Re: [asterisk-users] What does this error message mean
Show sip.conf and extensions.conf related part. Maybe I misread but did you mention you have a exten... Line in sip.conf??? The error is because the received user is not the same as the configured one. --Mensaje original-- De: Ira Remitente: asterisk-users-boun...@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] What does this error message mean Enviado: 26 Mar, 2010 21:01 At 05:47 PM 3/26/2010, you wrote: You have used this same username/password combination for another SIP client, or maybe the same one but with different IP. Even when that one is offline from some time on, Asterisk doesn't renew it's internal database, so still thinks it might be somewhere there. Why thanks. The sad part is it means madly flailing has exactly the opposite effect as you might expect. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need help on setup rtp directly between 2 sipclients
Try setting canreinvite and nat to no for those extensions. Saludos, Juan E. Rodríguez -Original Message- From: Alyed al...@vivoxie.com Date: Fri, 26 Mar 2010 10:56:50 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] need help on setup rtp directly between 2 sip clients -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)
On 25 March 2010 02:42, James Lamanna jlama...@gmail.com wrote: Hi, I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). Read all the discussion about many spans - and I've run 16 E1 spans in one box, and run 8 spans under 200+ concurrent calls. Your 1st gen TE410 card is very old and I'd suggest to contact Digium about a firmware upgrade or a hardware upgrade. As for the spans going down:- 1) Make sure you are syncing your clock to your telco (span=1,1,0,...) 2) Make sure you are using the right IDE driver module for your chipset and not the generic one 3) Avoid long runs (25m?) of unscreened cable on the T1/E1 span Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Background noise
Thank you very mutch Philip, i'll use these commands and get back with the output. 2010/3/26 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Hi! it should be some commands that can give me a better idea about the codecs, if anyone know them, please help! Use sip show channels and iax show channels and look at the Format column. About the Polycom devices: Others will have to help you there. I have no good guess why you might have the issue only on speakerphone, but not in handset mode. Could it maybe be some kind of electrical grounding issue (instead of something caused by transcoding)? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] configure the sound for inbound calls
Hi Salah, what's the problem? For playbacks upload a soundfile to your asterisk /var/lib/asterisk/sounds/hello.wav and setup the Routing Script to Playback(hello); reload asterisk and watch asterisk and aheeva logfiles. And yes there is a possibility to retrieve customer information from your CRM as long as you get customer phone number. Hello All, I do have asterisk installed for a call centre with aheeva application and i would like to know how to configure the sound for the inbound calls and if there is any possibility for agent to receive a file with the phone number and name of clients: For your information there is no problem related to the outbound call An help would be appreciated Kind Regards Salah. -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall
Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] Thank you. -- James (Please CC me on all replies) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_ss7 issue
Gentler reminderany body to help me pls On Tue, Mar 23, 2010 at 2:20 PM, Kasun Daminda damind...@gmail.com wrote: Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server. This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total 4034145216, 4031118560 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 5, tx: 3/3, sentseq/lastack: 95/95, total 4030833616, 4028245568 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4034149872, 4031123216 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 2, tx: 3/3, sentseq/lastack: 100/100, total 4030838272, 4028250224 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 3/4, sentseq/lastack: 127/127, total 4034154480, 4031127824 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 5, tx: 4/4, sentseq/lastack: 100/101, total 4030842880, 4028254832 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 3, tx: 1/4, sentseq/lastack: 127/127, total 4034159456, 4031132800 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 0, tx: 0/4, sentseq/lastack: 100/101, total 4030847840, 4028259792 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 4, tx: 4/4, sentseq/lastack: 127/127, total 4034164432, 4031137776 linkset siuc, link l5, schannel 1, sls 1, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4030852816, 4028264768 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 5, tx: 3/4, sentseq/lastack: 127/127, total 4034169312, 4031142640 linkset siuc, link l5, schannel 1, sls 1, PROVING, rx: 4, tx: 2/4, sentseq/lastack: 127/127, total 4030857696, 4028269632 [r...@localhost ~]# asterisk -rx ss7 link status And I get a log as [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Excessive poll delay 12718! [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [r...@localhost ~]# Can anybody help me on this. It will be great help. Kind Rgds Daminda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with polarity reverse
Hi, I have a problem with polarity reverse I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and analog card is Sangoma a400 with fxo ports this is my config [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no answeronpolarityswitch=yes and then i call from sip to mobile over gsm gw (nokia 32) which have a polarity reverse i pick up the mobile phone in sip phone i hear that polarity revers was but at the in asterisk core show channels verbose Channel Context ExtensionPrio State Application Data CallerID Duration Accountcode BridgedTo DAHDI/11-1 from-zaptel 8685
[asterisk-users] migration
My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does he need his current phones? How will they ring if I port the number? Should I simply have him remove the phones and I can send the calls to his cell phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] migration
Re: [asterisk-users] migration Of geese or ducks? (A more descriptive subject will yield better replies.) On Sat, 27 Mar 2010, Thomas Perron wrote: My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does he need his current phones? How will they ring if I port the number? Should I simply have him remove the phones and I can send the calls to his cell phones? If his current POTS line number rings his current phones, and you port the POTS line number to a SIP provider, what would you expect to ring them? You could dial his cell phones or any other endpoint. You could connect the existing phones to your Asterisk server using ATAs or channel banks or ??? What is in the best interest of your client? There isn't enough information present to offer a reasonable opinion. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to configure xorcom on Suse 11
I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi from trunk (last thursday) # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools dahdihardware -v sees the box but no spans. I get a syslog _full_ of this: 2010-03-25T21:35:22.338865-10:00 pbx kernel: [185556.006494] INFO-xpp_usb: XUSB: Xorcom LTD -- Astribank -- FPGA 2010-03-25T21:35:22.338878-10:00 pbx kernel: [185556.006695] INFO-xpp: XBUS-00: [usb:X1037246] Activating 2010-03-25T21:35:59.930296-10:00 pbx kernel: [185593.410290] INFO-xpp: XBUS-00: [usb:X1037246] Disconnecting 2010-03-25T21:35:59.930374-10:00 pbx kernel: [185593.410305] INFO-xpp: XBUS-00: [usb:X1037246] Deactivating 2010-03-25T21:35:59.930527-10:00 pbx kernel: [185593.410324] INFO-xpp: XBUS-00: [usb:X1037246] Release XPDS 2010-03-25T21:35:59.930696-10:00 pbx kernel: [185593.410423] INFO-xpp: XBUS-00: [usb:X1037246] Atribank Remove 2010-03-25T21:35:59.930776-10:00 pbx kernel: [185593.410455] INFO-xpp: XBUS-00: [usb:X1037246] Astribank Release 2010-03-25T21:35:59.930854-10:00 pbx kernel: [185593.410730] INFO-xpp_usb: xusb-0 (usb-:00:1a.7-4) [X1037246]: now disconnected 2010-03-25T21:36:14.944476-10:00 pbx kernel: [185608.613407] INFO-xpp: revision Unknown MAX_XPDS=64 (8*8) 2010-03-25T21:36:14.944488-10:00 pbx kernel: [185608.613422] INFO-xpp: FEATURE: without BRISTUFF support 2010-03-25T21:36:14.944493-10:00 pbx kernel: [185608.613436] INFO-xpp: FEATURE: with PROTOCOL_DEBUG 2010-03-25T21:36:14.944498-10:00 pbx kernel: [185608.613601] INFO-xpp: FEATURE: with sync_tick() from DAHDI 2010-03-25T21:36:14.946463-10:00 pbx kernel: [185608.615753] INFO-xpp_usb: revision Unknown 2010-03-25T21:36:15.162342-10:00 pbx kernel: [185608.831539] INFO-xpp_usb: XUSB: Xorcom LTD -- Astribank -- FPGA 2010-03-25T21:36:15.166858-10:00 pbx kernel: [185608.834398] INFO-xpp: XBUS-00: [usb:X1037246] Activating 2010-03-25T21:41:37.626186-10:00 pbx kernel: [185931.095914] INFO-xpp: XBUS-00: [usb:X1037246] Disconnecting 2010-03-25T21:41:37.626263-10:00 pbx kernel: [185931.095923] INFO-xpp: XBUS-00: [usb:X1037246] Deactivating 2010-03-25T21:41:37.626417-10:00 pbx kernel: [185931.095941] INFO-xpp: XBUS-00: [usb:X1037246] Release XPDS 2010-03-25T21:41:37.626579-10:00 pbx kernel: [185931.096024] INFO-xpp: XBUS-00: [usb:X1037246] Atribank Remove 2010-03-25T21:41:37.626659-10:00 pbx kernel: [185931.096126] INFO-xpp: XBUS-00: [usb:X1037246] Astribank Release 2010-03-25T21:41:37.626744-10:00 pbx kernel: [185931.096588] INFO-xpp_usb: xusb-0 (usb-:00:1a.7-4) [X1037246]: now disconnected 2010-03-25T21:41:47.445482-10:00 pbx kernel: [185941.114843] INFO-xpp: revision Unknown MAX_XPDS=64 (8*8) 2010-03-25T21:41:47.445492-10:00 pbx kernel: [185941.114855] INFO-xpp: FEATURE: without BRISTUFF support 2010-03-25T21:41:47.445497-10:00 pbx kernel: [185941.114868] INFO-xpp: FEATURE: with PROTOCOL_DEBUG 2010-03-25T21:41:47.445503-10:00 pbx kernel: [185941.115024] INFO-xpp: FEATURE: with sync_tick() from DAHDI 2010-03-25T21:41:47.447466-10:00 pbx kernel: [185941.117112] INFO-xpp_usb: revision Unknown 2010-03-25T21:41:47.665867-10:00 pbx kernel: [185941.333559] INFO-xpp_usb: XUSB: Xorcom LTD -- Astribank -- FPGA Any hints as to what I'm doing wrong would be much appreciated. (here's some project background for the curious http://babyis60.wordpress.com/2010/02/25/the-island-phone-system-adventure/ ) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall
From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify If you turn on *qualify* in the configuration of a SIP device in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf, Asterisk will send a SIP OPTIONShttp://www.voip-info.org/wiki/view/SIP+method+optionscommand regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER functionhttp://www.voip-info.org/wiki/view/Asterisk+func+sippeer, and inversely this function will only provide status information for peers which have *qualify=yes*. My guess is that your Nat/firewall is closing the connection after some time the phone is idle, so this way Asterisk will make sure to always have communication going trhough that connection so your NAT/firewall won't just close it. try playing with qualifyfreq as well. Let us know if it helped. Alyed 2010/3/27 James Lamanna jlama...@gmail.com Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] Thank you. -- James (Please CC me on all replies) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to configure xorcom on Suse 11
Xorcom hardware uses three layers; you must resolve issues in the following order: 1. USB 2. Dahdi 3. Asterisk I suspect you're having trouble with the usb layer. Run lsusb It will display a line like this if the firmware isn't loaded: Bus 001 Device 004: ID e4e4:1161 If it is e4e4:1162 then the firmware is loaded. You can manually load the firmware like this: /usr/share/dahdi/xpp_fxloader load or /usr/share/dahdi/xpp_fxloader usb -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
