon, 18 Jul 2005 17:14:18 +0100 From: "Martin Sutherland" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Transcoding problems To: <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII I have just purchased 20 licenses for the G729a codec from digium and set about changing the defaults to use this codec in all cases to reduce the bandwidth requirements (all my SIP devices support this codec). To my dismay I then find that calls coming from SIP devices to the outside via the chan_capi channel no longer work and give the message "no translator path exists for channel type CAPI (native 8) to 256". I understood that Asterisk would transcode between different codecs? In fact only two of my SIP users are allowed access to the outside via a BRI interface, but I now have to set them to always use g711a/u just in case they make a call via the chan_Capi. I am using chan_capi-cm-0.5.3 ------------------------------ Message: 16 Date: Mon, 18 Jul 2005 12:21:47 -0400 From: Mark Phillips <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] not accepting IAX calls from outside To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Problem solved. Wrong context supplied by me - doh!! Mark Phillips wrote: > I've been banging my head with this all day. > > I today switched from a very old CVS build to AAH1.3 and so far > everything has been easy. However I cannot accept calls from a > previously working IAX trunk. > > I've set up an trunk with all the same credentials as before and can > call the folks at the other pbx. However whenever they call me I tell > them that I don't have an extension/context by the name they dialed. > > Any ideas? > -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ------------------------------ Message: 17 Date: Mon, 18 Jul 2005 10:22:27 -0600 From: Tim Pushor <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Codecs and bandwidth To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Friends, Something I'd like to shed some light on if possible - how is it that a single ISDN conversation only uses 64K for bidirectional communication (using ulaw, correct?), but on several occasions now have seen references to ulaw voip conversations using 64K per side of the conversation, plus packet overhead (http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down now - plus other references) for a total of over 128K per ulaw 'full duplex' voice conversation? Thanks Tim ------------------------------ Message: 18 Date: Mon, 18 Jul 2005 09:34:11 -0700 From: "Wiley Siler" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Teliax to VoIPJet To: "Andrew Latham" <[EMAIL PROTECTED]>, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="US-ASCII" This sounds like DISA which is great for saving bucks on LD if used right... You will still need two channels and thus it will still cost for both legs... Nature of the beast... W -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, July 18, 2005 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Teliax to VoIPJet HUH? Why? If you are having Cellphones dialed for the user its one thing but what is the goal.... On 7/18/05, code select <[EMAIL PROTECTED]> wrote: > I'm trying to setup asterisk to accept call from Teliax, request the > 10 digit number from user, then dial it thru the VoIPJet. If I'm not > wrong I will be charged by both providers because both connection is > active during conversation. So my question is can I set the things so > that I pay only to VoIPJet? Specific configuration snippets will be > greatly appeciated. > > Thank you. > _______________________________________________ > 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 > -- <sig> Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! </sig> _______________________________________________ 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 ------------------------------ Message: 19 Date: Mon, 18 Jul 2005 12:34:51 -0400 From: Randy Williams <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] long pause on dialing.. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, "Goolsby, Daniel S (Daniel)" <[EMAIL PROTECTED]> Cc: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 Greetings, This may be an artifact of the particular phone you are using. I know that both Grandstream and SNOM products allow you to set a timeout for "auto-dial" (which is how long to wait after the last button press before dialing the number present). I have my units set to three settings: 2 seconds for the receptionist 5 seconds for most everyone else 30 seconds for some of our elder employees who need extra time while transcribing a phone number Check your phone settings to see if there is something you can set. However, there may be something else at fault... RandyW Quoting "Goolsby, Daniel S (Daniel)" <[EMAIL PROTECTED]>: > I have an Asterisk setup with AMP installed. I have phone extensions > from 7000 to 7010. > > I experience long delays when dialing a 9 digit number as opposed to a > 10-digit number. How do you get around not having to press the # key to > speed up the dialing process? For any length phone number for that > matter-- like dialing another extension. > > If I dial 7005, I'll have to wait a while.. but it's instant when I > press the # key. > > Daniel > > _______________________________________________ > 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 > ------------------------------ Message: 20 Date: Mon, 18 Jul 2005 18:32:24 +0200 From: "Florian Overkamp" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] swissvoice To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="US-ASCII" Hi, > -----Original Message----- > I have swissvoice phones and when i use one, a have in > asterisk lines like: > Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning > negative timestamp > -13691.-232125 > the swissvoice firmware is IP10 SP v1.0.0 (Build 11) and > asterisk version is > the cvs of 18 july 2005 (today). Swissvoice phones tend to have a few interesting side effects in their rtp timestamping, we have filed some issues on that. However, it would be fun to hear what the actual problem is you are experiencing :-) Best regards, Florian ------------------------------ Message: 21 Date: Mon, 18 Jul 2005 18:40:21 +0200 From: Giorgio Incantalupo <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] long pause on dialing.. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, it is hard to answer without the right piece of extensions.conf but remember there is a digit timeout in Asterisk: you enter 9 digits but in the dialplan there is a match for 10 digits so how can Asterisk know if you want to call the 9-digits number or the 10-digits? After 9 digits it waits for a while...if another digit is dialed then it can call the 10-digit number otherwise it calls the 9-digits number. You can lower Asterisk digit timeout but remember that not all users are so fast to dial... Giorgio. Goolsby, Daniel S (Daniel) wrote: >I have an Asterisk setup with AMP installed. I have phone extensions >from 7000 to 7010. > >I experience long delays when dialing a 9 digit number as opposed to a >10-digit number. How do you get around not having to press the # key to >speed up the dialing process? For any length phone number for that >matter-- like dialing another extension. > >If I dial 7005, I'll have to wait a while.. but it's instant when I >press the # key. > >Daniel > >_______________________________________________ >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 > > > ------------------------------ Message: 22 Date: Mon, 18 Jul 2005 09:29:47 -0700 From: Erik Espinoza <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Memory leak in asterisk CVS To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 Hi Walter, I had high load and extreme memory usage on my machine. My machine wasn't running on SMP. My point was that the cvs version you were using contained some bad patches, and it was probably a good idea to upgrade or move to stable. Thanks, Erik On 7/18/05, Walter Klomp <[EMAIL PROTECTED]> wrote: > Hi Erik, > > You put me to a page which refers to high load on CPU on SMP. Nothing to do > with memory leak. Furthermore I am not running SMP. > > Any other suggestions in which direction to look? Am I the only one > experiencing this ? > > Do you mean if I update to the today's CVS the memory leak issue will be > resolved ? > > Thanks > Walter > > --- Original Message below --- > > Message: 20 > Date: Sat, 16 Jul 2005 21:42:44 -0700 > From: Erik Espinoza <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Memory leak in asterisk CVS > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Known issue. This was reverted later. > > Check the thread on the mailing list > > http://lists.digium.com/pipermail/asterisk-users/2005-July/116246.html > > Thanks, > Erik > > On 7/16/05, Walter Klomp <[EMAIL PROTECTED]> wrote: > > Hi, > > > > My Asterisk CVS is apparently not doing much (other than keeping SIP & > > IAX2 registrations alive and doing some ZAP calls (without > > echo-cancellation), but slowly the memory is filling up, so much so that > > 100m virtual memory is used up within 12 hours and I have to restart the > > asterisk application every 48 hours to make sure I have enough memory... > > > > How can I help resolve this problem? > > > > Problem occurs on both Sangoma and Digium installed systems. Fedora Core > > 3 and Centos 4.1 don't make a difference either. > > > > My version is Asterisk CVS-HEAD built on a i686 running Linux on > > 2005-07-11 16:29:02 > > > > I have removed the mailbox entries in my sip.conf which greatly reduced > > this problem. So, I suspect it may be in the sip or iax channel > application. > > > > I also run quite a bit of agi scripts but none of them were "alive" when > > these memory-usage increases as shown below over a 1 minute interval > > with only 4 zap channels alive (2 calls) occured: > > > > ps -AF output... using this script: > > n=1;while [ 1 ]; do i=`ps -AF|grep ast|grep sbin|grep -v grep`; m=`echo > > $i|cut -f 6 -d\ `;if [ `echo $m` -ne `echo $n` ]; then echo $i; n=`echo > > $m`;fi;done > > > > root 15875 26881 0 15727 46240 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46248 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46256 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46268 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46280 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > root 15875 26881 0 15725 46288 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp > > > > Hope we can fix this somehow. > > > > Walter Klomp > > Singapore. > > > > > > > > > _______________________________________________ > 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 > ------------------------------ Message: 23 Date: Mon, 18 Jul 2005 12:54:41 -0400 From: Time Bandit <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] SoftPhones: Bad, or just bad QoS? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 > This is software. Use manageble software. If "software" means separate > setup on each desktop, then don't use it. If you spend that much time on > setting up phones, imagine how long it takes you to update other > software packages. This is, then, a symptom of a general problem. I would like to implement central management in my softphone. What would be the best way to accomplish this ? Currently, all the settings are stored in the registry under HKEY_CURRENT_USER. So, if you use a roaming profile, the settings follow you. I would appreciate people's input on what would be desirable, and I'll try to implement it so it would be more easy to manage. Thanks ------------------------------ Message: 24 Date: Mon, 18 Jul 2005 12:34:51 -0400 From: Randy Williams <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] long pause on dialing.. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, "Goolsby, Daniel S (Daniel)" <[EMAIL PROTECTED]> Cc: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 Greetings, This may be an artifact of the particular phone you are using. I know that both Grandstream and SNOM products allow you to set a timeout for "auto-dial" (which is how long to wait after the last button press before dialing the number present). I have my units set to three settings: 2 seconds for the receptionist 5 seconds for most everyone else 30 seconds for some of our elder employees who need extra time while transcribing a phone number Check your phone settings to see if there is something you can set. However, there may be something else at fault... RandyW Quoting "Goolsby, Daniel S (Daniel)" <[EMAIL PROTECTED]>: > I have an Asterisk setup with AMP installed. I have phone extensions > from 7000 to 7010. > > I experience long delays when dialing a 9 digit number as opposed to a > 10-digit number. How do you get around not having to press the # key to > speed up the dialing process? For any length phone number for that > matter-- like dialing another extension. > > If I dial 7005, I'll have to wait a while.. but it's instant when I > press the # key. > > Daniel > > _______________________________________________ > 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 12, Issue 117 *********************************************** _______________________________________________ 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