Re: [asterisk-users] Realtime vs Static Files

2013/1/24 Dan Journo > >> Its probably an issue with the version of Asterisk we are using > because I haven't had this problem in the past. > > > I am running the latest 1.8 version. Which version are you running? > > ** ** > > ** ** > > 1.8.15.0. I'll upgrade it to 1.8.20.1 when I can an

Re: [asterisk-users] Asterisk 11 with t38modem 2.0: "488 Not acceptable here"

Hello Larry, thank you for your input, it is highly appreciated. On 24.01.2013 00:41, Larry Moore wrote: > Turn off faxdetect in the peer configuration for Asterisk. Unfortunately faxdetect is allready off, so no luck here. > Failing that, try the Fax Gateway feature in Asterisk 11 to Hylafax >

Re: [asterisk-users] Uninitialized variable in main/pbx.c?

> > + dst_exten[0] = '\0'; > > Is this 'construct' prefered over > > dst_exten[0] = 0; > or > *dst_exten = 0; > > and why? I'm somewhat of a C pedant here. dst_exten is declared as an array, not a pointer. So if I want to clear the first byte of

Re: [asterisk-users] Uninitialized variable in main/pbx.c?

On Wed, 23 Jan 2013, Richard Kenner wrote: + dst_exten[0] = '\0'; Is this 'construct' prefered over dst_exten[0] = 0; or *dst_exten = 0; and why? -- Thanks in advance, -

Re: [asterisk-users] Uninitialized variable in main/pbx.c?

On 01/23/2013 08:13 PM, Richard Kenner wrote: > I think the below fixes what I reported earlier. Does that seem right? > > *** pbx.c.old 2013-01-23 21:08:51.0 -0500 > --- pbx.c 2013-01-23 21:09:31.0 -0500 > *** static enum ast_pbx_result __ast_pbx_run > *** 516

[asterisk-users] Uninitialized variable in main/pbx.c?

I think the below fixes what I reported earlier. Does that seem right? *** pbx.c.old 2013-01-23 21:08:51.0 -0500 --- pbx.c 2013-01-23 21:09:31.0 -0500 *** static enum ast_pbx_result __ast_pbx_run *** 5160,5163 --- 5160,5165 int timeout

Re: [asterisk-users] Asterisk 11 with t38modem 2.0: "488 Not acceptable here"

My 2 cents worth. Turn off faxdetect in the peer configuration for Asterisk. Failing that, try the Fax Gateway feature in Asterisk 11 to Hylafax listening on an IAX2 channel. Larry. On 24/01/2013 6:32 AM, Carsten Maass wrote: Hello all, we do have a problem here with Asterisk 11 talking T.

Re: [asterisk-users] Asterisk 11 with t38modem 2.0: "488 Not acceptable here"

On 01/23/2013 04:32 PM, Carsten Maass wrote: > Hello all, > > we do have a problem here with Asterisk 11 talking T.38 to a t38modem > 2.0. The callflow is: > > ISDN PRI --> Berofix (10.1.1.150) --> Asterisk (10.1.1.148) --> t38modem > (10.1.1.148) --> Hylafax [1] > > Although the call gets conne

Re: [asterisk-users] Realtime vs Static Files

>> Its probably an issue with the version of Asterisk we are using because I >> haven't had this problem in the past. > I am running the latest 1.8 version. Which version are you running? 1.8.15.0. I'll upgrade it to 1.8.20.1 when I can and see if it makes a difference. -- _

Re: [asterisk-users] DAHDI: How to supress notification of changing CallerID on transfer?

- Original Message - > Hello out there, > > I'm running an Asterisk 1.8.15-cert1 with DAHDI. > Today I noticed that Asterisk is signalling to the calling party the > current internal CallerID whenever I put a call to another internal > phone. > > Example: > > Customer calls 020212345-555

Re: [asterisk-users] Realtime vs Static Files

On 13-01-23 04:41 AM, Dan Journo wrote: Hi, We're trying to decide whether to switch back to a static file for sip.conf. Currently we use mysql realtime but can't see any real benefit. Why would someone choose realtime sip over static files? Thanks I'm interested in the feedback too. For ye

[asterisk-users] Asterisk 11 with t38modem 2.0: "488 Not acceptable here"

Hello all, we do have a problem here with Asterisk 11 talking T.38 to a t38modem 2.0. The callflow is: ISDN PRI --> Berofix (10.1.1.150) --> Asterisk (10.1.1.148) --> t38modem (10.1.1.148) --> Hylafax [1] Although the call gets connected, both parties are unable to negotiate the audio codecs: [

[asterisk-users] DAHDI: How to supress notification of changing CallerID on transfer?

Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 -> IVR answers and puts caller to the chosen qu

Re: [asterisk-users] Is there a need to secure RTP ports?

On 23/01/13 17:33, Carlos Alvarez wrote: On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus mailto:s...@open-t.co.uk>> wrote: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the

Re: [asterisk-users] Is there a need to secure RTP ports?

Thanks Danny. I've already reduced the number of RTP ports used in Asterisk configs and the firewall - as 1 seemed like a crazy number for my needs! On 23/01/13 17:27, Danny Nicholas wrote: As I am going to mis-explain this, an Asterisk SIP call originates on port 5060 (incoming or outgoin

Re: [asterisk-users] Realtime vs Static Files

2013/1/23 Dan Journo > > Maybe you are lacking some of the configuration. These is the relevant > part. > > ** ** > > > rtcachefriends=yes > > > rtsavesysname=yes > > > rtupdate=yes > > > rtautoclear=yes > > ** ** > > We have > > rtcachefriends=yes >

Re: [asterisk-users] Realtime vs Static Files

> Maybe you are lacking some of the configuration. These is the relevant part. > rtcachefriends=yes > rtsavesysname=yes > rtupdate=yes > rtautoclear=yes We have rtcachefriends=yes rtsavesysname=yes and these we don't have but they are set to YES by default > rtupdate=yes > rtautoclear=yes Its p

Re: [asterisk-users] DPMA and Sending fake auth rejection for device

Shouldn't make a difference. I always set phones as "friend". On Wed, Jan 23, 2013 at 10:26 AM, Frank wrote: > Hi George, > > My sip.conf as a "friend" and not a "peer". Does this make any change ? > F. > > > > > On 1/23/13 12:04 PM, George Joseph wrote: > >> You might want to start with simple

Re: [asterisk-users] Is there a need to secure RTP ports?

On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus wrote: > I have an Asterisk server with one SIP trunk to a SIP provider. As my > server registers with the SIP provider, I don't have any SIP ports open at > my end to the Internet. However, I have the RTP ports open (as SIP has some > trouble wit

Re: [asterisk-users] Is there a need to secure RTP ports?

As I am going to mis-explain this, an Asterisk SIP call originates on port 5060 (incoming or outgoing) then uses two RTP ports for audio in and audio out. Police and Hackers can tap into the RTP ports to monitor your conversations (I don't really know if the capabilities stop there) but you can li

Re: [asterisk-users] DPMA and Sending fake auth rejection for device

Hi George, My sip.conf as a "friend" and not a "peer". Does this make any change ? F. On 1/23/13 12:04 PM, George Joseph wrote: You might want to start with simple mac authentication. Use config_auth=mac in the general section, then mac= in the Home_phone section. This way you can at least

[asterisk-users] Is there a need to secure RTP ports?

I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has some trouble with my NAT). My question is - what are the vulnerabilities in thi

Re: [asterisk-users] Problems with 'i' extension

>> What's going on? Shouldn't this go to that extension? Check your extensions.conf, in the general section, for: autofallthrough=no priorityjumping=no Do

[asterisk-users] Problems with 'i' extension

I'm running Asterisk 10.7.1. In the log, I see: -- Goto (Conferences,70323,1) -- Auto fallthrough, But there is an 'i' extension: dialplan show i@Conferences [ Context 'Conferences' created by 'pbx_config' ] '_[ti]' =>1. GotoIf($[${SET(REC=$[${REC}--1])}>3]?999) [pbx_config

Re: [asterisk-users] DPMA and Sending fake auth rejection for device

You might want to start with simple mac authentication. Use config_auth=mac in the general section, then mac= in the Home_phone section. This way you can at least eliminate DPMA auth from the equation. If you still get the message, it usually means that there's no matching peer in sip.conf. I do

[asterisk-users] Digium Phones. Can BLF keys be made to function during conversation?

Can the Digium Phones BLF keys be made to function while in conversation without first havint to Park or Hold the current call? Currently they are essentially dead while I am in conversation other than the light state changing. For example, I may want a key that puts the current call on hold the

Re: [asterisk-users] two steps when calling from web!

On Wed, Jan 23, 2013 at 10:14 AM, Danny Nicholas wrote: > Originate is the answer here. Let’s say your X-lite is SIP/100 and you’re > dialing 555-1212. From the x-lite you dial 555-1212 and Asterisk does a > dial command to execute the call. From the web, we “originate” the call > from SIP/100

Re: [asterisk-users] two steps when calling from web!

Originate is the answer here. Let’s say your X-lite is SIP/100 and you’re dialing 555-1212. From the x-lite you dial 555-1212 and Asterisk does a dial command to execute the call. From the web, we “originate” the call from SIP/100 to 555-1212. Asterisk makes sure SIP/100 is available then di

Re: [asterisk-users] Execute a script outside Asterisk

Hello, the & behind the command to execute in the background is a great idea ! Jonas. On 01/23/2013 04:29 PM, Danny Nicholas wrote: Here is the way I got it to do what I think you want. '1250' => 1. answer() [pbx_config] 2. setMusiconhold(jazz)

Re: [asterisk-users] two steps when calling from web!

On Wed, Jan 23, 2013 at 1:09 AM, Muhammad wrote: > -1 in normal way, when I type the number in softphone, it call the number > and show me just "End" bottom. > 2- when I calling the number through the web, it show me "Answer" bottom > and I have to click answer to calling then number. it is 2 step

Re: [asterisk-users] Execute a script outside Asterisk

Here is the way I got it to do what I think you want. '1250' => 1. answer() [pbx_config] 2. setMusiconhold(jazz) [pbx_config] 3. AGI(wait10.sh) [pbx_config] 4. playback(vm-goodbye) [pbx_config] 5. setMusi

Re: [asterisk-users] Execute a script outside Asterisk

Hello, will this : Exten => 2,n,playback(vm-goodbye) be executed even when Exten => 2,1,system(Jonas.php) is still executing ?? The exact snippet would be : Exten => s,1,answer() Exten => s,n,system(Jonas.php) ; script that may take a minute Exten => s,n,do something Exten => s,n,Dial(SIP

Re: [asterisk-users] Execute a script outside Asterisk

Let's assume you're using this snippet [default] Exten => s,1,answer() Exten => s,n,playback(tt-monkeys) Exten => s,n,waitexten(6) Exten => s,n,hangup() Exten => 1,1,AGI(Jonas.php) Exten => 1,n,playback(vm-goodbye) Exten => 1,n,hangup() Exten => 2,1,system(Jonas.php) Exten => 2,n,playback

Re: [asterisk-users] Execute a script outside Asterisk

Hello, thank you for your answer. The most important here is that Asterisk continues with the rest of the dialplan, in case the database-connection fails or hangs or ... I don't think the System()-command makes this true. Jonas. On 01/23/2013 03:27 PM, Danny Nicholas wrote: I would vo

Re: [asterisk-users] Execute a script outside Asterisk

I would vote for system() on two accounts. #1 AGI requires more overhead and protocol #2 you are not expecting a result to return to the dialplan. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January

Re: [asterisk-users] Realtime vs Static Files

2013/1/23 Dan Journo > > We have never experienced that and use realtime with multiple asterisk > servers. > > We've only recently started seeing the problem. > > To simplify the issue, assuming we have two servers, Asterisk1 and > Asterisk2... > > Asterisk1 is a primary server and Asterisk2 is a

Re: [asterisk-users] Realtime vs Static Files

> We have never experienced that and use realtime with multiple asterisk > servers. We've only recently started seeing the problem. To simplify the issue, assuming we have two servers, Asterisk1 and Asterisk2... Asterisk1 is a primary server and Asterisk2 is a backup and used as a failover. As

Re: [asterisk-users] Realtime vs Static Files

On Wed, 2013-01-23 at 05:53 -0500, Dan Journo wrote: > For example, I've noticed that if two asterisk servers share the > database, often they both think that the peer is registered. Also the > issues with MWI and Qualify= We have never experienced that and use realtime with multiple asterisk serv

Re: [asterisk-users] Realtime vs Static Files

>All depends by the number of sip peers and the number of addition/deletion you >make. If you have static files, you have to "sip reload" every time you >add/remove a peer. With realtime is all "realtime". I have switched to >realtime peers some times ago with great benefit. However, there are

Re: [asterisk-users] [SOLVED] Blind transfer behavior - Asterisk 1.8 and 10

Le 22/01/2013 18:22, Leandro Dardini a écrit : Can you please post a dialplan excerpt about using these variables. I just tried using them, but they are all empty. Maybe I am making the same mistake of you. Sure. Don't forget to add option T or t in the dial string. [from_to-OFFICE] ; [...] e

Re: [asterisk-users] Realtime vs Static Files

2013/1/23 Dan Journo > Hi, > > ** ** > > We're trying to decide whether to switch back to a static file for > sip.conf. Currently we use mysql realtime but can't see any real benefit.* > *** > > ** ** > > Why would someone choose realtime sip over static files? > > ** ** > > Thanks >

[asterisk-users] Execute a script outside Asterisk

Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB).

[asterisk-users] Realtime vs Static Files

Hi, We're trying to decide whether to switch back to a static file for sip.conf. Currently we use mysql realtime but can't see any real benefit. Why would someone choose realtime sip over static files? Thanks Dan Journo Kesher Communications (UK) Business Phone Systems