[asterisk-users] ODBC connection not connected at 1st call.

2011-09-30 Thread virendra bhati
Hi list, This is comman issue when we use ODBC connection to get Database values. At the 1st attempt connection is not connected but soon automatically connected by aterisk. What is the real problem ? it this asterisk ODBC issue or else ? -- Executing [_36899XX@incoming:1]

[asterisk-users] invite authentication error !?

2011-09-30 Thread cnasterisk
hi, Dear all. I setted a sip account on a sip trunk. when a client call via this sip trunk, asterisk call failed on this trunk. I have captured the sip messages on the host where asterisk located, and found that: 1. asterisk send a INVITE message to remote sip proxy without

Re: [asterisk-users] invite authentication error !?

2011-09-30 Thread Alex Balashov
This is just a speculative shot in the dark, but remember that the domain of the From URI is important, and that the authentication realm (domain) is part of the authentication credentials. So, what you have in your 'fromdomain' and 'host' settings on the peer does matter. -- This message was

Re: [asterisk-users] invite authentication error !?

2011-09-30 Thread Sam Govind
What Sip declaration are you using for the remote sip proxy in sip.conf? On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov abalas...@evaristesys.comwrote: This is just a speculative shot in the dark, but remember that the domain of the From URI is important, and that the authentication realm

Re: [asterisk-users] invite authentication error !?

2011-09-30 Thread cnasterisk
hi, alex thanks for your kindly reply. the remote proxy do not use domain as realm, the following is part of the message 210.83.80.xxx is the ip of asterisk 202.104.188.xx is the ip of remote proxy sip server ( not asterisk) --

Re: [asterisk-users] invite authentication error !?

2011-09-30 Thread cnasterisk
hi, Sam thanks for your kindly reply. The remote proxy is not asterisk 2011-09-30 cnasterisk 发件人: Sam Govind 发送时间: 2011-09-30 15:36:41 收件人: Asterisk Users Mailing List - Non-Commercial Discussion 抄送: 主题: Re: [asterisk-users] invite authentication error !? What Sip declaration are

Re: [asterisk-users] invite authentication error !?

2011-09-30 Thread Sam Govind
Whatever that remote party is, you are most definitely using a username/secret declaration for that. So the sip attributes set for that proxy define the behaviour for this. On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk cnaster...@163.com wrote: ** hi, Sam thanks for your kindly reply. The

Re: [asterisk-users] [asterik-users] Installing PRI card

2011-09-30 Thread A J Stiles
On Friday 30 September 2011, NaJIm wrote: Am I getting these error messages due to wrong configurations in my zapata.conf. ?? I have got the following configurations in my zapata-channels.conf. ; Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) B8ZS/ESF RED group=0,11 context=from-pstn

Re: [asterisk-users] invite authentication error !?

2011-09-30 Thread cnasterisk
asterisk can register successfully on the remote party. so i think username password must be ok 2011-09-30 cnasterisk 发件人: Sam Govind 发送时间: 2011-09-30 16:05:21 收件人: Asterisk Users Mailing List - Non-Commercial Discussion 抄送: 主题: Re: [asterisk-users] invite authentication error !?

Re: [asterisk-users] invite authentication error !?

2011-09-30 Thread Sam Govind
Please post some configurations. YOU CAN REMOVE username / secret from the sip configs but posting remaining configs will only help to your issue resolution, Alex first words were This is just a speculative shot in the dark . Fine if you were just Trolling here ! 2011/9/30 cnasterisk

[asterisk-users] Core show translation 4000ms

2011-09-30 Thread Administrator TOOTAI
Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the

[asterisk-users] SIPit 29 in Monaco - interoperability by hard work

2011-09-30 Thread Olle E. Johansson
Friends, SIPit is an event organized by the SIP Forum and partners. It has been running for 15 years twice a year, making sure that SIP clients and servers interoperate. By testing, we also find issues with the myriad of RFCs in this area and correct them. Testing interoperability is

Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-30 Thread salaheddine elharit
Hi Thanks everyone for your help and support all works perfectly Best Regards 2011/9/29 salaheddine elharit salah.elharit...@gmail.com ok thanks for your response i will try that and i will update you as soon as i have any result best regards 2011/9/29 A J Stiles

Re: [asterisk-users] Core show translation 4000ms

2011-09-30 Thread Kevin P. Fleming
On 09/30/2011 03:56 AM, Administrator TOOTAI wrote: Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme

Re: [asterisk-users] Core show translation 4000ms

2011-09-30 Thread Administrator TOOTAI
Le 30/09/2011 14:05, Kevin P. Fleming a écrit : On 09/30/2011 03:56 AM, Administrator TOOTAI wrote: Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no

Re: [asterisk-users] record calls of specific agnets

2011-09-30 Thread Lyle McKarns
On 09/29/2011 12:37 PM, A J Stiles wrote: On Thursday 29 September 2011, Lyle McKarns wrote: Hello Asterisk List! I have been asked to record calls from specific agents, and I am having difficulty finding if this is possible, and if so, how exactly to do it. Have a recorded context and an

Re: [asterisk-users] record calls of specific agnets

2011-09-30 Thread Gohar Ahmed
Hi, I think use of any Macro in queue can serve you well. Macro will be called whenever the call is established to the agent. In that Macro check your desired Agent and if condition matched trigger MixMonitor else do nothing. Regards, Gohar A. From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Core show translation 4000ms

2011-09-30 Thread Kevin P. Fleming
On 09/30/2011 07:49 AM, Administrator TOOTAI wrote: Le 30/09/2011 14:05, Kevin P. Fleming a écrit : On 09/30/2011 03:56 AM, Administrator TOOTAI wrote: Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS

[asterisk-users] asterisk hardware

2011-09-30 Thread Adam Moffett
Is there any reason not to run Asterisk on an Intel Atom board? -- _ -- 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] Core show translation 4000ms

2011-09-30 Thread Tony Mountifield
In article 4e85d19f.4090...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: On 09/30/2011 07:49 AM, Administrator TOOTAI wrote: Le 30/09/2011 14:05, Kevin P. Fleming a écrit : On 09/30/2011 03:56 AM, Administrator TOOTAI wrote: Hi list, we have 2 asterisk boxes in VM (kvm) on 2

Re: [asterisk-users] Core show translation 4000ms

2011-09-30 Thread Eric Wieling
I always use the recalc option to show translations, it seems to provide much more accurate numbers. Example: core show translation recalc 20 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield

Re: [asterisk-users] Core show translation 4000ms

2011-09-30 Thread Jason Parker
On 09/30/2011 09:53 AM, Tony Mountifield wrote: In article 4e85d19f.4090...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: This is why the output was changed to microseconds from milliseconds; in the older version, the lowest number that should be shown was 1 millisecond, even if

Re: [asterisk-users] Core show translation 4000ms

2011-09-30 Thread Administrator TOOTAI
Le 30/09/2011 16:59, Eric Wieling a écrit : I always use the recalc option to show translations, it seems to provide much more accurate numbers. Example: core show translation recalc 20 Lenny kernel, new values, still 1000 microseconds between both directions Translation times

Re: [asterisk-users] Core show translation 4000ms

2011-09-30 Thread Administrator TOOTAI
Le 30/09/2011 17:02, Jason Parker a écrit : On 09/30/2011 09:53 AM, Tony Mountifield wrote: In article4e85d19f.4090...@digium.com, Kevin P. Flemingkpflem...@digium.com wrote: This is why the output was changed to microseconds from milliseconds; in the older version, the lowest number that

[asterisk-users] Strange Asterisk HA Behaviour

2011-09-30 Thread Nick Khamis
Hello Everyone, We have setup Asterisk HA, basically what we have is: Virtual IP (for asterisk 1 and 2)192.168.2.6 asterisk1 192.168.2.110 asterisk2 192.168.2.111 mysql192.168.2.100

Re: [asterisk-users] asterisk hardware

2011-09-30 Thread Ira
At 07:50 AM 9/30/2011, you wrote: Is there any reason not to run Asterisk on an Intel Atom board? Mine's been running that way for 3 years or so. 2 users 6 extensions, SIP + 3 POTs lines with a TDM04. Ira -- _ --

[asterisk-users] USA Did required

2011-09-30 Thread amit mehta
Hello members, I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls. Let me know your best per month prices/cost for the above. Regards, Amit Mehta --

Re: [asterisk-users] [asterik-users] Installing PRI card

2011-09-30 Thread NaJIm
Hi, As Eric mentioned I made my zapata.conf and zaptel.conf to match each other. My *zapata.conf *was *group=1* *switchtype=euroisdn* *signalling=pri_cpe* *callerid=asreceived* *usecallerid=yes* *cidsignalling=dtmf* *cidstart=ring* *context=TEST_EXTERNAL* *channel=1-15* *channel=17-24* This

Re: [asterisk-users] USA Did required

2011-09-30 Thread Tarek Sawah
Google is your best friend when looking for this type of assistance my friend. try callcentric vonage packet8 for reliable retail DIDs. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 1 Oct 2011 00:51:59

Re: [asterisk-users] USA Did required

2011-09-30 Thread Tamer Higazi
Go to ATT and ask for a BRI ISDN Line. Tell them that you want a system access with a number block. that would be typically 54448[0-9] where your extensions are from 0-9 I don't know what protocol the americans are using, as I know the americans made for sure their own thing which is known as