[asterisk-users] X100P FXO PCI card not receiving calls
Hello users, i have recently purchased Authentica x100p Fxo card for asterisk 1.4 i have following settings # /etc/zaptel.conf fxsks=1 loadzone=in defaultzone=in # /etc/asterisk/zapata.conf [channels] context=from-pstn usecallerid=no hidecallerid=yes immediate=no signalling=fxs_ks echocancel=yes channel = 1 #/etc/asterisk/extensions.conf [from-pstn] exten = s,1,Answer() exten = s,n,Playback(vm-intro) exten = s,n,Hangup() # lspci -vv 11:00.0 Communication controller: Motorola Wildcard X100P Subsystem: Efar Microsystems: Unknown device 0001 Flags: bus master, medium devsel, latency 32, IRQ 193 I/O ports at 2000 [size=256] Memory at ed10 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 # cat /proc/interrupts CPU0 CPU1 0: 921447 931323IO-APIC-edge timer 1: 5 5IO-APIC-edge i8042 8: 0 1IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 34 33IO-APIC-edge i8042 14: 140339 90649IO-APIC-edge ide0 15: 0 0IO-APIC-edge libata 169: 38 25 IO-APIC-level ehci_hcd, uhci_hcd, uhci_hcd 177: 0 0 IO-APIC-level uhci_hcd, uhci_hcd 185: 9688 9482 IO-APIC-level libata, ehci_hcd, uhci_hcd, uhci_hcd 193:116 36 IO-APIC-level wcfxo 209: 15304 0 PCI-MSI eth0 NMI: 1 0 LOC:18441221843838 ERR: 0 MIS: 0 #cat /proc/zaptel/1 Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 WCFXO/0/0 FXSKS (In use) i am unable to make incoming calls route to the asterisk. i didnt see anything in my asterisk CLI. can anybody advise?? Thanks in advance Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regcontext regexten
Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? Sure... let's say I have a phone with the following configuration in sip.conf: [myphone] type=friend context=inside host=dynamic ; phone will register w/ Asterisk secret=mysecret regcontext=some-context regexten=6123 Thank Jared. So I have one more and last question about regcontext. Where do asterisk create context some-context ? I see context by taping dialplan show some-context in CLI but I dont know in which config file it's created. Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM Phones Displays NR Frequently
Chris Bagnall wrote: First things first. You are running /very/ old versions of firmware - particularly on the 300 and 320. Upgrade them. I've been running 7.3.14 for some time without a problem, though it appears that 7.3.23 is now out. I concur about upgrading the software, but I'd stick with 7.3.14 for now - at least until there's more feedback on stability of .23 and .24. On the OP's original question, I have noticed Snoms (of any version) aren't very good at handling DNS failure - they tend to cache DNS lookup failure almost indefinitely. If DNS lookups to your registrar occasionally fail, you might want to specify registrar via IP rather than by name. Regards, Chris We always upgrade to v7.3.7 and find it very stable -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regcontext regexten
On Aug 10, 2009, at 9:52 AM, harry R wrote: Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? Sure... let's say I have a phone with the following configuration in sip.conf: [myphone] type=friend context=inside host=dynamic ; phone will register w/ Asterisk secret=mysecret regcontext=some-context regexten=6123 Thank Jared. So I have one more and last question about regcontext. Where do asterisk create context some-context ? I see context by taping dialplan show some-context in CLI but I dont know in which config file it's created. Harry, The context is created in the running dialplan. It's not stored in a configuration file. You can however create this context in extensions.conf or extensions.ael and asterisk will use that. Basically how it works is: On a sip register asterisk checks if the regcontext exists. if not it will create it, if it exists asterisk will add the regexten to it. Michiel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues with sound quality and HDLC
Hi all, We've been having a very frustrating time with our Asterisk install (well, okay, actually Switchvox). We have an open ticket with Digium/Switchvox but I was wondering if anyone here might have some helpful tips. We're basically getting glitching on the line, and in the error logs it's showing: chan_dahdi.c: PRI got event: HDLC Bad FCS We're running our PBX on the recommended PDSMA+ motherboard (so timing should have been tested), and we've tried two different Wildcards (TE207P). We've tried removing the hardware echo canceler. We've tried moving the card to different slots. We've tried an entirely different system (Dell SC400 tower - unsupported, I know, but I wanted to get a comparison). ATT have tested the line (although I'm getting them onsite to do a more thorough check), and we've replaced all the cables we can. Unfortunately, since I don't have root access to the box, I can't run console tests - but I've checked things like interrupts and it appears the card is not sharing. Anyone have any clue as to what it could be? Thanks! S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regcontext regexten
/etc/asterisk/extensions.conf /etc/asterisk/extensions.ael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of harry R Sent: Monday, August 10, 2009 1:22 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] regcontext regexten Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? Sure... let's say I have a phone with the following configuration in sip.conf: [myphone] type=friend context=inside host=dynamic ; phone will register w/ Asterisk secret=mysecret regcontext=some-context regexten=6123 Thank Jared. So I have one more and last question about regcontext. Where do asterisk create context some-context ? I see context by taping dialplan show some-context in CLI but I dont know in which config file it's created. Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] context does not work
Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Try prefix your extension in extensions.conf with _, e.g. exten = _123,1,... -- Sent from mobile device On Aug 10, 2009, at 6:55 AM, Patrick Plattes patr...@erdbeere.net wrote: Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Thanks for the fast reply, but it does not help :-(. Bye, Patrick On Mon, Aug 10, 2009 at 1:01 PM, Alex Balashovabalas...@evaristesys.com wrote: Try prefix your extension in extensions.conf with _, e.g. exten = _123,1,... -- Sent from mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Patrick Plattes wrote: extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) What does dialplan show testing output? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
Word of advice: When you try SIP clients, focus on how the far-end is hearing you, not whether you can hear them. In my experience, that's where 90% of the deal-breakers lie with the iPhone. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
What does dialplan show testing output? [ Context 'testing' created by 'pbx_config' ] '261' = 1. Noop(261) [SIP] '262' = 1. Noop(262) [SIP] '263' = 1. Noop(263) [SIP] '264' = 1. Noop(264) [SIP] '_8001187e0' = 1. Dial(SIP/263) [pbx_config] -= 5 extensions (5 priorities) in 1 context. =- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Try putting exten = 8001187e0,1,Dial(SIP/263) in the [default]-context. I have the same issue. Apparently your SIP-provider send calls to your Asterisk-box from multiple IP's so that Asterisk cannot match the inbound call on source IP and therefore sends it to the default-context. Jonas. On Mon, 2009-08-10 at 13:26 +0200, Patrick Plattes wrote: Thanks for the fast reply, but it does not help :-(. Bye, Patrick On Mon, Aug 10, 2009 at 1:01 PM, Alex Balashovabalas...@evaristesys.com wrote: Try prefix your extension in extensions.conf with _, e.g. exten = _123,1,... -- Sent from mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
jonas kellens wrote: I have the same issue. Apparently your SIP-provider send calls to your Asterisk-box from multiple IP's so that Asterisk cannot match the inbound call on source IP and therefore sends it to the default-context. I'd second this suggestion. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Hi Andrew, it didn't help. Which version of Asterisk do you use? Thanks On Mon, Aug 10, 2009 at 1:55 PM, Andrew Thomasa...@datavox.co.uk wrote: Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Hi Jonas, that works fine, but I think its just a work arround and not a real fix :-). For the moment it is okay and I'll try to fix the error next days. Thanks, Patrick Plattes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
i faced the same problem with callcentric.. when i register i had to add the extension .. like this egister = 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID which caused my context to go to the default context and never use the one i already setup.. so removing the extension in the registration string will solve the issue for me.. and i think it will do the same for you. regards -- AHD Tarek Sawah Date: Mon, 10 Aug 2009 12:55:41 +0200 From: patr...@erdbeere.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
V1.6.1.0 [9290740] type = peer username = 9290740 fromuser = 9290740 secret = you-wish! host = sipgate.co.uk fromdomain = sipgate.co.uk insecure = port,invite context = inbound caninvite = no canreinvite = no nat = yes disallow = all allow = ulaw allow = alaw dtmfmode = info qualify = 5000 That works for me. Any inbound call to my 9290740 number goes to my inbound context and does what it should. PS - Don't forget to do a 'sip reload' when you change the sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 13:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] context does not work Hi Andrew, it didn't help. Which version of Asterisk do you use? Thanks On Mon, Aug 10, 2009 at 1:55 PM, Andrew Thomasa...@datavox.co.uk wrote: Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Placing a SIP Call on Hold
Venkateshwarlu Kakkireni wrote: Can I mute a connected channel? Also, can I play MOH on a specific channel without transferring it to a different MOH extension or MeetMe? I am pretty new to the dialplans your help would be very much appreciated... Thanks in advance... No, Asterisk does not offer methods to do those operations right now; I'm sure they could be added, but it hasn't been done. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer after pickup
I am probably just being stupid again, but... I have some non-SIP phones which are set up for doing transfers by DTMF, by simply adding T or t to the appropriate Dial options. This works quite well in general. They can also do non-directed call pickup with *8. However, after a call pickup they can't transfer the call by DTMF -- there is no Dial command where I can add the t or T option. How do I configure *8 to allow blind and attended transfers for the person who dialled *8? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AddQueueMember with Agents.conf
Hello Team As you are all aware, digium has removed agentcallbacklogin as from 1.6. Is anyone knows any work around to have say 20seats (SIP Clients), 100 agents call center for which user will have to login to the queue dynamically from any extension and yet populate queue information with own's information instead of SIP or Local channels for reporting purpose I have tried both AddQueueMember(QueueName,Local/Extension@context) and AddQueueMember(queueName,agent/agentcodeSIP/extension) in vain Please advice Thanks Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A problem with recoding agents calls via monitor
Hooman Peiro escribió: Hello everyone, Hey I can not get the name of the recoding file of agents calls. I set agents.conf as following: ; Insert into CDR userfield a name of the the created recording ; By default it's turned off. createlink=yes ; as you can see I set createlink=yes so name of the recoding file should be inserted in userfield column of cdr. But it doesn't work. because the file is created but userfield is empty. Any solution to this problem is really appreciated. If you need more information you can ask me. Are you storing the CDR in a database? What kind? Or are you checking the plain CSV CDR file? Supposing that you have a MySQL backend, you should have the userfield=1 setting in cdr_mysql.conf to tell the backend to save the userfield of the record. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
On Mon, Aug 10, 2009 at 4:53 AM, Alex Balashovabalas...@evaristesys.com wrote: Word of advice: When you try SIP clients, focus on how the far-end is hearing you, not whether you can hear them. In my experience, that's where 90% of the deal-breakers lie with the iPhone. Absolutely right! When testing call quality I always used to call my mother and after several minutes ask her how well she heard me, etc. We joked about how she was my VoIP tester. Now that she's been gone a few years, my brother, the user from Hell can't fill those shoes. Regardless of call setup or source: - he will ALWAYS find something to complain about and often it's an issue with his phone - his phones all sound like crap - he lets the batteries on his cordless run way down causing beeping and clicking - he is unable to describe in any useful way what the problem is, he just says it stinks I miss being able to call my mom on four different providers and ask her which was better :) Back to your comment Alex, I agree whole heartedly which is why being able to record (like the Skype test call) is better than an echo test for checking clients out locally. Then you can test by leaving messages on a PSTN connected phone of know quality. Then and only then, call people who you know have an ear for sound quality of a call and are in a location quiet enough to judge. I was about to post on this thread that I have contacted the makers of iSip and they got back to me, we're working on a fix. We because I did not realize something specific to the iPhone platform. Because of the App store, an new test verison of an app can not be installed because it is not approved by Apple. Developers can make ad hoc versions for a single phone, though and the need the UDID to do this. I never heard of UDID before, but it's like a MAC. I hope the iSip people will be able to fix the issue because I like the client a lot. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
On Fri, Aug 7, 2009 at 7:25 PM, Pascal Bruno tipas...@gmail.com wrote: Where you able to compile DAHDI in a virtual environment? How about skype for asterisk? Has anyone tried that in a virtual environment? Seems like to register the license, digium tool is looking for a connection on eth0, and in a virtual environment I see the name as vnet0 or vnet1. At least that what I see on godaddy's virtual servers. I did that under VMWare (Server / formerly GSX), including the Skype for Asterisk, and it works (only after upgrading to 1.6.1.3-rc1, earlier version crashed after Skype call setup, but that's not related to the VM, but an asterisk bug...). Though it is merely a test environment, I haven't even tried more than one simultaneous call. HTH, -- Shimi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Outgoing Trunk
On Thu, 06 Aug 2009 21:28:01 +0530 wrote On 6 Aug 2009, at 16:32, kumarshantanu wrote: Hello Everybody, Hi. I have a genuine problem in Asterisk setup. Ok. I have three inbound trunks in my asterisk box, everything is What kind of trunks. These are sip trunks working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Ok. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should get different no and if extn: 202 make an outgoing call the recipient should get different one. Ok. Please can someone help me in this. If you show us some config, tell us trunk types and generally 'giving us something to go on. What config you want from me. I am not very much friendly to asterisk, for now I manage it from freePBX. Let me please know if I can provide you some more information Thanks Shantanu Steve Heh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7940g
I have 6 Cisco 7940g phones and I would like to add them to my Asterisk 2.6.2 box. My SNOM 320 work just fine but I cannot get the Cisco's to register. They pull the TFTP P0S3-08-9-00 just fine. I change the NAT to no but it still does not register. Please advise. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Outgoing Trunk
Assuming you are configuring your Asterisk using the configuration
files -- if you want the caller ID on phone calls between users to be
the same as the caller id on calls made with the trunk lines, set the
external caller id information for the users in sip.conf (i.e.
callerid=9995551212) or to simply change the caller id information
for calls made on the trunk lines, modify the extensions.conf file by
inserting something like the following line before the Dial command.
An example:
[outgoing-calls-context]
exten = _NXXNXX,1,Set(CALLERID(num)=${EXTCALLERID})
exten = _NXXNXX,2,Dial(${TRUNK}/${EXTEN})
The problem here is that if you are using text files for configuration
you are should know this, so this advice is either almost redundant or
not helpful. What kind of Asterisk GUI tool are you using?
On Mon, Aug 10, 2009 at 9:47 AM,
kumarshantanushantanu1...@rediffmail.com wrote:
On Thu, 06 Aug 2009 21:28:01 +0530 wrote
On 6 Aug 2009, at 16:32, kumarshantanu wrote:
Hello Everybody,
Hi.
I have a genuine problem in Asterisk setup.
Ok.
I have three inbound trunks in my asterisk box, everything is
What kind of trunks.
These are sip trunks
working fine but the only problem is when any user make an out-
going call through his/her extension it goes with same number labeled
on this.
Ok.
Can we set each of these lines to have fixed outgoing numbers
like if extn: 201 make an outgoing call the recipient should get
different no and if extn: 202 make an outgoing call the recipient
should
get different one.
Ok.
Please can someone help me in this.
If you show us some config, tell us trunk types and generally 'giving
us something to go on.
What config you want from me. I am not very much friendly to asterisk,
for now I manage it from freePBX. Let me please know if I can provide you
some more information
Thanks
Shantanu
Steve
Heh
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[asterisk-users] SNOM 870
Anybody tried one with Asterisk yet ? Views ? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
On Aug 6, 2009, at 9:43 PM, randulo wrote: Hi, I've tried two SIP clients so far and both have unusable outgoing audio quality. Skype app sounds fine, and recording the same mic sounds fine, so I can only assume there is an issue with the clients themselves. Both clients allow you to register and make calls via SIP with any abitrary provider and credentials, so they'll work with Asterisk. I've tried them with two good providers and one has unrecognizable audio and the other has noises as if the cable was badly soldered. I've never experienced such troubles with regular SIP clients. Anyone have any recommendations? Contact me off-list and I can get you a beta version of RF Dial. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7940g
2009/8/11 Chuck Coleman p...@2cci.com I have 6 Cisco 7940g phones and I would like to add them to my Asterisk 2.6.2 box. My SNOM 320 work just fine but I cannot get the Cisco’s to register. They pull the TFTP P0S3-08-9-00 just fine. I change the NAT to * no* but it still does not register. Please advise. It just works... unless there's something very wrong with your config... let us see your config... Erm, * 2.6.2? d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sflphone questions
I want to set sflphone as extension on asterisk. I have a sip account/DID with vitelity.net. Not sure what to put in the wizard: alias ??? hostname ??? is this the asterisk server hostname, or the hostname where my sflphone is sitting on the lan (it's a home network) username ??? is this the assigned extension number? password ??? is this the assigned extension number password? Any help appreciated. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to adjust the timeout to send CANCEL?
I want to adjust the timeout to send CANCEL after sending out INVITE. How to do it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] waitfordialtone patch
Does anyone have a working patch for the following issue on Asterisk 1.4.26 or an earlier version of 1.6 than 1.6.2? It looks like it got committed somewhere after 1.6.1 was branched and is only available natively in Asterisk 1.6.2.x. https://issues.asterisk.org/view.php?id=12382 I have a crappy Cisco IAD that doesn't seem to be consistent in it's operation when the Analog card attempts to place an outbound call. I've inserted several pauses in the dialing string which helps in most cases, but in the few cases with the Cisco is ready immediately, the extra pauses create other issues. This seems like the best work around, but I'm not really ready to try Asterisk 1.6.2.x in a production setting. Thanks, Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
