[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.

Hi Group. I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing registration it registered fine on asterisk as peer is available in Database. But now i am doing 'sip reload' or 'reload' due to

Re: [asterisk-users] Is this doable?

On Wednesday, February 8, 2012, Josh wrote: > >> I don't get this. Didnt EVERYONE know it's insecure? > > Can you read? Can everyone? > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to

[asterisk-users] criteria for setting registration expiration

Hi, Are there any guidelines/recommended values for setting the registration expiration and subscription expiration for SIP phones? The default values for those settings on our phones are 60 secs. Any disadvantages for making them longer; e.g. 300 secs or more? Thanks, Matt

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

> >No, unfortunately that's not quite correct. The UPGRADE files list > >*important* changes that users need to know about because they are > >changes in behavior of existing functionality. New features, even > >really > >useful and widely anticipated ones, that don't cause backwards > >compatibili

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On Thu, 09 Feb 2012 02:31:00 +0100 Gilles wrote: > On Wed, 08 Feb 2012 15:58:43 -0600, "Kevin P. Fleming" > wrote: > >No, unfortunately that's not quite correct. The UPGRADE files list > >*important* changes that users need to know about because they are > >changes in behavior of existing func

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On Wed, 08 Feb 2012 15:58:43 -0600, "Kevin P. Fleming" wrote: >No, unfortunately that's not quite correct. The UPGRADE files list >*important* changes that users need to know about because they are >changes in behavior of existing functionality. New features, even really >useful and widely anti

Re: [asterisk-users] Is this doable?

I don't get this. Didnt EVERYONE know it's insecure? Can you read? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Is this doable?

On Wednesday, February 8, 2012, Josh wrote: > >> http://www.asterisk.org/astdocs/node66.html > > Thanks, never knew that! > >> Yes, I understand that it's not what you want, but that doesn't make it a security concern. If Asterisk is publicly available on one interface, making it available on ano

Re: [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles

On 02/08/2012 04:02 PM, Danny Nicholas wrote: Not a complaint, per se, just a question. Why are the LTS versions "odd" (11, 13, 15, etc) and the non-LTS (10, 12, etc) even? As I read the chart, Digium/Asterisk is committing to a new LTS version every 2 years? Well, the first LTS was Asterisk

Re: [asterisk-users] T38 faxing - UDPTL creation failed

On 02/02/2012 12:24 PM, Danny Nicholas wrote: Agreed - I think the "solution" is a patch to udptl.c to reset the counter instead of dying with this message (just my opinion). Asterisk was recently changed to only allocate UDPTL port numbers to channels that actually need them; prior to this ch

Re: [asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles

Not a complaint, per se, just a question. Why are the LTS versions "odd" (11, 13, 15, etc) and the non-LTS (10, 12, etc) even? As I read the chart, Digium/Asterisk is committing to a new LTS version every 2 years? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:

Re: [asterisk-users] res_http_post.so questions

On 02/07/2012 11:41 AM, Josh wrote: The primary goal was to upload audio for IVRs in the Asterisk GUI. Thanks, if I don't use the GUI is it safe to exclude it from the build (it is just that I want to avoid a bunch of other dependencies which come with that module)? Yes, you can exclude it.

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On 02/07/2012 09:07 AM, Gilles wrote: On Tue, 7 Feb 2012 14:31:31 +, Steven Howes wrote: The upgrade files may be more to your tastes than changes files. Thanks. I downloaded and untarred asterisk-1.8.8.0.tar.gz, and it looks like the UPGRADE*.txt files within tarballs are the closest th

[asterisk-users] (last call for comments) Proposed changes to Asterisk release and support cycles

I've created a page on wiki.asterisk.org outlining some changes we're proposing to make to the Asterisk release and support cycles. As always, before implementing any changes of this type, we'd like to collect some community feedback on the proposal. The page is here: https://wiki.asterisk.org

[asterisk-users] HT286 dialtone and ring cadence

Hi all, I have a user that has reported that his HT286 doesn't have a ring tone; it just buzzes. Also, the ring doesn't sound right. I can see in the provisioning file where this is configured, but I'm not a musician so I don't know what to put in for a value. Does anyone have suggestions for

[asterisk-users] Problem callerid ignored by using callfiles

hi all, the same problem for a long time. i upgraded now from 1.8.8.2 to 1.8.9.1. I generate a callfile with the option: Callerid: test Callback Service <00498925007676> The callback is established correctly, but the variable ${CALLERID(num):} is empty no output: NoOp(Calleridnumber: ${CALLER

Re: [asterisk-users] Automatic Number Identification and anonymous calls

On 02/08/2012 12:40 PM, Olivier wrote: 2012/2/8, Kevin P. Fleming: On 02/08/2012 10:06 AM, Carlos Alvarez wrote: On Wed, Feb 8, 2012 at 2:35 AM, Oliviermailto:oza_4...@yahoo.fr>> wrote: I always thought that ANI (Automatic Number Identification) could not directly be set or changed

Re: [asterisk-users] Automatic Number Identification and anonymous calls

2012/2/8, Kevin P. Fleming : > On 02/08/2012 10:06 AM, Carlos Alvarez wrote: >> >> On Wed, Feb 8, 2012 at 2:35 AM, Olivier > > wrote: >> >> I always thought that ANI (Automatic Number Identification) could not >> directly be set or changed by end users. >> In t

Re: [asterisk-users] SIP hardware phones

Let me answer that, Carlos. A big hospital. These big infrastructures can be quite outdated and messy. Getting someone to cable old parts of the buildings can be very expensive. However, replacing just the backbone switches is something they can afford. And they don't need PoE, really. What kin

[asterisk-users] finally, an open alternative to Viber

The rise of Viber has, for some people, been a case of `skype, not again!' until now... Lumicall is now in the Android market - and it fully interacts with other SIP products using ENUM and SRV records. Any feedback on this is welcome. Interconnect (Asterisk calling to/from Lumicall users) is

Re: [asterisk-users] Is this doable?

http://www.asterisk.org/astdocs/node66.html Thanks, never knew that! Yes, I understand that it's not what you want, but that doesn't make it a security concern. If Asterisk is publicly available on one interface, making it available on another interface doesn't make you less secure. You lo

Re: [asterisk-users] Automatic Number Identification and anonymous calls

On 02/08/2012 10:06 AM, Carlos Alvarez wrote: On Wed, Feb 8, 2012 at 2:35 AM, Olivier mailto:oza_4...@yahoo.fr>> wrote: I always thought that ANI (Automatic Number Identification) could not directly be set or changed by end users. In this experiment, it seems that if an end user cal

Re: [asterisk-users] Calling a group of phones and force the speaker

> > On Wed, Feb 8, 2012 at 12:48 PM, Danny Dias wrote: > >> Hi, >> >> I wonder, if there is a way to call from A phone to a group of phones (B, >> C and D) and force these phones to activate automatically the speaker >> >> Is that possible? >> >> Many thanks in advance >> >> I've done this quite a

Re: [asterisk-users] Automatic Number Identification and anonymous calls

On Wed, Feb 8, 2012 at 2:35 AM, Olivier wrote: > I always thought that ANI (Automatic Number Identification) could not > directly be set or changed by end users. > In this experiment, it seems that if an end user calls anonymously, > the ANI is also hidden to the receiving party. > I never work

Re: [asterisk-users] SIP hardware phones

On Wed, Feb 8, 2012 at 8:12 AM, Patrick Lists < asterisk-l...@puzzled.xs4all.nl> wrote: > >> Is this everyone else's experience as well? >> > > No the opposite. I have never heard of troubleshooting problems when using > the switch on the phone. Maybe cheap crappy phones give you problems but I >

Re: [asterisk-users] Asterisk V/s FreeSwitch

It's 4 core Intel(R) Xeon(R) CPUX3220 with 6GB RAM Regards, Zohair Raza On Wed, Feb 8, 2012 at 5:46 PM, Bryant Zimmerman wrote: > Zohair > > What kind of hardware spec are you running CPU, MEM, Drives? > > Thanks > > Bryant Zimmerman (ZK Tech Inc.) > 616-855-1030 Ext. 2003 > > > --

Re: [asterisk-users] Asterisk V/s FreeSwitch

On 08-02-12 14:54, Jeff Brower wrote: Brynjolfur- According to this article here: http://anders.com/cms/266 the difference mainly lies in how FreeSwitchs handles open channels in comparison with Asterisk. FS uses one thread per channel while * keeps jumping between threads. At least that's ho

Re: [asterisk-users] SIP hardware phones

On 08-02-12 14:37, Jason W. Parks wrote: From everything I've researched to date, my understanding is most locations have chosen to double their port density and continue to service the phone and computer on separate ports than to share a single line for both computer and phone. Reason primarily

Re: [asterisk-users] SIP hardware phones

If the customer is so cheap that they won't properly build out the network, why would they have gigabit switches to the desktop which have a limited set of applications that actually benefit from it? Then there's PoE, which is expensive to start and very expensive with gigabit. So this mythical c

Re: [asterisk-users] Calling a group of phones and force the speaker

On Wed, Feb 8, 2012 at 12:48 PM, Danny Dias wrote: > Hi, > > I wonder, if there is a way to call from A phone to a group of phones (B, > C and D) and force these phones to activate automatically the speaker > > Is that possible? > > Many thanks in advance > > Hi Danny, http://www.voip-info.org

Re: [asterisk-users] SIP hardware phones

--- On Wed, 2/8/12, Jason W. Parks wrote: > From everything I've researched to > date, my understanding is most > locations have chosen to double their port density and > continue to > service the phone and computer on separate ports than to > share a single > line for both computer and phon

Re: [asterisk-users] Asterisk V/s FreeSwitch

I run about 150 cc on a xen vps with no problem mostly with no transcoding but I could have 15 channels transcoding and 15 channels are recorded I have a fs server on it too but not much more traffic so can't compare If asterisk would use the sofia sip stack it would probably be about the same

Re: [asterisk-users] Asterisk V/s FreeSwitch

From: "Jeff Brower" Sent: Wednesday, February 08, 2012 8:49 AM To: "Brynjolfur Thorvardsson" Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch Brynjolfur- > According to this article here: > > http://anders.com/cms/266 > > the difference mainly lie

Re: [asterisk-users] Asterisk V/s FreeSwitch

Brynjolfur- > According to this article here: > > http://anders.com/cms/266 > > the difference mainly lies in how FreeSwitchs handles open > channels in comparison with Asterisk. FS uses one thread > per channel while * keeps jumping between threads. At least > that's how I understand it. If the

Re: [asterisk-users] Asterisk V/s FreeSwitch

Zohair What kind of hardware spec are you running CPU, MEM, Drives? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Zohair Raza" Sent: Wednesday, February 08, 2012 3:08 AM To: "Asterisk Users Mailing List - Non-Commercial Discussio

Re: [asterisk-users] SIP hardware phones

From everything I've researched to date, my understanding is most locations have chosen to double their port density and continue to service the phone and computer on separate ports than to share a single line for both computer and phone. Reason primarily mentioned being troubleshooting concern

Re: [asterisk-users] Calling a group of phones and force the speaker

On Wednesday 08 February 2012, Danny Dias wrote: > Hi, > > I wonder, if there is a way to call from A phone to a group of phones (B, C > and D) and force these phones to activate automatically the speaker > > Is that possible? > > Many thanks in advance That depends on the phones supporting suc

[asterisk-users] Calling a group of phones and force the speaker

Hi, I wonder, if there is a way to call from A phone to a group of phones (B, C and D) and force these phones to activate automatically the speaker Is that possible? Many thanks in advance -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.org -- _

Re: [asterisk-users] Asterisk 1.8.9.1 Now Available

Not seeing it on packages.asterisk.org or digium. Will it be up soon? Thanks! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Development Team Sent: Tuesday, February 07, 2012 3:28 PM To: Aste

[asterisk-users] R: Asterisk V/s FreeSwitch

>From the second link Gilles suggested: "Aterisk vs FreeSWITCH" " [...] The mailing list (FreeSWITCH) is also a very nice place. In both places (IRC and mailing list) they are very friendly and supportive, unlike the Asterisk/Digium community. [...]" BAH... -Messaggio originale-

[asterisk-users] SIP trunk audio bad but is OK again after SIP re-registration

Hi, When *ANY* SIP client (softphone, hardphone, ATA) registers to an Asterisk server on my LAN and the extension dials out through a remote SIP provider, the audio is fine for "a while". It then degrades and starts to be "cracky"/jittery. The extension can call once and again and it will alway

Re: [asterisk-users] Binding to 0.0.0.0 a security risk?

In article <4f324279.70...@message-id.plonk.de>, Jakob Hirsch wrote: > Raj Mathur (राज माथुर), 2012-02-08 03:27: > > Packets not going out on the same interface as the one they were > > received on is a general IP issue, not just for connectionless > > Right, this was a inaccura

Re: [asterisk-users] SIP hardware phones

2012/2/8, Vieri : > I'm trying to understand why vendors keep making 100Mbps integrated 1-port > switches in their hardware SIP phones. Even the recently-announced D40 and > D50 Digium phones are limited to 100Mbps. Only the more expensive models > (like the D70) can run at 1000Mbps. > However, you

Re: [asterisk-users] Binding to 0.0.0.0 a security risk?

Raj Mathur (राज माथुर), 2012-02-08 03:27: > Packets not going out on the same interface as the one they were > received on is a general IP issue, not just for connectionless Right, this was a inaccuracy. It should say "Asterisk does not reply with the IP address with which packets were received"

[asterisk-users] Automatic Number Identification and anonymous calls

Hi, This morning I called an analog line from a cell phone. For the first call, I made a standard call and could both read my CID and ANI in my Asterisk console. For the second call, I choosed to hide my ID and, to my surprise, I read that my CID and ANI were both empty. I always thought that AN

[asterisk-users] SIP hardware phones

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with

Re: [asterisk-users] Asterisk V/s FreeSwitch

Virendra, You can test your box with sipp http://etel.wiki.oreilly.com/wiki/index.php/Using_SIPp_to_Stress_Test_Asterisk I have verified my Asterisk 1.8 box handling 500 concurrent calls and 15 calls per seconds with 20% cpu, without transcoding. Regards, Zohair Raza On Wed, Feb 8, 2012 at 1

Re: [asterisk-users] Asterisk V/s FreeSwitch

My Asterisk 1.4.19 happily excepts up to 80 concurrent calls (the most I've seen so far) which sends the CPU load up to ~20% on a fairly old server. In our busiest period, from 8 to 8:05 I see up to 200 incoming calls, somewhat less than one call/second. My superiors want to expand and increase