Re: [asterisk-users] Asterisk SIP TCP

2013-04-16 Thread Bharat Lalcheta
;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the

Re: [asterisk-users] Asterisk SIP TCP

2013-04-16 Thread Zohair Raza
On Tue, Apr 16, 2013 at 10:12 AM, Bharat Lalcheta bharatlalch...@gmail.comwrote: ;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the

Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists

2013-04-16 Thread s m
thanks guys, i solve my problem. as Asghar said, i remove 2 and forget to add it again therefore asterisk can not recognize extension 200 in extension.conf file. this is my extension that works properly: exten=_2.,1,Dial(SIP/to-231/1${EXTEN:2}) thanks every body for your attention. Sam On

[asterisk-users] Access postgresql directly from dialplan?

2013-04-16 Thread Sebastian Arcus
I would like to access a Postgresql database directly from my dialplan (to lookup names based on callerid numbers for incoming calls). Based on everywhere I looked - it seems the only way to do this is with func_odbc. Considering that Asterisk seems to be able to access Postgresql databases

Re: [asterisk-users] Access postgresql directly from dialplan?

2013-04-16 Thread Gertjan Baarda
On 16 apr. 2013, at 15:08, Sebastian Arcus s...@open-t.co.uk wrote: I would like to access a Postgresql database directly from my dialplan (to lookup names based on callerid numbers for incoming calls). Based on everywhere I looked - it seems the only way to do this is with func_odbc.

[asterisk-users] erro compiling dahdi

2013-04-16 Thread Jonas Kellens
Hello, when compiling dahdi (CentOS 2.6.18-348.3.1.el5) I get the following error : In file included from /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xpd.h:26, from /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.c:29:

Re: [asterisk-users] erro compiling dahdi

2013-04-16 Thread Shaun Ruffell
On Tue, Apr 16, 2013 at 09:03:21PM +0200, Jonas Kellens wrote: Hello, when compiling dahdi (CentOS 2.6.18-348.3.1.el5) I get the following error : In file included from /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xpd.h:26, from

Re: [asterisk-users] erro compiling dahdi

2013-04-16 Thread Russ Meyerriecks
On Tue, Apr 16, 2013 at 09:03:21PM +0200, Jonas Kellens wrote: when compiling dahdi (CentOS 2.6.18-348.3.1.el5) I get the following error : /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xdefs.h:152: error: conflicting types for 'bool' This is fixed in dahdi 2.6.2 --

[asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

2013-04-16 Thread Markus
Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using BroadWorks. The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port 36252 but

Re: [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

2013-04-16 Thread Matthew J. Roth
Markus, I think I know what's wrong here but I did a fair amount of research while digging into your problem. I may have misinterpreted something along the way so you should also consider other responses, especially if they come from someone who claims greater expertise. I did this to help you

Re: [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

2013-04-16 Thread Joshua Colp
Markus wrote: Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using BroadWorks. The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port

Re: [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

2013-04-16 Thread Matthew J. Roth
Joshua Colp wrote: If you set nat=no for that specific peer it should work as you need. 'rport' is forced on these days which works for most situations, except with some platforms and Cisco phones. _ Joshua, That sounds much easier than what I came up with, so I'd recommend to Markus that

Re: [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

2013-04-16 Thread Markus
Joshua, Matthew, Am 17.04.2013 02:01, schrieb Matthew J. Roth: Joshua Colp wrote: If you set nat=no for that specific peer it should work as you need. 'rport' is forced on these days which works for most situations, except with some platforms and Cisco phones. _ Joshua, That sounds much

[asterisk-users] Transfer only, no outbound calling

2013-04-16 Thread Todd Routhier
OK, it's been a while since I drank from the pool of wisdom hear on the list. After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink. So, on to my question: I have some agents/operators setup in sip.conf which

Re: [asterisk-users] Transfer only, no outbound calling

2013-04-16 Thread Nathan Anderson
On Tuesday, April 16, 2013 6:25 PM, Todd Routhier wrote: New Problem, now operators can pick up the previous inbound only line and dial out to anything that matches the patterns I have defined in the context for their extension in sip.conf. What I really need to make work here is