[asterisk-users] authentication user with custom authentication key

2014-07-31 Thread Sameer Rathod
Hi, I want to authenticate user with a random authentication key before registration in asterisk for a click2dial feature in my website. The goal is to not to display the password to the client. The client will be provided with a authentication key and when the request comes to the server form

[asterisk-users] AGI Record File / what does randomerror mean? res_agi.c / line 2377

2014-07-31 Thread Thorsten Göllner
Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307

[asterisk-users] Subscription-State always active ?

2014-07-31 Thread Jonas Kellens
Hello, I notice that Asterisk always sends Subscription-State: active, even when the SIP-peer is offline (IP-phone cut from power) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49// //[Jul 31 11:56:58] Really

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Joshua Colp
Jonas Kellens wrote: Hello, Kia ora, I notice that Asterisk always sends Subscription-State: active, even when the SIP-peer is offline (IP-phone cut from power) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now UNREACHABLE! Last qualify:

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Jonas Kellens
On 31-07-14 12:13, Joshua Colp wrote: Jonas Kellens wrote: I notice that Asterisk always sends Subscription-State: active, even when the SIP-peer is offline (IP-phone cut from power) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Joshua Colp
Jonas Kellens wrote: Hello, I read on Yealink support that Yealink IP-phones expect Subscription-State:terminated for there Presence/BLF-functionality. So how can I get Subscription-State:terminated on Asterisk ? That would be a bit strange as the subscription would then be terminated, no

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Jonas Kellens
On 31-07-14 14:28, Joshua Colp wrote: Jonas Kellens wrote: Hello, I read on Yealink support that Yealink IP-phones expect Subscription-State:terminated for there Presence/BLF-functionality. So how can I get Subscription-State:terminated on Asterisk ? That would be a bit strange as the

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Joshua Colp
Jonas Kellens wrote: On 31-07-14 14:28, Joshua Colp wrote: Jonas Kellens wrote: Hello, I read on Yealink support that Yealink IP-phones expect Subscription-State:terminated for there Presence/BLF-functionality. So how can I get Subscription-State:terminated on Asterisk ? That would be a

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Jonas Kellens
On 31-07-14 15:06, Joshua Colp wrote: Jonas Kellens wrote: On 31-07-14 14:28, Joshua Colp wrote: Jonas Kellens wrote: Hello, I read on Yealink support that Yealink IP-phones expect Subscription-State:terminated for there Presence/BLF-functionality. So how can I get

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Joshua Colp
Jonas Kellens wrote: Hello, I was reading this post : http://forum.yealink.com/forum/showthread.php?tid=894 http://forum.yealink.com/forum/showthread.php?tid=894pid=4794#pid4794 Has the explanation. Since they are using dialog-info+xml there's nothing different between not in use and

Re: [asterisk-users] Internal timing under load is critical ?

2014-07-31 Thread Chris Bagnall
On 30/7/14 10:08 am, babak wrote: I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. As a matter of curiosity, what do people use these voice broadcasting solutions for? I'm genuinely struggling to think of (legal) reasons why you'd

Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

2014-07-31 Thread James Thomas
Is the quality the same incoming from mobile as outgoing to mobile? On Wed, Jul 30, 2014 at 4:51 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio

Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

2014-07-31 Thread A J Stiles
On Thursday 31 Jul 2014, James Thomas wrote: Is the quality the same incoming from mobile as outgoing to mobile? It's a one-way trunk (outgoing only). Anyway, I've now fixed it, with help from the trunk provider. Details to follow in a separate message. -- AJS Note: Originating address

Re: [asterisk-users] Internal timing under load is critical ?

2014-07-31 Thread Daniel Taylor
On 07/31/2014 09:51 AM, Chris Bagnall wrote: On 30/7/14 10:08 am, babak wrote: I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. As a matter of curiosity, what do people use these voice broadcasting solutions for? I'm genuinely

Re: [asterisk-users] *SOLVED* SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

2014-07-31 Thread A J Stiles
I have now fixed this issue, and am posting this for the benefit of anyone else who may be suffering with a similar problem. It was, as I suspected all along, a subtle misconfiguration at this end. The fix was to give the SIP trunk its own configuration stanza in sip.conf as follows;

Re: [asterisk-users] Directory app not working with realtime

2014-07-31 Thread Tech Support
This is what I have in my configs. My debug shows that it is using realtime for voicemail when using the ‘directory’ application, but it's not working and can't find any extensions. However, checking voicemail does work with this configuration. My voicemail context is called

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Jonas Kellens
On 31-07-14 16:14, Joshua Colp wrote: Jonas Kellens wrote: Hello, I was reading this post : http://forum.yealink.com/forum/showthread.php?tid=894 http://forum.yealink.com/forum/showthread.php?tid=894pid=4794#pid4794 Has the explanation. Since they are using dialog-info+xml there's

Re: [asterisk-users] Subscription-State always active ?

2014-07-31 Thread Joshua Colp
Jonas Kellens wrote: Hello, the explanation is that is does not work with Asterisk ? I don't understand. Asterisk sends dialog-info+xml, right ?! You can see it in my first post : It does not work because the specification for dialog-info+xml has no difference between not on the phone and

Re: [asterisk-users] Internal timing under load is critical ?

2014-07-31 Thread babak
As a matter of curiosity, what do people use these voice broadcasting solutions for? I'm genuinely struggling to think of (legal) reasons why you'd want to broadcast 1000+ simultaneous calls. Perhaps I'm just not being imaginative enough... :-) Kind regards, Chris For a big PSTN

Re: [asterisk-users] Asterisk 12.4.0 not able to install pjsip

2014-07-31 Thread Sameer Rathod
Hi any progress on this issue ?? anyone??? On Fri, Jul 25, 2014 at 1:22 PM, Sameer Rathod sam...@hostnsoft.com wrote: after exporting ie export PKG_CONFIG_PATH=/usr/lib/pkgconfig This command pkg-config --print-provides libpjproject shows *libpjproject = 2.2.0* and for pkg-config

Re: [asterisk-users] Asterisk 12.4.0 not able to install pjsip

2014-07-31 Thread Matthew Jordan
On Thu, Jul 31, 2014 at 1:53 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi any progress on this issue ?? anyone??? Can you attach your config.log output? Since pkg-config does find your pjproject installation, and the Asterisk configure script uses pkg-config to find pjproject, this

[asterisk-users] Asterisk 1.8.8.0 is raising...

2014-07-31 Thread Leonardo Sandoval Lozano
Hello, I have Asterisk 1.8.8.0 install in a HP DL180G6, Asterisk was usually working as normal and suddenly Asterisk start raising up the CPU very high untill the server crashes. Any suggestion ??? Best regards -- Leonardo Sandoval Lozano Estudiante I am the master of my fate: I am the