[asterisk-users] Asterisk Tag Berlin live notes page
Hi, If you're interested in what's happening in Berlin at Asterisk Tag for the next few days, you can look here: http://www.scribblelive.com/Thread.aspx?Id=815 I will try to keep an event trail going and notes if interviews are posted. Anyone can post questions and I don't think there's even a need to create an account or log in; You can also follow asterisktag on some of these http://twitter.com/asterisktag http://twitter.com/randulo http://twitter.com/wintermeyer YATS! IF YOU ARE IN BERLIN please watch this space and contribute: http://www.scribblelive.com/Thread.aspx?Id=815 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls not being answered by asterisk
Hey thanks for the help :) though i already did that, and the sip debugging info shows me tht its ringing on the respective sip extension (1002) but nothing is happening.. so i guess its true it IS a diala plan issue tht i am yet to figure it out ... Date: Sat, 24 May 2008 14:20:45 +0100 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming calls not being answered by asterisk The first thing to do is type sip debug on the console and place the call from the Sipura. If you get a bunch of SIP messages flashing down your console you know the call is reaching Asterisk and it's most likely going to be an issue authenticating the call or a problem in your dial plan. If no SIP messages flash up then the call is not reaching your Asterisk server. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cmd Dial (a group) and 1) who pick up the call 2) How to use the G option
Hello, 1) The goal is to store the id of the operator who take the call in a mysql database andI don't know the (best) way to know which device take the call when we do a Dial cmd to a group of phones Some $var as DIALEDPEERNUMBER ? some inheritance ? using the G extension ? (and how to?) Thanks a lot for any reference/links or example 2) In the same idea, i don't mean exactely what is it possible to do with the G option of Dial cmd Ok the wiki is clear...callee/caller go to the same extension with priority x and x+1.. Sure, i'm not inventive.. what can i do with that ? are the $var of each caller/callee channel accessibles from the other leg (example: transfer the $id of the callee channel to a caller channel $var) ? And when a leg hangup, the other could continue on dialplan (as g option for the caller)? More: could the two legs do separate action in the dialplan during the call ? If someone have some example of use (or links etc) Have a nice week end ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trying directrtpsetup
Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool or can u just sniff? regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logical AND
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten = s,n,NoOp(${PSTN_NUM})
exten = s,n,ExecIf( $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten = s,n,NoOp(${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d0f518,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d0f518, 0
1|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d0f518,
0010123456789) in new stack
It should evaluate PSTN_NUM and add a 001 only if: PSTN_NUM is 10 digits
long and doesn't start with a 0.
However, from the debug it's being changed, even though the first test
operator logically is 0. If seems as though the isnt being applied.
Any ideas?
Thanks
Adrian
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Re: [asterisk-users] trying directrtpsetup
ronald ramos wrote: Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool or can u just sniff? Sniff. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logical AND (resent due to bounces)
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten = s,n,NoOp(${PSTN_NUM})
exten = s,n,ExecIf( $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten = s,n,NoOp(${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d0f518,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d0f518, 0
1|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d0f518,
0010123456789) in new stack
It should evaluate PSTN_NUM and add a 001 only if: PSTN_NUM is 10 digits
long and doesn't start with a 0.
However, from the debug it's being changed, even though the first test
operator logically is 0. If seems as though the isnt being applied.
Any ideas?
Thanks
Adrian
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Re: [asterisk-users] Transfer
Thanks Sherwood, But how do I send back a 302, once I'm already in the dialplan (hasn't asterisk already sent back a 200 OK by this point??) Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 23 May 2008 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Adrian Marsh wrote: Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number into the SS7 headers so that the originating number, dialed number and transfer number all stayed to specs, and everyone was happy. In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to have at least the control packets go via my SIP server), and use a Dial out to the far end. So - is there a way of handing the call back to the network in asterisk ? My detailed problem is this: When a call comes in, I want to send it onto users mobiles, if I hairpin the call that's OK, except the CLI needs to be that of the originator (from the USERS point of view) so they can decide if they want to accept the call. Here in the UK, this is where the issues begin... the carriers here don't like it if your sending CLI for other countries, that don't match what they think they should receive from that connecting carrier. Eg, if a call coming to them is 13 digits, but they only expect 11 from that carrier, then they cut the digits. This turns a US originated call into a Southampton UK originated call! So I was hoping that handing the call back to the network in the traditional sense would make it their problem and not mine... lol Thanks, Adrian -- -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.google.com/search?hl=enq=asterisk+302+redirect+sipbtnG=Sear ch I believe you're looking for a 302 Redirect? Sorry if you're not, but that sounds like what you want ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logical AND
Adrian Marsh schrieb:
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten = s,n,NoOp(${PSTN_NUM})
exten = s,n,ExecIf( $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten = s,n,NoOp(${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d0f518,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d0f518, 0
1|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d0f518,
0010123456789) in new stack
It should evaluate PSTN_NUM and add a 001 only if: PSTN_NUM is 10 digits
long and doesn't start with a 0.
However, from the debug it's being changed, even though the first test
operator logically is 0. If seems as though the isnt being applied.
Any ideas?
Thanks
Adrian
hello,
you should try this:
exten = s,n,ExecIf($[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]]|Set|PSTN_NUM=001${PSTN_NUM})
cause the AND Operator is another thing to work, so the result at your
way look like this ExecIf(11 | ...) and with my way it looks like this
ExecIf($[11]|...) which is the right syntax for it.
best regards
steve smith
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Re: [asterisk-users] Transfer
Thanks Sherwood, But how do I send back a 302, once I'm already in the dialplan (hasn't asterisk already sent back a 200 OK by this point??) Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 23 May 2008 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Adrian Marsh wrote: Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number into the SS7 headers so that the originating number, dialed number and transfer number all stayed to specs, and everyone was happy. In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to have at least the control packets go via my SIP server), and use a Dial out to the far end. So - is there a way of handing the call back to the network in asterisk ? My detailed problem is this: When a call comes in, I want to send it onto users mobiles, if I hairpin the call that's OK, except the CLI needs to be that of the originator (from the USERS point of view) so they can decide if they want to accept the call. Here in the UK, this is where the issues begin... the carriers here don't like it if your sending CLI for other countries, that don't match what they think they should receive from that connecting carrier. Eg, if a call coming to them is 13 digits, but they only expect 11 from that carrier, then they cut the digits. This turns a US originated call into a Southampton UK originated call! So I was hoping that handing the call back to the network in the traditional sense would make it their problem and not mine... lol Thanks, Adrian -- -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.google.com/search?hl=enq=asterisk+302+redirect+sipbtnG=Sear ch I believe you're looking for a 302 Redirect? Sorry if you're not, but that sounds like what you want ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trying directrtpsetup
On Sun, May 25, 2008 at 10:45 AM, ronald ramos [EMAIL PROTECTED] wrote: Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool or can u just sniff? You could type rtp debug on your console and if your call is in progress and you don't get a stack of rtp debug messages flying past then you will know the rtp is not going through your Asterisk server. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logical AND
Hi Steve,
I can see what yours does, but I still get the same end result (even
though theres only a single 0 result now)
:
exten = s,n,ExecIf( $[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d9a9a0,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d9a9a0, 0
|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d9a9a0,
0010123456789) in new stack
-
hello,
you should try this:
exten = s,n,ExecIf($[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]]|Set|PSTN_NUM=001${PSTN_NUM})
cause the AND Operator is another thing to work, so the result at your
way look like this ExecIf(11 | ...) and with my way it looks like this
ExecIf($[11]|...) which is the right syntax for it.
best regards
steve smith
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Re: [asterisk-users] Asterisk Tag Berlin live notes page
Hello Randy,I would like to read your daily updates in regards to Asterisk Tag in Berlin, Germany.Regards,Eddie Original Message Subject: [asterisk-users] Asterisk Tag Berlin live notes page From: randulo [EMAIL PROTECTED] Date: Sun, May 25, 2008 2:46 am To: "VOIP Users Conference" [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial Discussion" [EMAIL PROTECTED].com, "Commercial and Business-Oriented Asterisk Discussion" [EMAIL PROTECTED].com Hi, If you're interested in what's happening in Berlin at Asterisk Tag for the next few days, you can look here: http://www.scribblelive.com/Thread.aspx?Id=815 I will try to keep an event trail going and notes if interviews are posted. Anyone can post questions and I don't think there's even a need to create an account or log in; You can also follow asterisktag on some of these http://twitter.com/asterisktag http://twitter.com/randulo http://twitter.com/wintermeyer YATS! IF YOU ARE IN BERLIN please watch this space and contribute: http://www.scribblelive.com/Thread.aspx?Id=815 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logical AND
On Sunday 25 May 2008 07:10:22 Adrian Marsh wrote:
exten = s,n,ExecIf( $[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d9a9a0,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d9a9a0, 0
|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d9a9a0,
0010123456789) in new stack
There's an extra space between the opening ( and the opening $[, so
the result is space-zero, which is not the same thing as zero. Any string
that is not exactly 0 (or the empty string), such as foo, , or 0 is
true.
--
Tilghman
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Re: [asterisk-users] Logical AND
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 25 May 2008 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Logical AND
On Sunday 25 May 2008 07:10:22 Adrian Marsh wrote:
exten = s,n,ExecIf( $[ $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ] ] |Set|PSTN_NUM=001${PSTN_NUM})
-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/427-b7d9a9a0,
0123456789) in new stack
-- Executing [EMAIL PROTECTED]:9] ExecIf(SIP/427-b7d9a9a0, 0
|Set|PSTN_NUM=0010123456789) in new stack
-- Executing [EMAIL PROTECTED]:10] NoOp(SIP/427-b7d9a9a0,
0010123456789) in new stack
There's an extra space between the opening ( and the opening $[, so
the result is space-zero, which is not the same thing as zero. Any
string
that is not exactly 0 (or the empty string), such as foo, , or
0 is
true.
--
Tilghman
-
I did wonder where the extra spaces were coming from, but I thought that
was where the quotes were supposed to come into play... Well that got
it working so thanks guys..
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[asterisk-users] No Audio on Meetme
Hi All, What could be the cause why there is no audio coming form the participants. ztdummy is loaded, ZTDUMMY/1 (source: HRtimer) 1. I can hear Please enter your PIN, User blah blah has enttered...etc etc But when the particpants talk, we hear nothing. What are the possible mistakes i did on these? TIA Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logical AND
On Sunday 25 May 2008 09:38:28 Adrian Marsh wrote: I did wonder where the extra spaces were coming from, but I thought that was where the quotes were supposed to come into play... Well that got it working so thanks guys.. The quotes generally aren't supposed to take care of spaces; they are generally used so that variables which are unset do not create an invalid expression, i.e. $[ != foo] is valid, but $[ != foo] is not. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] End call behaviour
When I exit voicemail or an inbound caller hangs up I hear a busy signal for a few seconds before Asterisk terminates the call. I thought this behavior was handled in the dial plan with a Hangup() command? How can I correct this? Thanks, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP call ring timeout
I had my incoming call time set 120 seconds before going to voicemail, apparently this timeout is longer than some existing timeout of ~60 seconds and the call terminates before it reaches my voicemail command. Is this an Asterisk default setting or could this be something on my SIP providers end? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider
Shaun schrieb: Hi All, This is puzzling me greatly. The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to Asterisk are SIP clients. Codec throughout G729 (only have 1 license on Asterisk server loaded though). When calling the SIP clients from PAP2T I can't hear them but they can hear me. If I call from PAP2T through Asterisk using?IAX2 to a VOIP provider there is speach in both directions! Any suggestions? Thanks Shaun check your firewall/nat settings. If your setup will work for around 5 minutes after you have rebooted the pap2t then you have to active the nat keep alive and nat mapping service in the pap2t. best regards steve smith DEAR STEVE, THANKS, I DID CHECK THEM AND THEY NEEDED TO BE ACTIVATED. THIS DOES NOT SOLVE THE PROBLE. STILL ONE WAY SPEECH WHEN CALLING PAP2T --ASTERISK--SIP DEVICE ATTACHED. HOWEVER PAP2T--ASTERISK--SIP PROVIDER WORKS FINE AS WELL AS SIP DEVICE ATTACHED--ASTERISK--SIP RPOVIDER. ANY SUGGESTIOPS WELCOME___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I invite you to join my Ziki Network !
Bonjour, connaissez-vous Ziki.com ? Ziki est le lieu où chacun peut se promouvoir librement. Promouvoir votre identité, vos contenus, vos compétences, vos services... Etre référencé en 1ère position sur le moteur de recherche Google. Découvrir et contacter gratuitement des personnes qui vous ressemblent. Check out my Ziki at : http://www.ziki.com/en/nacef-labidi?invitation_code=e0b027b93b4af38213c8c94dc6fd058e Then use this link to accept the invitation and register for your FREE Ziki account: http://www.ziki.com/en/signup?invitation_code=e0b027b93b4af38213c8c94dc6fd058e (or copy and paste the link into your browser's address bar) It's free, fun and easy! Create your webpage now and join my network. See you there ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider
In your account settings (sip.conf) for the PAP2T device, do you have reinvite enabled for one or both.. the SIP provider and/or the PAP2T device ? From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Shaun Wingrin [EMAIL PROTECTED] Sent: Sunday, May 25, 2008 3:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider Shaun schrieb: Hi All, This is puzzling me greatly. The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to Asterisk are SIP clients. Codec throughout G729 (only have 1 license on Asterisk server loaded though). When calling the SIP clients from PAP2T I can't hear them but they can hear me. If I call from PAP2T through Asterisk using?IAX2 to a VOIP provider there is speach in both directions! Any suggestions? Thanks Shaun check your firewall/nat settings. If your setup will work for around 5 minutes after you have rebooted the pap2t then you have to active the nat keep alive and nat mapping service in the pap2t. best regards steve smith DEAR STEVE, THANKS, I DID CHECK THEM AND THEY NEEDED TO BE ACTIVATED. THIS DOES NOT SOLVE THE PROBLE. STILL ONE WAY SPEECH WHEN CALLING PAP2T --ASTERISK--SIP DEVICE ATTACHED. HOWEVER PAP2T--ASTERISK--SIP PROVIDER WORKS FINE AS WELL AS SIP DEVICE ATTACHED--ASTERISK--SIP RPOVIDER. ANY SUGGESTIOPS WELCOME ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End call behaviour
What type endpoint do you have ? Channel bank perhaps ? Is it an ATA ? a SIP phone ? From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Joseph L. Casale [EMAIL PROTECTED] Sent: Sunday, May 25, 2008 11:58 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] End call behaviour When I exit voicemail or an inbound caller hangs up I hear a busy signal for a few seconds before Asterisk terminates the call. I thought this behavior was handled in the dial plan with a Hangup() command? How can I correct this? Thanks, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trying directrtpsetup
RTP DEBUG IP Device IP Address from the asterisk CLI From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of ronald ramos [EMAIL PROTECTED] Sent: Sunday, May 25, 2008 4:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] trying directrtpsetup Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool or can u just sniff? regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End call behaviour
What type endpoint do you have ? Channel bank perhaps ? Is it an ATA ? a SIP phone ? Hi, These are SIP phones (Snom M3's and Astra 480i's), I didn't notice this when I was testing with my softphone but I cant recall :) Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with e1 connection
I have a lot of these messages popping up in my mesages, E1 connection shows provisioned up active but I cant seem to be able to make a call. It was previously working before but stopped working after I did a reboot to the box this weekend. Anything I am missing out May 26 08:21:38 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:21:38 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:16 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:16 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:54 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:54 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:23:32 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:23:32 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 usr/local/etc/asterisk/zapata.conf [channels] signalling=pri_cpe context=tme1_incoming group=1 callgroup=1 pickupgroup=1 priindication=outofband switchtype=euroisdn context=tme1_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=no faxdetect=no overlapdial=yes prilocaldialplan=national pridialplan=unknown channel = 1-15 channel = 17-31 signalling=pri_cpe context=tme1_incoming group=2 callgroup=2 pickupgroup=2 priindication=outofband switchtype=euroisdn context=tme1_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=yes faxdetect=incoming overlapdial=yes prilocaldialplan=national pridialplan=unknown channel = 32-46 channel = 48-62 signalling=pri_net context=md110_incoming group=3 callgroup=3 pickupgroup=3 priindication=outofband switchtype=euroisdn context=md110_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=no faxdetect=no overlapdial=yes prilocaldialplan=unknown pridialplan=unknown channel = 63-77 channel = 79-93 signalling=pri_net context=md110_incoming group=4 callgroup=4 pickupgroup=4 priindication=outofband switchtype=euroisdn context=md110_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=no faxdetect=no overlapdial=yes prilocaldialplan=unknown pridialplan=unknown channel = 94-108 channel = 110-124 /usr/local/etc/zaptel.conf loadzone=my defaultzone=my span=1,1,0,ccs,hdb3,crc4 span=2,2,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 bchan=63-77 dchan=78 bchan=79-93 bchan=94-108 dchan=109 bchan=110-124 specs 1.80Ghz Dual Core with Sangoma 104, running asterisk 1.2.26.2 ontop of FreeBSD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End call behaviour
Is reinvite set tp yes for the device? -Original Message- From: Joseph L. Casale [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 5/25/08 7:52 PM Subject: Re: [asterisk-users] End call behaviour What type endpoint do you have ? Channel bank perhaps ? Is it an ATA ? a SIP phone ? Hi, These are SIP phones (Snom M3's and Astra 480i's), I didn't notice this when I was testing with my softphone but I cant recall :) Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
