[asterisk-users] SER+iptables+Asterisk
I have ser sitting on my iptables nat box and my asterisk box on the lan . Ser does forwarding so that any requests (register,invite,ack,...) to the nat box at 5060 r sent to my asterisk box on the lan .I can register from outside to my asterisk box but there is only one way audio , reason being that when the asterisk box send a sip packet whith session description the sdp part of the sip packet is not natted . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Ring on Multiple Phones
William, thanks for the info on macros. I'll try to implement some macros using several different callgroups. I have in mind: ALL, all upstairs, all downstairs, her normal domain and my normal domain. Normal domain for me is my upstairs office, ham radio 'shack' and lab and for her is her downstairs office, kitchen and family room. If wife and I are both on the same floor (rare), we can just shout g I looked up call queues at http://www.voip-info.org/wiki-Asterisk+call+queues and it seems pretty complex for the simple job I want to do. What I want to do is have all the phones ring some callgroup. Intercom and paging in selected callgroups and incoming calls probably in ALL. Normally, the only people who would answer would be either wife or me. In the case of intercom calls, it's me calling her or vice versa. In case of incoming calls, again either me or her would pick up. Does that sound like a situation that would be helped by call queues? Larry William Piper wrote: Sure, do something like this: [telasip-in] exten = _512879677[67],1,macro(callgroup,s,1) exten = _879677[67],1,macro(callgroup,s,1) [macro-callgroup] exten = s,1,Dial(SIP/120SIP/121SIP/122SIP/124SIP/125SIP/126SIP/127) exten = s,2,hangup From the sounds of it, this would probably work better if you setup call queues, but the above will do what you are asking. bp On 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote: William I found and fixed the problem. Your comment gave me the kick to persevere. Thank you very much. My exten line had a comment at the end that contained a close paren. That apparently screwed up the context line - although it shouldn't have. Now all three extensions ring. Note my mail program wrapped the line but it's not wrapped in the file: [telasip-in] ;=== exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) ; to be all extensions) exten =_512879677[67],1,Dial(SIP/120SIP/122SIP/124) This leads to another problem. I have 13 sip phones for [telasip-in] and other contexts to call ring groups for extension intercomming. Is there some kind of macro I could have to replace the instances of: (SIP/120SIP/122SIP/124) I have not yet written or read up on macros. Larry William Piper wrote: I don't know then, I do the same exact thing: exten = _352688,3,Dial,SIP/202SIP/214|20 Perhaps try sending everything in that context exactly as it is typed let us look at it. I'm pretty sure you have something configured incorrectly. Thanks, bp On 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote: Sorry I was not clear Rushowr. In the actual extensions.conf as used, the 'old' line is commented out so only 'new' is active. Then I reload. However, only the single 120 line rings instead of all. Larry Rushowr wrote: Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP NOTIFY
Hi,I have trixbox and Audiocodes MP-124 FXS. In Asterisk console I often get this message:Got SIP response 481 Call/Transaction Does Not Exist back from 86.38.10.233 So I have traced the sip packets, and I saw that Audiocodes MP-124 FXS sends this message ė81 Call/Transaction Does Not Exist, because Asterisk sends him NOTIFY. So my question is how to turn off this notify? Asterisk IP : 86.38.10.227MP-124 IP : 86.38.10.233#U 86.38.10.227:5060 - 86.38.10.233:5060NOTIFY sip:[EMAIL PROTECTED] SIP/2.0.Via: SIP/2.0/UDP 86.38.10.227:5060;branch=z9hG4bK1919bb63.From: Unknown sip:[EMAIL PROTECTED];tag=as029d6bdc.To: sip:[EMAIL PROTECTED].Contact: sip:[EMAIL PROTECTED].Call-ID: [EMAIL PROTECTED] .CSeq: 102 NOTIFY.User-Agent: Asterisk PBX.Max-Forwards: 70. Event: message-summary.Content-Type: application/simple-message-summary.Content-Length: 92..Messages-Waiting: no.Message-Account: sip:[EMAIL PROTECTED] .Voice-Message: 0/0 (0/0).#U 86.38.10.233:5060 - 86.38.10.227:5060SIP/2.0 481 Call/Transaction Does Not Exist. Via: SIP/2.0/UDP 86.38.10.227:5060;branch=z9hG4bK1919bb63.From: Unknown sip:[EMAIL PROTECTED];tag=as029d6bdc. To: sip:[EMAIL PROTECTED];tag=1c1266127160.Call-ID: [EMAIL PROTECTED] .CSeq: 102 NOTIFY.Contact: sip:86.38.10.233.Supported: em,timer,replaces,path.Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Content-Length: 0..Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Toll-Free numbers
Hi Everyone, Currently in my country, there is no toll free service provider. My company has been thinking of starting such a service (using Asterisk as a soft switch)but really we dont know how to go about this. Can anyone assist us with information/documentations, etc Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can anyone recommend a large button sip phone for the elderley.
Chuck Bunn wrote: Can anyone recommend a large button/type sip phone (VOIP) that an older person could use. I have a client that needs to have large button phones for elderly residents in her facility. You might want to look into the original Grandstream Phone, the BT-101. I havn't found any phone with bigger buttons. And each time the phone rings, it lights up a dark room. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GIZMO and Asterisk, Failed to authenticate
[Aug 31 04:32:22] NOTICE[20241]: chan_sip.c:5291 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #984) [Aug 31 04:32:23] NOTICE[20241]: chan_sip.c:9600 handle_response_register: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3) sip.conf: register = 1747mynumber:[EMAIL PROTECTED] ; Gizmoproject [proxy01.sipphone.com] type=friend context=default disallow=all allow=ulaw allow=alaw allow=ilbc dtmfmode=rfc2833 host=proxy01.sipphone.com insecure=very secret=mypassword username=1747mynumber canreinvite=yes What did I wrong? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wellgate 3804a: Got SIP response 486 Busy Here
I cannot explain why I get all the time: Got SIP response 486 Busy Here back from 192.168.250.244 I have a Wellgate 3804a there. How can I solve it? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER+iptables+Asterisk
Siqhamo Sifo wrote: I have ser sitting on my iptables nat box and my asterisk box on the lan . Ser does forwarding so that any requests (register,invite,ack,...) to the nat box at 5060 r sent to my asterisk box on the lan .I can register from outside to my asterisk box but there is only one way audio , reason being that when the asterisk box send a sip packet whith session description the sdp part of the sip packet is not natted . Use rtproxy for SER and an according ser.cfg (see SER example configs) -- Arnd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID and call progress pri
the configuration is this : NT PRI TD405P TE A -- B (Asterisk) A make a call to B. A can display the ID (caller ID , example John) of B ? these information are exchanged in the call progress ? B can change the called number and communicate this change to A whene the call is hangup ? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax vs. sip?
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros and cons. We support both and whilst we have more customers on SIP than IAX, currently favour IAX for new customers where they are undecided given lower support overhead and simplified load-balancing. I'd recommend you try both with the provider you're considering. Simonwww.esms.comOn 8/31/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller id problem
Hi, Does anyone can tell me how to set the caller id shown in the callee phone? When I use hard IP phone to make a PSTN call, the number displayed in PSTN phone correctly using set(callerid(num)). However, the caller id won't be displayed when I use software IP phone to PSTN. Does any method/function to control the caller display in all case (including call forwarding)? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail as email and attachment
Hello All, Am relatively new to Asterisk, but kinda slogging my ass off on it. My first couple of qs to begin with : 1) I tried the voicemail on no-answer thing. and my line in the voicemail.conf, duz have an email address and also attach=yes, 5600 = 5600, Benjamin Jacob, [EMAIL PROTECTED]|attach=yes I still havent really received a mail or the attachment. Don't I have to specify the mail server IP etc??I searched high and low for this. 2) For configuration changes, which is the best option to take up, use Asterisk Realtime, or Asterisk Manager APIs. Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with recording
Hi,I am trying to record a speech with this command:exten=205,3,Record(speech:wav).But it records aproximately about 10 seconds and asterisk hangs up. Does somebody know how to solve this problem, I also tried with max duration, but it didn't help.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: voicemail as email and attachment
asterisk uses the sendmail daemon. Make sure it is installed and working. -- -- Steven http://www.glimasoutheast.org Benjamin Jacob [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello All, Am relatively new to Asterisk, but kinda slogging my ass off on it. My first couple of qs to begin with : 1) I tried the voicemail on no-answer thing. and my line in the voicemail.conf, duz have an email address and also attach=yes, 5600 = 5600, Benjamin Jacob, [EMAIL PROTECTED]|attach=yes I still havent really received a mail or the attachment. Don't I have to specify the mail server IP etc??I searched high and low for this. 2) For configuration changes, which is the best option to take up, use Asterisk Realtime, or Asterisk Manager APIs. Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Err, wasn't the patch for H.264 just changing one digit for another? Hi Thomas, I don't know. I should check BUG page for that. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Other cool things: make menuconfig Jingle/jabber support IAX2 media transfers new sound files New answer machine detection (AMD) and much much more! Hi Matt, thank you for info! Bye. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: voicemail as email and attachment
On Thu, Aug 31, 2006 at 07:25:22AM -0400, Steven wrote: asterisk uses the sendmail daemon. A sendmail daemon. could be sendmail, postfix, exim, qmail, xmail, smail, or whatever. Or even a non-queueing non-daemon /usr/sbin/sendmail such as ssmtp and nullmailer . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cisco 7960G SIP firmware 8.4
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Seems to be working ok on my handset for the past couple of weeks. No major bugs, registration, xml services and MWI works etc..etc.. Have not given it a thorough testing though. Hi Nathan, Does it have any new options? I would like to see hinting on 7940/7960. Can you send me your's Phone Directory xml files? I can't manage to add second page so I have only 32 numbers :(( Also, I can't manage to enable search thru directory. Other thing, can personal directory be in xml file? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cisco 7960G SIP firmware 8.4
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any info? What version should I download? Is this one all right? cmterm-7940-7960-8.4.00-sip.cop.sgn Signed SIP Firmware for CCM versions 5.0(4) and later -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use a Half E1 with Asterisk?
Hello I want to know which hardware I have to use in order to use a half E1 with Asterisk (the second half will be used by a PABX PANASONIC). I have already a succesfull experience in Asterisk with an entire E1 (TE110P card) or 4 analogic channels (TDM400P) but I have no idea how physically connect to a half E1 Thank you for your help _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with asterisk?
Hi, I use here mgetty+sendfax with a modem to receive and send fax messages. Is it possible to receive and send a fax with asterisk directly? I have two passive ISDN card (HFC-S chipset, one in NT mode the other in TE-mode) and a old ELSA Microlink modem via serial on my computer. The OS is FreeBSD. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Matthias Fechner schrieb: ... I use here mgetty+sendfax with a modem to receive and send fax messages. Is it possible to receive and send a fax with asterisk directly? Hi, did google for asterisk and fax show no results? Strange! Ok, what you need is Steve Underwood's package spandsp and the two asterisk applications app_rxfax and app_txfax, which is not included in spandsp nor in asterisk, but is generally to be downloaded near to spandsp. Regards, Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Ring on Multiple Phones
Na, this will be fine for that... when you said 15 phones, I thought of a call center. Having queues gives you reporting tools. For what you are talking about though... the macro will be fine. bp On 8/31/06, Larry Alkoff [EMAIL PROTECTED] wrote: William, thanks for the info on macros.I'll try to implement somemacros using several different callgroups.I have in mind:ALL, all upstairs, all downstairs, her normal domain and my normal domain.Normal domain for me is my upstairs office, ham radio 'shack' and laband for her is her downstairs office, kitchen and family room.If wife and I are both on the same floor (rare), we can just shout gI looked up call queues athttp://www.voip-info.org/wiki-Asterisk+call+queues and it seems pretty complex for the simple job I want to do.What I want to do is have all the phones ring some callgroup.Intercom and paging in selected callgroups and incoming calls probablyin ALL. Normally, the only people who would answer would be either wife or me.In the case of intercom calls, it's me calling her or vice versa.In case of incoming calls, again either me or her would pick up.Does that sound like a situation that would be helped by call queues? LarryWilliam Piper wrote: Sure, do something like this: [telasip-in] exten = _512879677[67],1,macro(callgroup,s,1) exten = _879677[67],1,macro(callgroup,s,1) [macro-callgroup] exten = s,1,Dial(SIP/120SIP/121SIP/122SIP/124SIP/125SIP/126SIP/127) exten = s,2,hangup From the sounds of it, this would probably work better if you setup call queues, but the above will do what you are asking. bp On 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote: William I found and fixed the problem.Your comment gave me the kick to persevere.Thank you very much. My exten line had a comment at the end that contained a close paren. That apparently screwed up the context line - although it shouldn't have.Now all three extensions ring. Note my mail program wrapped the line but it's not wrapped in the file: [telasip-in] ;=== exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124); to be all extensions) exten =_512879677[67],1,Dial(SIP/120SIP/122SIP/124) This leads to another problem.I have 13 sip phones for [telasip-in] and other contexts to call ring groups for extension intercomming. Is there some kind of macro I could have to replace the instances of: (SIP/120SIP/122SIP/124) I have not yet written or read up on macros. Larry William Piper wrote: I don't know then, I do the same exact thing: exten = _352688,3,Dial,SIP/202SIP/214|20 Perhaps try sending everything in that context exactly as it is typed let us look at it. I'm pretty sure you have something configured incorrectly. Thanks, bpOn 8/30/06, Larry Alkoff [EMAIL PROTECTED] wrote: Sorry I was not clear Rushowr. In the actual extensions.conf as used, the 'old' line is commented out so only 'new' is active.Then I reload.However, only the single 120 line rings instead of all. Larry Rushowr wrote: Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled.What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp --Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP Proxy
Hi, Can I do RTP Proxy in asterisk? As our requirement says that voice packet should also go though sip server, so that billing should be perfect. Thanks, Ranjeet Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Proxy
canreinvite=no will force all rtp packets through *. Ranjeet Kumar wrote: Hi, Can I do RTP Proxy in asterisk? As our requirement says that voice packet should also go though sip server, so that billing should be perfect. Thanks, Ranjeet Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Network stuff you didn't know http://www.networkoblivion.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DTMF between cisco and sipura going throughasterisk
Figured it out, so here it is for archives sake: I set the dtmf mode to info instead of rfc2833 works with asterisk clients and sipura (Cisco gateway sends everything rtp-nte). Thanks to all who helped. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Lawetz Sent: August 29, 2006 1:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] DTMF between cisco and sipura going throughasterisk Hello all, we're having an issue with DTMFs being sent to Sipura's. Calls are originating from a Cisco AS5300 being sent to asterisk which in turn sends it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows the same problem with a cheap answering machine). The DTMFs sent from the AS5300 aren't recognised by the legacy PBX. - DTMFs are recognised correctly on the asterisk (when we check voicemail) - The cisco is setup with dtmf-relay rtp-nte - in sip.conf the cisco and sipura are set to rfc2833 If I set the cisco in dtmf-relay rtp-cisco it works on the sipura, but not on the asterisk. Unfortunately I can only set one dtmf-relay mode on the cisco. Is there anything I can change on asterisk or sipura to get the sipura to work with the rtp-nte (or to get asterisk to work with the cisco-rtp)? Any hints can help, Thanks Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Junk at beginning of frame
I am using format_mp3 to play mp3 files for musiconhold. I am getting warning's like: 2006-08-31_08:53:28 WARNING[4961]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443302 Is this something to worry about? FB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Barzilai escribi:
Franciso,
can you make a call to the outside world, from the FXS port and going
out the FXO port?
I mean, without Asterisk in between. (The SPA300 can be configured that
way)
I'm asking because I remember having trouble with the SPA recognizing
that the FXO line was "alive" when I plugged in a Panasonic PBX line.
But when I connected it directly to the phone company it recognized the
voltage or whatever.
If I remember correctly, the "503" error message is exactly what I was
getting. But this was almost 2 years ago.
BarZ
Francisco Seratti wrote:
Dave Fullerton escribi:
Francisco Seratti wrote:
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
Im using this extension to match cell calls and sendthem to the spa:
exten = _15.,1,Dial(SIP/300/${EXTEN}) ; cellphones are 15
and 300 is the spa3000 extension, registered OK
exten = _15.,2,Hangup
In the dialplan section of the sipura, i ve tried many different
options like xx.:@gw0, (xx.) and many others.
I cannot find a formal configuration doc for this device, so if you
giveme a hand to configure it or tellme where to start, or where
is the problem i would be very pleased.
This may be a stupid question, but are you sure you are using the
registration for the FXO port (PSTN Line) and not the FXS port (line 1)
? I usually don't give my trunk lines extension numbers is why I ask.
-Dave
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Dave, thanks for your time. Yes, im sure it was the FXO port that i
regitered in the Asterisk. Do you know in which cases a 503 "Service
Unavailable" is obtained?
I also configured the syslog for this line and im getting just before
the 503 response, the line: "151AUD:Stop PSTN Tone". I dont
know what is this, but maybe a clue. If you need an extra data or
config,, ask to me.
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I did not tried that configuration, how should i configure Line 1 to
make calls through PSTN Line ? I will also try with a telco line, maybe
i have to config the tone frequencies to adapt them to the gsm gw tone.
F
--
Francisco Seratti
Sunesys Telecomunicaciones
Bouchard 644. 5to A. Puerto Madero
[EMAIL PROTECTED]
Tel: (54) 011- 4311-9009 (Rotativas)
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Re: [asterisk-users] upgrade problem on IP phone 9133i
it is fixed !!! i tried again this morning and it worked the first time ! it will remain a mystery 2006/8/31, shadowym [EMAIL PROTECTED]: I don't remember all the details. I think you have to set the IP of the PC with the TFTP client as the tftp server on the phone. I seem to recall something about the name of the file as well. Again, it's quite foggy as I did this about a year ago so sorry I can't be of more help. I remember getting those packet fails too until I set up all the parts correctly. The documentation for the phone should have all the info. If I could figure it out I'm sure you can. From: Jean-Louis curty [mailto:[EMAIL PROTECTED]] Sent: Wednesday, August 30, 2006 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] upgrade problem on IP phone 9133i good idea! I tried but it doesn't work either...[08/30/06 23:12:33] UDP packet receive failed[08/30/06 23:12:33] Invalid opcode (0) during transfer received[08/30/06 23:12:34] Sending ' firmware.st' to '192.168.0.101'[08/30/06 23:12:34] UDP packet receive failed[08/30/06 23:12:34] Invalid opcode (12800) during transfer received [08/30/06 23:12:43] UDP packet receive failed [08/30/06 23:12:43] Invalid opcode (12800) during transfer received[08/30/06 23:12:44] UDP packet receive failed[08/30/06 23:12:44] Invalid opcode (12800) during transfer received [08/30/06 23:12:53] UDP packet receive failed [08/30/06 23:12:53] Invalid opcode (12800) during transfer received[08/30/06 23:12:54] UDP packet receive failed[08/30/06 23:12:54] Invalid opcode (12800) during transfer received 2006/8/30, shadowym [EMAIL PROTECTED]: I think this is a known problem that was fixed in v1.3.I think you need to do this upgrade using a 'put' install via tftp client rather than trying to configure it to 'get' from a tftp server.It's beenawhile so my memory is a bit foggy.I used pumpKIN. http://kin.klever.net/pumpkin/-Original Message-From: Jean-Louis curty [mailto:[EMAIL PROTECTED] ]Sent: Wednesday, August 30, 2006 9:07 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] upgrade problem on IP phone 9133ihi everybody,I bought few units for evaluation but we were not able to upgrade the firmware to 1.4 , it's currently set at 1.2, when we go to the webadminpage,whether we try to change the IP of the tftp server or the firmware name andset values, the reply is alwaysInvalid IP address Please try againif we change the value via the phone itself , the tftp ip is changed but thefirmware does not come up we are sure of our tftp server since it's used to upgrade other phones from other brands any idea ?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] editing configs thru web/ apps
Thanks for the sendmail tip guys. Now the 2nd q was the more urgent one and still is. How on earth do you edit cofigurations in Asterisk. (na.. am not talking thru your fav editor). Like say a web application wants to add an exten, or change the forwarding of some extension, etc. all this cannot be done manually. If I am repeating these questions, please point me to the posts where these qs have been answered. Goddamn urgent!! me almost sweating. :-) Thanks Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oddity with TDM400P / Asterisk setup
Ted Wallingford wrote: Hi List, I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything. The * version running is 1.2.7.1. All of the ports on the switch with voice devices, including the server, have a service class of 5, while non-voice devices are connected to other ports that have a service class of best effort. The problem, which began this morning, is very elusive. Calls-in-progress from zap-to-sip or sip-to-zap or sip-to-Asterisk will drop at odd times during the call, anywhere from 2 minutes to 15 minutes into the call. At the same time the call drops, my SSH session to the server will hang. After 10 to 15 seconds, the output and input from ssh session appears on my terminal and I am able to resume working in the shell. Zap-to-Asterisk doens't seem to cause the problem. Only when I dial through to a SIP device does it seem to hang. Top reveals nothing out the ordinary, utilization wise, the disk has plenty of free space, and the arp cache doesn't ever indicate a duplicate IP address with the server's NIC, which I thought might have been the problem. I also attempted to move the server to another port on the switch. No improvement. Anybody have a problem like this? Have not seen anything close to that problem. You might check the linksys switch to see if it has Spanning Tree turned on. Spanning Tree (depending on vendor code) will disable a port from forwarding traffic for about 10 to 15 seconds as a means of detecting layer two loops. If it is turned on, turn it off and test again. Also, you should be able to set up a series of pings from different sources to determine exactly which component in the infrastructure is failing. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 8.0.4 SIP firmware
Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got error when compiling asterisk 1.2.11
I got follwing error when tried to compile asterisk 1.2.11 on redhat linux 9: make[1]: Entering directory `/home/voipuser/asterisk-1.2.11/db1-ast'make[1]: `libdb1.a' is up to date.make[1]: Leaving directory `/home/voipuser/asterisk-1.2.11/db1-ast'make[1]: Entering directory `/home/voipuser/asterisk-1.2.11/stdtime'make[1]: `libtime.a' is up to date.make[1]: Leaving directory `/home/voipuser/asterisk-1.2.11/stdtime'for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x || exit 1 ; donemake[1]: Entering directory `/home/voipuser/asterisk-1.2.11/res'make[1]: Nothing to be done for `all'.make[1]: Leaving directory `/home/voipuser/asterisk-1.2.11/res'make[1]: Entering directory `/home/voipuser/asterisk-1.2.11/channels'gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o chan_zap.cchan_zap.c: In function `pri_dchannel':chan_zap.c:9025: structure has no member named `call'make[1]: *** [chan_zap.o] Error 1make[1]: Leaving directory `/home/voipuser/asterisk-1.2.11/channels'make: *** [subdirs] Error 1 How can I fix it? gc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP ProLiant and Digium 24xxp
Kevin P. Fleming wrote: Robert Roach wrote: I have a customer request to deploy an HP rack server (ProLiant DL series) as the base system for an Asterisk install. They also want to use the Digium 24xxp card. I have heard that the Digium card is oversized and does not fit in a normal size chassis. Does anyone know if it will fit in the ProLiant chassis, or have a recommendation on another HP box to use? This is incorrect. Nobody (including Digium) makes 'oversized' PCI cards, because there are no chassis in the world they would ever fit in. However, it is true that the TDM2400P is the maximum possible size of a PCI card, both full-length and full-height. In addition, it requires a standard hard-drive power connector (Molex) to supply 12V power if any FXS modules are used, which are often hard (or impossible) to find in a 1U or 2U rack-mount server. There are some available, though, and shortly Digium will have an external power solution available for the TDM400P and TDM2400P cards. The issue is not the TDM2400 is over sized, but rather some PC hardware vendors assuming no one ever uses full sized cards anymore. Lots of systems have crowded fixed drive bays and other stuff into their cases that preclude using full sized cards (from any source, not just the TDM2400). And as Kevin just mentioned, they also assumed there will never be a need for a Molex power connector in the pci bus area of the box. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] best BRI card ?
anyone got any views on what card I should get for a single isdn BRI line, and the pros / cons of the card ? Thanks. Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
Hello Guys,We have a problem in configuring Sangoma A104. We want the 2 ports to beconfigured as E1 and the 2 ports as T1.We already run wancfg and configure the 2 ports as T1 and the last 2 ports as t1. Below is the logs when we issue wanrouter restart.[EMAIL PROTECTED]:/tmp# wanrouter restartShutting down wanpipe1 interface: w1g1Shutting down device: wanpipe4Shutting down device: wanpipe3 Shutting down device: wanpipe2Shutting down device: wanpipe1No devices running, Unloading ModulesStarting WAN Router...Loading WAN drivers: wanpipe done.Starting up device: wanpipe1Starting up device: wanpipe2 wanconfig: WAN device wanpipe2 driver load failed !! : ioctl(wanpipe2,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errorsStarting up device: wanpipe3 wanconfig: WAN device wanpipe3 driver load failed !! : ioctl(wanpipe3,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errorsStarting up device: wanpipe4 wanconfig: WAN device wanpipe4 driver load failed !! : ioctl(wanpipe4,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errorsConfiguring interfaces: w1g1done.Configuring interfaces: w2g1 w2g1: unknown interface: No such devicedone. Configuring interfaces: w3g1 w3g1: unknown interface: No such devicedone.Configuring interfaces: w4g1 w4g1: unknown interface: No such devicedone.Any help will be appreciated.Thanks, Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best BRI card ?
Quoting Julian Lyndon-Smith [EMAIL PROTECTED]: anyone got any views on what card I should get for a single isdn BRI line, and the pros / cons of the card ? I'll add to the question - anyone found any that work with ISDN in Canada, and what provider did you get the lines from ? If you had problems what were they ? Thanks. Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? Now I have downgrade to 8.0.2 version and phone has registered fine. Does anybody know what is the problem with SIP 8.0.4 firmware and how to solve it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Hello Roger, * Roger Schreiter [EMAIL PROTECTED] [31-08-06 14:19]: did google for asterisk and fax show no results? yes I found spandsp but it will do everything in software. Is it not a good idea to use my modem for the fax stuff? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
We have a problem in configuring Sangoma A104. We want the 2 ports to be configured as E1 and the 2 ports as T1. If I'm not mistaken, you can't do that with the A104D, that's why they sold me 2 x A102 for the same price as a A104. Better check with Sangoma. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
Sangoma has excellent support, why dont you ask them? On 8/31/06, Angelito Manansala [EMAIL PROTECTED] wrote: Hello Guys, We have a problem in configuring Sangoma A104. We want the 2 ports to be configured as E1 and the 2 ports as T1. We already run wancfg and configure the 2 ports as T1 and the last 2 ports as t1. Below is the logs when we issue wanrouter restart. [EMAIL PROTECTED]:/tmp# wanrouter restart Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe4 Shutting down device: wanpipe3 Shutting down device: wanpipe2 Shutting down device: wanpipe1 No devices running, Unloading Modules Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 Starting up device: wanpipe2 wanconfig: WAN device wanpipe2 driver load failed !! : ioctl(wanpipe2,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe3 wanconfig: WAN device wanpipe3 driver load failed !! : ioctl(wanpipe3,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe4 wanconfig: WAN device wanpipe4 driver load failed !! : ioctl(wanpipe4,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Configuring interfaces: w1g1 done. Configuring interfaces: w2g1 w2g1: unknown interface: No such device done. Configuring interfaces: w3g1 w3g1: unknown interface: No such device done. Configuring interfaces: w4g1 w4g1: unknown interface: No such device done. Any help will be appreciated. Thanks, Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
Does the 8.0.3 image has the same flaws as 8.0.4? Wasn't even able to register with * at all since most configuration examples from voip-info.org wouldn't work... Do you have any example config for me to try with SIP image on 7970G? Only tried 8.0.3 on my 7970G and had to switch to SCCP image...which is now 8.0.4 cheers rick Aaron Daniel schrieb: I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax vs. sip?
We've been using iax with teliax.com for a couple of years, and it seems the quality of calls varies with time. Sometimes it is good and next time its not so good. There has been changes occurring to iax and the jitterbuffer stuff over the last two years, and I'm reasonably certain that some poor quality is related to differences between teliax.com's implementation (eg, s/w versions) and ours. I've not bother to try sip since our asterisk implementation is truly both a production box for our small office, and a test box for various version testing, etc. We used iax for more than a year and moved to sip about 6 months ago. The quality from termination providers seems much better now with sip. Tom At 09:38 PM 8/30/2006, you wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIP might be better to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
I emailed then last 2 hours ago. Just waiting for their reply.ThanksOn 8/31/06, Moises Silva [EMAIL PROTECTED] wrote:Sangoma has excellent support, why dont you ask them?On 8/31/06, Angelito Manansala [EMAIL PROTECTED] wrote: Hello Guys, We have a problem in configuring Sangoma A104. We want the 2 ports to be configured as E1 and the 2 ports as T1. We already run wancfg and configure the 2 ports as T1 and the last 2 ports as t1. Below is the logs when we issue wanrouter restart. [EMAIL PROTECTED]:/tmp# wanrouter restart Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe4 Shutting down device: wanpipe3 Shutting down device: wanpipe2 Shutting down device: wanpipe1 No devices running, Unloading Modules Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 Starting up device: wanpipe2 wanconfig: WAN device wanpipe2 driver load failed !! : ioctl(wanpipe2,ROUTER_SETUP) failed::22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe3 wanconfig: WAN device wanpipe3 driver load failed !!: ioctl(wanpipe3,ROUTER_SETUP) failed: :22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe4 wanconfig: WAN device wanpipe4 driver load failed !!: ioctl(wanpipe4,ROUTER_SETUP) failed::22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Configuring interfaces: w1g1 done. Configuring interfaces: w2g1 w2g1: unknown interface: No such device done.Configuring interfaces: w3g1 w3g1: unknown interface: No such device done. Configuring interfaces: w4g1 w4g1: unknown interface: No such device done. Any help will be appreciated. Thanks, Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lito Manansala www.voicefidelity.netMobile: +63.906.437.0459PSTN: +63.44.790.6292sip:[EMAIL PROTECTED]msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Matthias Fechner schrieb: ... yes I found spandsp but it will do everything in software. Is it not a good idea to use my modem for the fax stuff? Hi, ok, you want to use an external faxmodem? Something like that: outside (PSTN or anythin else) | V asterisk box | | (via analogue phone line) | V external faxmodem | | RS232 cable or similar | V PC with faxmodem support Yes, that's possible. You will need an ATA or in your asterisk box a card for the analogue phone line. Ok, what is your question now? Please ask more precisely! (Or is Yes, that's possible already the desired answer?) Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got error when compiling asterisk 1.2.11
You need to update your version of libpri to the latest as well. gc wrote: I got follwing error when tried to compile asterisk 1.2.11 on redhat linux 9: gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o chan_zap.c chan_zap.c: In function `pri_dchannel': chan_zap.c:9025: structure has no member named `call' make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/home/voipuser/asterisk-1.2.11/channels' make: *** [subdirs] Error 1 How can I fix it? gc -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parked call, park-dail context
Has anybody noticed that, if a call is parked; times out and returns to the employee parking the call, but that employee fails to answer the call for whatever reason, the caller gets hung up on? I got the following log entry: == Everyone is busy/congested at this time (1:1/0/0) Aug 31 10:38:00 WARNING[4006]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'park-dial' Looking at the configs, I see no area specified for this context. But, doing a show dialplan park-dial, I do see an entry. Guessing this is automatically created. To fix this, I created a park-dial context within the dialplan that points back to our operator. Should I have to create such a context, or do I have something incorrectly configured? 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
I had similar error messages when I configured an A101, using the latest stable drivers, and found that restarting LINUX seemed to solve the problem Seems wanrouter stop doesn't clean up after itself. do a shutdown -r now and see if it comes up properly John Novack Angelito Manansala wrote: I emailed then last 2 hours ago. Just waiting for their reply. Thanks On 8/31/06, *Moises Silva* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sangoma has excellent support, why dont you ask them? On 8/31/06, Angelito Manansala [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello Guys, We have a problem in configuring Sangoma A104. We want the 2 ports to be configured as E1 and the 2 ports as T1. We already run wancfg and configure the 2 ports as T1 and the last 2 ports as t1. Below is the logs when we issue wanrouter restart. [EMAIL PROTECTED]:/tmp# wanrouter restart Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe4 Shutting down device: wanpipe3 Shutting down device: wanpipe2 Shutting down device: wanpipe1 No devices running, Unloading Modules Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 Starting up device: wanpipe2 wanconfig: WAN device wanpipe2 driver load failed !! : ioctl(wanpipe2,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe3 wanconfig: WAN device wanpipe3 driver load failed !! : ioctl(wanpipe3,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe4 wanconfig: WAN device wanpipe4 driver load failed !! : ioctl(wanpipe4,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Configuring interfaces: w1g1 done. Configuring interfaces: w2g1 w2g1: unknown interface: No such device done. Configuring interfaces: w3g1 w3g1: unknown interface: No such device done. Configuring interfaces: w4g1 w4g1: unknown interface: No such device done. Any help will be appreciated. Thanks, Lito ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lito Manansala www.voicefidelity.net http://www.voicefidelity.net Mobile: +63.906.437.0459 PSTN: +63.44.790.6292 sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Quoting Roger Schreiter [EMAIL PROTECTED]: Matthias Fechner schrieb: ... yes I found spandsp but it will do everything in software. Is it not a good idea to use my modem for the fax stuff? I have the configuration below and its fine (usr usb modem plugged back into the asterisk machine with hylafax) but I intend to replace with a software only config now that they are mature enough. didn't exist yet when this setup was built. it eliminates a device, some cables, and frees up a channel bank port. I have still found though the odd fax just won't go through, so I have a real fax machine hooked up that is used for outbound anyway, and have people dial that extension if the automatic setup just refuses to work with the fax on their end. Hi, ok, you want to use an external faxmodem? Something like that: outside (PSTN or anythin else) | V asterisk box | | (via analogue phone line) | V external faxmodem | | RS232 cable or similar | V PC with faxmodem support Yes, that's possible. You will need an ATA or in your asterisk box a card for the analogue phone line. Ok, what is your question now? Please ask more precisely! (Or is Yes, that's possible already the desired answer?) Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 config questions
I was expecting a more elegant answer to the 9 to dial out problem with the Polycom 501. Sure I can change my dialplan, but that means I have to adapt my dialplan to the phone, while the opposite seems like the way to go. Thanks for the answer, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: August 30, 2006 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 config questions On Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the Call List button (on the left row of buttons), I get the call lists (as expected). When I press the Directory button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102? Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself? It will not. either add to your contact entries, or alternatively have your dial plan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missing Agent Function
Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT function since i need something to offer the agents a way to check if they are logged in or not. i was specting to use AGENT function for this. and i found out this: asterisk*CLI show function AGENT No function by that name registered. As i read here http://www.voip-info.org/wiki/view/Asterisk+functions . AGENT should be available for 1.2.x.x and i don't have it :( (chan_agent.so is loaded). Do i have to enable something else in order to use this function? or anyone else knows any other way to offer a way to check if an agent is logged in or not? (without using show agents, since it must be used phone-side and by agents). Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help me!!Problem on incoming calls
Hi,Please Help me!!!I've installed TrixBox and VISDN (snapshot 20060802) on a PC with anHFC-4s card. Outbound Calls work fine, and inbound calls from Cellphoneswork fine too.I have a problem with incoming calls beginning with 0 (national andinternational calls-I stay in Italy) Thanks in advance for answersAndrea De Nadai Here thereis my VISDN.conffile [general] [global]network_role = privatetones_option = yeslanguage = itoutbound_called_ton = unknownforce_outbound_cli =force_outbound_cli_ton = noclir_mode = unrestricted_defaultcli_rewiting = Nonational_prefix = 0international_prefix = 00network_specific_prefix = subscriber_prefix = abbreviated_prefix = overlap_sending = Yesoverlap_receiving = No;default--overlap_receiving = Noautorelease_dlc = 10call_bumping = No [visdn1.0]network_role = private;default--context = from-trunkcontext = from-internaltones_option = yesoutbound_called_ton = unknownforce_outbound_cli =force_outbound_cli_ton = noclip_default_name = Esternoclip_default_number = 400clip_numbers = clir_mode = unrestricted_defaultoverlap_sending = Yesoverlap_receiving = No;default--overlap_receiving = No [visdn1.1]network_role = private;default--context = from-trunkcontext = from-internaltones_option = yesoutbound_called_ton = unknownforce_outbound_cli =force_outbound_cli_ton = noclip_default_name = Esternoclip_default_number = 400clip_numbers = clir_mode = unrestricted_defaultoverlap_sending = Yesoverlap_receiving = No;default--overlap_receiving = No [visdn1.2]network_role = private;default--context = from-trunkcontext = from-internaltones_option = yesoutbound_called_ton = unknownforce_outbound_cli = force_outbound_cli_ton = noclip_default_name = Esternoclip_default_number = 400clip_numbers = clir_mode = unrestricted_defaultoverlap_sending = Yesoverlap_receiving = No;default--overlap_receiving = No [visdn1.3]network_role = private;default--context = from-trunkcontext = from-internaltones_option = yesoutbound_called_ton = unknownforce_outbound_cli =force_outbound_cli_ton = noclip_default_name = Esternoclip_default_number = 400clip_numbers = clir_mode = unrestricted_defaultoverlap_sending = Yesoverlap_receiving = No;default--overlap_receiving = No ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missing Agent Function
The Flash Operator Panel (http://www.asternic.org/) can be configured to change the color of a phone's icon to indicate whether that agent is logged in or not. I've found it to be very useful and the agents don't mind using that to check their status as well as the queue status (how many callers are in the queue, etc.). Delca wrote: Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT function since i need something to offer the agents a way to check if they are logged in or not. i was specting to use AGENT function for this. and i found out this: asterisk*CLI show function AGENT No function by that name registered. As i read here http://www.voip-info.org/wiki/view/Asterisk+functions . AGENT should be available for 1.2.x.x and i don't have it :( (chan_agent.so is loaded). Do i have to enable something else in order to use this function? or anyone else knows any other way to offer a way to check if an agent is logged in or not? (without using show agents, since it must be used phone-side and by agents). Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!
Hi all,I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers (extra charged numbers: like games, erotic lines...) in a remote country Then i just go to a click to call website and start an attack inserting this Special Phone Number like 1$/min. Even if i control call duration on asterisk, and also my recepcionist will notice this is a fraud call, i'll be charged for the 1st minute every time My Idea would be if someone has already worried with this, wouldn't be great to create a list on wiki or something where we can share this pattern Numbers?Is very hard to discover all the patterns for all the countries without sharing our knowledge... Any tips?-- Best regards,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rxfax and T.38
Hi all -- Perhaps I haven't been looking in the right place, but is there a T.38 capable version of app_rxfax? I got T.38 working in passthru mode in Asterisk (thanks Steve!) with a Sipura ATA and the PSTN switch, and so far so good. I got app_rxfax working with the ulaw codec (which works most of the time) but having it receive faxes with T.38 would be ideal. Can this be done already? --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
Matthias Fechner wrote: Hello Roger, * Roger Schreiter [EMAIL PROTECTED] [31-08-06 14:19]: did google for asterisk and fax show no results? yes I found spandsp but it will do everything in software. Is it not a good idea to use my modem for the fax stuff? Why would it not be a good idea to do things in software? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 config questions
Title: RE: [asterisk-users] Polycom 501 config questions Dumb question here: Why the need to dial 9 for an outside line? If your extensions are less than 7 digits long then you know anything "XXX." is an outside call Maybe this isn't true everywhere, just curious. -Jonathan From: [EMAIL PROTECTED] on behalf of MikeSent: Thu 8/31/2006 10:46 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Polycom 501 config questions I was expecting a more elegant answer to the "9 to dial out" problem withthe Polycom 501. Sure I can change my dialplan, but that means I have toadapt my dialplan to the phone, while the opposite seems like the way to go.Thanks for the answer,Mike-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Jerry JonesSent: August 30, 2006 6:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom 501 config questionsOn Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the "Call List" button (on the left row of buttons), I get the call lists (as expected). When I press the "Directory" button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102?Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself?It will not.either add to your contact entries, or alternatively have your dialplan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help me!!Problem on incoming calls
On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi, Please Help me!!! I've installed TrixBox and VISDN (snapshot 20060802) on a PC with an HFC-4s card. Outbound Calls work fine, and inbound calls from Cellphones work fine too. I have a problem with incoming calls beginning with 0 (national and international calls-I stay in Italy) Thanks in advance for answers Andrea De Nadai Wouldn't it be a better idea to ask on the vISDN mailing list and/or the Trixbox forum? You are after all using their software... Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Sending Data to a Web Page
How do I get Asterisk to send streaming data, such as incoming calls, call times, etc. to a web page? I have a web app that I'm trying to use as a call manager.Thanks,David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems compil 1.2.11
Hi when i want compile asterisk 1.2.11, i have this error : make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime' cd editline unset CFLAGS LIBS test -f config.h || CFLAGS=-O6 ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc -O6 ) works... no configure: error: installation or configuration problem: C compiler cannot create executables. make: *** [editline/libedit.a] Erreur 1 [EMAIL PROTECTED] asterisk-1.2.11]# what is the library that i don't have put on my server ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help me!!Problem on incoming calls
Hi Please Post you Asterisk CLi when incoming is arriving.On 8/31/06, Patrick [EMAIL PROTECTED] wrote:On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi, Please Help me!!! I've installed TrixBoxand VISDN (snapshot 20060802) on a PC with an HFC-4s card. Outbound Calls work fine, and inbound calls from Cellphones work fine too. I have a problem with incoming calls beginning with 0 (national and international calls-I stay in Italy) Thanks in advance for answers Andrea De NadaiWouldn't it be a better idea to ask on the vISDN mailing list and/or the Trixbox forum? You are after all using their software...Regards,Patrick___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 config questions
Jonathan k. Creasy wrote: Dumb question here: Why the need to dial 9 for an outside line? If your extensions are less than 7 digits long then you know anything XXX. is an outside call We did it, because most of the users expected it. No other reason. 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How is GXP2000 with latest firmware
I noticed there is newer firmware for the GXP2000 so I updated (v1.1.0.16). Release notes are dated June28. I was wondering how that phone is working now with this latest firmware. I had sort of written it off awhile ago as not good enough for production. Has anything changed? I doubt the sound quality is any better with firmware updates but does it at least work properly, not lock up etc.? I'm talking an office environment and not for in your kids room or anything like that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help me!!Problem on incoming calls
forgot to mention, it may help if you post your extensions.confAs you are using from-internal context for this calls,and you are using trixbox, have look in extensions_additional.conf and all extension_*.conf to find out your [from-internal] context. By the way I wouldn't use the from-internal context for incoming calls from PSTN line...On 8/31/06, Marco Mouta [EMAIL PROTECTED] wrote:Hi Please Post you Asterisk CLi when incoming is arriving. On 8/31/06, Patrick [EMAIL PROTECTED] wrote:On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi, Please Help me!!! I've installed TrixBoxand VISDN (snapshot 20060802) on a PC with an HFC-4s card. Outbound Calls work fine, and inbound calls from Cellphones work fine too. I have a problem with incoming calls beginning with 0 (national and international calls-I stay in Italy) Thanks in advance for answers Andrea De NadaiWouldn't it be a better idea to ask on the vISDN mailing list and/or the Trixbox forum? You are after all using their software...Regards,Patrick___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best BRI card ?
Hi Julian, I'm using beronet BRI cards which are good and have autoconfiguring sw for installation. (I tried junghanns bristuff and I had more problems to install but maybe it is been improved lately). The only little disadvantage with beronet driver is that you have to use different configuration file (misdn.conf) because channels are treated in groups of ports and not as channels like Junghanns does. Giorgio Incantalupo Julian Lyndon-Smith wrote: anyone got any views on what card I should get for a single isdn BRI line, and the pros / cons of the card ? Thanks. Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 config questions
if you really DO need to dial 9 to get out because of the lengths of your extension numbers (re: Jonathan's post) then Jerry was right -- you have to modify the directory of the phone to 955. Moj Mike wrote: I was expecting a more elegant answer to the 9 to dial out problem with the Polycom 501. Sure I can change my dialplan, but that means I have to adapt my dialplan to the phone, while the opposite seems like the way to go. Thanks for the answer, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: August 30, 2006 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 config questions On Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the Call List button (on the left row of buttons), I get the call lists (as expected). When I press the Directory button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102? Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself? It will not. either add to your contact entries, or alternatively have your dial plan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:500,44f6fab011901298614243! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Sending Data to a Web Page
Hi,As far as I know you must have a look on Asterisk Manager Interface, the HTTP way to communicate with asterisk and send and receive commands/call states etcHave a look on wiki for AMI, or Asterisk Manager Interface. On 8/31/06, David R. [EMAIL PROTECTED] wrote: How do I get Asterisk to send streaming data, such as incoming calls, call times, etc. to a web page? I have a web app that I'm trying to use as a call manager.Thanks,David ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 config questions
With regard to your question about adding a 9 to get the dial from the call list to work. We sis this in the dialplan by catching 10 digit numbers and adding the nine. However we have since moved away from needing the 9. I originally put it there to be consitent with our previous pbx. On 8/31/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote: Dumb question here: Why the need to dial 9 for an outside line? If your extensions are less than 7 digits long then you know anything XXX. is an outside call Maybe this isn't true everywhere, just curious. -Jonathan From: [EMAIL PROTECTED] on behalf of MikeSent: Thu 8/31/2006 10:46 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Polycom 501 config questions I was expecting a more elegant answer to the 9 to dial out problem withthe Polycom 501. Sure I can change my dialplan, but that means I have toadapt my dialplan to the phone, while the opposite seems like the way to go.Thanks for the answer,Mike-Original Message-From: [EMAIL PROTECTED][ mailto:[EMAIL PROTECTED]] On Behalf Of Jerry JonesSent: August 30, 2006 6:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom 501 config questionsOn Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the Call List button (on the left row of buttons), I get the call lists (as expected). When I press the Directory button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102?Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself?It will not.either add to your contact entries, or alternatively have your dialplan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!
Marco Mouta [EMAIL PROTECTED] wrote: Hi all, I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers (extra charged numbers: like games, erotic lines...) in a remote country Then i just go to a click to call website and start an attack inserting this Special Phone Number like 1$/min. Even if i control call duration on asterisk, and also my recepcionist will notice this is a fraud call, i'll be charged for the 1st minute every time Why not exclude international and 809 outbound calls entirely and then bless specific countries as needed? You could include your phone number on your site so that people from other countries could call your center as needed. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Francisco Seratti wrote: Hi pals, im trying to save some money in cellphones calls, so i bought a GSM gateway and a Sipura SPA3000 gateway. The GSM gw is currently working, and now im trying to configure the SPA, but every call i send, i get a 503 service unavailable. It does that if no line is plugged in. Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems using Queues with Autofill option
Hi, is anybody using autofill option in queues??This option is not in the asterisk distribution.Is described in http://bugs.digium.com/view.php?id=5577I have problems with it. Can sombody help me?Thanks, Esteban ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom HD Voice
Polycom is announcing a technology called HD Voice in a new IP650 phone, which is basically support for G.722.What is the current status of G.722 support within Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Per DID Codec Negotiation
Hi Everyone, From my research I believe I am asking the impossible but perhaps I am missing something. Any help would be greatly appreciated. I receive many DIDs from the same SIP provider coming from the same IP. I have a peer setup in sip.conf for this provider and this is where the codec negotiation happens. The problem comes in when I have to send a given SIP call to a device that only accepts g729. I have to accept ulaw from the provider so that fax will work and transcoding to g729 is not a good option because it takes up CPU resources and costs extra money. Since the provider supports g729, I would like to receive calls for some DIDs as g729 and the rest as ulaw. Is there any possible way to have this kind of configuration when all the DIDs come from the same IP and there is no SIP registration? As a side note, if I accepted the calls through IAX from the provider and sent them out as SIP, would this help? Thanks, Damien Gabrielson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF between cisco and sipura going through asterisk
On Tue, 29 Aug 2006, Benjamin Lawetz wrote: Hello all, we're having an issue with DTMFs being sent to Sipura's. Calls are originating from a Cisco AS5300 being sent to asterisk which in turn sends it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows the same problem with a cheap answering machine). The DTMFs sent from the AS5300 aren't recognised by the legacy PBX. - DTMFs are recognised correctly on the asterisk (when we check voicemail) - The cisco is setup with dtmf-relay rtp-nte - in sip.conf the cisco and sipura are set to rfc2833 If I set the cisco in dtmf-relay rtp-cisco it works on the sipura, but not on the asterisk. Unfortunately I can only set one dtmf-relay mode on the cisco. Is there anything I can change on asterisk or sipura to get the sipura to work with the rtp-nte (or to get asterisk to work with the cisco-rtp)? Any hints can help, Ben, What version of Aserisk are you using? If it is the 1.2 series, there are all sorts of RFC-2833 DTMF Relay issues that can crop up. My suggestion is that if you are willing to take the time, it might be worth it to Upgrade to the pre-release version of Asterisk that is currently in TRUNK. This supports the new Variable Length DTMF code that should knock out nearly all of the DTMF issues that Asterisk has had. The 1.2 and earlier RTP stack and RFC-2833 implementation, while not technically wrong according to the RFC, did things a bit differently than the rest of the world has chosen, and therefore can cause DTMF instability. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about 7940s and call forwarding
Hello, I need some advice on the following problem I'm trying to solve: At the office we are using 7940s as our phones, connected to an asterisk box via SIP. Pretty standard setup, nothing fancy. Everyone has an extension that comes out as a single line button on the phones, with the second line unused at this point. Certain things on our phone system are set to ring all the phones in the office (support queue fills up, etc.). I just simply do something like this: Dial(SIP/123SIP/456SIP/789) which works just fine, except if someone wants to use the phone's built-in call forwarding to send their calls to their house or cell phone. Then, any call that rings all the phones gets picked up and forwarded by that person's phone and they end up getting all the calls. Originally I had set up everyone as agents and let them forward their extensions anywhere, but the agent callback stuff makes things really unstable, plus it's cumbersome to use compared to using the phone. The solution I'm considering is to make the ring all calls ring to the second line button on all the phones and disabling call forwarding in sip.conf for those lines. I'm just not sure how a phone set to forward calls will react...will it properly not answer the calls on this line, or will it try to forward them and fail and cause the calls to get dropped? Will call transfers still work for calls that came in on the second line? Also if there's a better way of doing what I want to do I'm all ears. :) -- Joshua M Thompson [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!
Yeah,Could be a solution! Thanks for your reply.On 8/31/06, Henry J. Cobb [EMAIL PROTECTED] wrote: Marco Mouta [EMAIL PROTECTED] wrote: Hi all, I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers (extra charged numbers: like games, erotic lines...) in a remote country Then i just go to a click to call website and start an attack inserting this Special Phone Number like 1$/min. Even if i control call duration on asterisk, and also my recepcionist will notice this is a fraud call, i'll be charged for the 1st minute every timeWhy not exclude international and 809 outbound calls entirely and then bless specific countries as needed?You could include your phone number on your site so that people from othercountries could call your center as needed.-HJC___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agent autologoff
Hello, I found commands AddQueueMember and RemoveQueueMember so no need for agent id and password. You just dial the extension and your extension are in the game. Nice. ;Agent Login exten = 450,1,Noop exten = 450,n,AddQueueMember(q1) exten = 450,n,AddQueueMember(q2) exten = 450,n,Wait(1) exten = 450,n,Playback(agent-loginok) exten = 450,n,Wait(1) exten = 450,n,Hangup ;Agent Logout exten = 451,1,Noop exten = 451,n,RemoveQueueMember(q1) exten = 451,n,RemoveQueueMember(q2) exten = 451,n,Wait(1) exten = 451,n,Playback(agent-loggedoff) exten = 451,n,Wait(1) exten = 451,n,Hangup But now looks like queues doesn't have automatic logout feature only agents. Means agents added by AddQueueMember must be removed by RemoveQueueMember. Is there any solution for automatic logout or I must have to use AgentCallbackLogin? Is there any function for checking that I'm in a specified queue? bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Mark Willis escribi: Francisco Seratti wrote: Hi pals, im trying to save some money in cellphones calls, so i bought a GSM gateway and a Sipura SPA3000 gateway. The GSM gw is currently working, and now im trying to configure the SPA, but every call i send, i get a 503 service unavailable. It does that if no line is plugged in. Mark __ NOD32 1.1733 (20060831) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1733 (20060831) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com Im trying so many config's and i cant get ride of this. I realized that sometimes, when everything is connected (line to spa, phone to spa) and then i unplug and plug the power cable, the phone rings inmediatly, pick up, i have dialtone from the line and im able to make calls; but just when i hangup, the phone rings again and all the same.. that is very strange. After a few picks up and hangups it no rings anymore, and if i pickup there is no more dialtone but fast-busy :( (all this happens with factory default settings) I guess the spa "sometimes" detects the line, and some voltage or freq variety make the spa think it is an incoming call, so that's why it rings.. Please, correctme if im wrong, or tell me what you think about it. Thanks, Francisco. -- Francisco Seratti Sunesys Telecomunicaciones Bouchard 644. 5to A. Puerto Madero [EMAIL PROTECTED] Tel: (54) 011- 4311-9009 (Rotativas) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DTMF between cisco and sipura going throughasterisk
We're actually using a mix of 1.2.11 and 1.0.7 (in the process of upgrading). The problem occurs on both versions. But I seem to have found a solution by setting the dtmf mode to info (it's always the simple things ;-)) Thanks for the help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: August 31, 2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF between cisco and sipura going throughasterisk On Tue, 29 Aug 2006, Benjamin Lawetz wrote: Hello all, we're having an issue with DTMFs being sent to Sipura's. Calls are originating from a Cisco AS5300 being sent to asterisk which in turn sends it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows the same problem with a cheap answering machine). The DTMFs sent from the AS5300 aren't recognised by the legacy PBX. - DTMFs are recognised correctly on the asterisk (when we check voicemail) - The cisco is setup with dtmf-relay rtp-nte - in sip.conf the cisco and sipura are set to rfc2833 If I set the cisco in dtmf-relay rtp-cisco it works on the sipura, but not on the asterisk. Unfortunately I can only set one dtmf-relay mode on the cisco. Is there anything I can change on asterisk or sipura to get the sipura to work with the rtp-nte (or to get asterisk to work with the cisco-rtp)? Any hints can help, Ben, What version of Aserisk are you using? If it is the 1.2 series, there are all sorts of RFC-2833 DTMF Relay issues that can crop up. My suggestion is that if you are willing to take the time, it might be worth it to Upgrade to the pre-release version of Asterisk that is currently in TRUNK. This supports the new Variable Length DTMF code that should knock out nearly all of the DTMF issues that Asterisk has had. The 1.2 and earlier RTP stack and RFC-2833 implementation, while not technically wrong according to the RFC, did things a bit differently than the rest of the world has chosen, and therefore can cause DTMF instability. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Per DID Codec Negotiation
Damien Gabrielson wrote: Hi Everyone, From my research I believe I am asking the impossible but perhaps I am missing something. Any help would be greatly appreciated. I receive many DIDs from the same SIP provider coming from the same IP. I have a peer setup in sip.conf for this provider and this is where the codec negotiation happens. The problem comes in when I have to send a given SIP call to a device that only accepts g729. I have to accept ulaw from the provider so that fax will work and transcoding to g729 is not a good option because it takes up CPU resources and costs extra money. Since the provider supports g729, I would like to receive calls for some DIDs as g729 and the rest as ulaw. Is there any possible way to have this kind of configuration when all the DIDs come from the same IP and there is no SIP registration? As a side note, if I accepted the calls through IAX from the provider and sent them out as SIP, would this help? I could be wrong, but I believe this is being addressed soon. (not in 1.4 release). Assuming the SIP provider supports this, you would offer both codecs, and then the terminating device would only accept one type, then there will be a renegotiation so that all the traffic uses the same codec. ie. call comes in from provider as G.711u, G.711a, G.726, G.729a, iLBC Your asterisk box allows G.711u and G.729. The call is terminated to a SIP device that only supports G.729. Then the call gets renegotiated as G.729 all the way from provider to device. (And the same with G.711u). This way you make the fax ATA G.711u only and the SIP handsets G.729 only and allow both. This doesn't work yet, but is apprently in the pipeline. In the meantime, I'd suggest using 2 sets of credentials in sip.conf from the SIP provider and change the SIP address the provider terminates to accordingly. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom HD Voice
Yeh, Ive been surprised that there hasnt been more development in this space. Is there a bounty needed to get this happening? Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eldon Neustaeter Sent: Thursday, 31 August 2006 12:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom HD Voice Polycom is announcing a technology called HD Voice in a new IP650 phone, which is basically support for G.722. What is the current status of G.722 support within Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 and Asterisk
Mark Willis escribi: Francisco Seratti wrote: Hi pals, im trying to save some money in cellphones calls, so i bought a GSM gateway and a Sipura SPA3000 gateway. The GSM gw is currently working, and now im trying to configure the SPA, but every call i send, i get a 503 service unavailable. It does that if no line is plugged in. Mark __ NOD32 1.1733 (20060831) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1733 (20060831) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com Ive been reading Regional settings documentation, but i cant realize what to correct. These are some specs of the GSM gw: Gain: 5dB Impedance: 50 ohms Vertical Polarity Dialtone freq: 450 Hz +/- 15 Hz Ring freq: 25 Hz +/- 3 Hz Ring voltage: 60 V +/- 20 V Line voltage: 26 V +/- 2 V Well, thats all, i hope you can help me. Thanks in advance, Francisco. -- Francisco Seratti Sunesys Telecomunicaciones Bouchard 644. 5to A. Puerto Madero [EMAIL PROTECTED] Tel: (54) 011- 4311-9009 (Rotativas) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compil 1.2.11
Anyone have a idea ? Noc Phibee a écrit : Hi when i want compile asterisk 1.2.11, i have this error : make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime' cd editline unset CFLAGS LIBS test -f config.h || CFLAGS=-O6 ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc -O6 ) works... no configure: error: installation or configuration problem: C compiler cannot create executables. make: *** [editline/libedit.a] Erreur 1 [EMAIL PROTECTED] asterisk-1.2.11]# what is the library that i don't have put on my server ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 config questions
Title: RE: [asterisk-users] Polycom 501 config questions Pretty much like Doug said: because people expect it. Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. CreasySent: August 31, 2006 11:45 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom 501 config questions Dumb question here: Why the need to dial 9 for an outside line? If your extensions are less than 7 digits long then you know anything "XXX." is an outside call Maybe this isn't true everywhere, just curious. -Jonathan From: [EMAIL PROTECTED] on behalf of MikeSent: Thu 8/31/2006 10:46 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Polycom 501 config questions I was expecting a more elegant answer to the "9 to dial out" problem withthe Polycom 501. Sure I can change my dialplan, but that means I have toadapt my dialplan to the phone, while the opposite seems like the way to go.Thanks for the answer,Mike-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Jerry JonesSent: August 30, 2006 6:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom 501 config questionsOn Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the "Call List" button (on the left row of buttons), I get the call lists (as expected). When I press the "Directory" button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102?Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself?It will not.either add to your contact entries, or alternatively have your dialplan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 config questions
Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce ReevesSent: August 31, 2006 12:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom 501 config questions With regard to your question about adding a 9 to get the dial from the call list to work. We sis this in the dialplan by catching 10 digit numbers and adding the nine. However we have since moved away from needing the 9. I originally put it there to be consitent with our previous pbx. On 8/31/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote: Dumb question here: Why the need to dial 9 for an outside line? If your extensions are less than 7 digits long then you know anything "XXX." is an outside call Maybe this isn't true everywhere, just curious. -Jonathan From: [EMAIL PROTECTED] on behalf of MikeSent: Thu 8/31/2006 10:46 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Polycom 501 config questions I was expecting a more elegant answer to the "9 to dial out" problem withthe Polycom 501. Sure I can change my dialplan, but that means I have toadapt my dialplan to the phone, while the opposite seems like the way to go.Thanks for the answer,Mike-Original Message-From: [EMAIL PROTECTED][ mailto:[EMAIL PROTECTED]] On Behalf Of Jerry JonesSent: August 30, 2006 6:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom 501 config questionsOn Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the "Call List" button (on the left row of buttons), I get the call lists (as expected). When I press the "Directory" button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102?Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself?It will not.either add to your contact entries, or alternatively have your dialplan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id problem
You cannot set callerid on POTs lines. You my have more luck if you place your call via a T1 - but it's still up to your carrier. Some VoIP providrs also allow you to set callerid on SIP calls, but you need to check. I fear you'll have a hard time finding a carrier that will allow you to set CALLERID(name), they usually limit it to number only. H On 8/31/06, unplug [EMAIL PROTECTED] wrote: Hi, Does anyone can tell me how to set the caller id shown in the callee phone? When I use hard IP phone to make a PSTN call, the number displayed in PSTN phone correctly using set(callerid(num)). However, the caller id won't be displayed when I use software IP phone to PSTN. Does any method/function to control the caller display in all case (including call forwarding)? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: GSM gateway and FXO ATA
On 2006-08-26 18:35:27 -0700, Tzafrir Cohen [EMAIL PROTECTED] said: On Sat, Aug 26, 2006 at 02:02:51PM -0700, Martin Joseph wrote: On 2006-08-22 01:59:09 -0700, Tomislav ParÄina [EMAIL PROTECTED] said: Hi list! I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2. 5 over Grandstream HT488 ATA. snip Personally I found the FXO port on the HT-488 to unworkable except as a backup for power outages. I found several problems with it. 1) serious echo issues (I have a long loop). But the OP will have a very short loop. 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. The user here seems to be the GSM gateway. 3) The device hangs and reboots itself occasionally. Finally something relevant. Thanks so much for your critique! Usually when I buy hardware I like to know if it works, even if I plan to use it in a limited manor. I have found it's good to know how wel it works so that it can repurposed for other applications when/if the time comes. Although I see that your point is correct regarding some of my points not being applicable to this installation, I still feel it's relevant to the hardware he mentioned. Sorry to bother you, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compatibility INTEL E7520
Hello,I am buying a server for my Asterisk PBX (with 2 Digium TE110P):- IBM x346: (2005-2006) ChipSet: INTEL E7520Bus: 800 MHz.Proccessor Support: 2.8, 3.0, 3.2, 3.4 , 3.6 y 3.8 GHz, del tipoExteneded Memory 64 Technology (EM64T)I read the Digium Compatibility List and there isn't among incompatible chipset models.Does Anybody uses that chipset (or machine) with Digium cards successfully? Thanks in advanceDaniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Wellgate 3804a
On 2006-08-28 00:30:22 -0700, Martin Joseph [EMAIL PROTECTED] said: On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said: I want that each call from PSTN goes to Asterisk to the context for this line. Within this context can be a menu or a dial command, ... As more I read, as more I get confused, ... and each try is not working ! My sip.conf: [WG88621001] type=friend defaultip=192.168.250.244 insecure=very context=incoming_WG dtmfmode=rfc2833snip Also, RFC2833 doesn't work right on my Wellgate 3701a, I had to switch to using inband. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: GSM gateway and FXO ATA
On 2006-08-29 01:06:39 -0700, Tomislav Parčina [EMAIL PROTECTED] said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. Hi Marty! Can you tell me more about this? You mean when call from SIP goes to FXO port, if phone attached on FXO port answers after the first ring (before second) ATA will always stop to work? Actually it's kind of the opposite... When a call comes in to the FXO, and it rings the FXS, if the FXS answers on the first ring, the call goes somewhere but who knows where. The picking up party hears a dial tone, and the caller hears dead air. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missing Agent Function
Hi, Seems that FOP is a great tool and the person who made it is from my country :). But I'm having some problems configuring it. I made it possible to connect to the Asterisk as manager. Also I see a lot of output/input when I set debug=1. But, at the flash interface, the button that is under the arrow it's blinking... and as I can see in the official page demo, it isn't normal and I don't really know what could it be causing it. Cheers, Santiago On 8/31/06, Joe Dennick [EMAIL PROTECTED] wrote: The Flash Operator Panel (http://www.asternic.org/) can be configured to change the color of a phone's icon to indicate whether that agent is logged in or not. I've found it to be very useful and the agents don't mind using that to check their status as well as the queue status (how many callers are in the queue, etc.). Delca wrote: Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT function since i need something to offer the agents a way to check if they are logged in or not. i was specting to use AGENT function for this. and i found out this: asterisk*CLI show function AGENT No function by that name registered. As i read here http://www.voip-info.org/wiki/view/Asterisk+functions . AGENT should be available for 1.2.x.x and i don't have it :( (chan_agent.so is loaded). Do i have to enable something else in order to use this function? or anyone else knows any other way to offer a way to check if an agent is logged in or not? (without using show agents, since it must be used phone-side and by agents). Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] weird sound with IAX
Hello, I am having very weird sound on IAX protocol (using SIP, it seems to work OK). I use Asterisk 1.2.10. As a client, I use Idefisk. Today, i let two completely different asterisk machines talk to each other, with more or less same results. I currently do not use IAX trunking. This test was performed on 1Gbps ethernet with no packet loss with ulaw codec (no transcoding on the way): http://flz.sk.cx/audio/20060831-181241_59206988_to_.wav.mp3 Any help would be greatly appreciated. I looked at voip-info.org, but saw no troubleshooting info for IAX. Thanks, Juraj Bednar. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk presence (from manager API)
Hello, Did you try a combination of qualify=yes in sip.conf and then try the ExtensionState in the manager? yes, I have qualify=yes in the IAX config for peers I'm interested in. Seems like if qualify=yes or 2000... whatever, is not set then asterisk will not always know the state of the phone if it looses registration. That would seem to explain the problem you have with extensionstate. I can set qualify=2000, currently I have qualify=yes. Thank you, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Sending Data to a Web Page
I write an enhanced CDR (adding AGENT, ANI, DNIS, GLOBALID (unique across hosts), PRODUCT, PER-MINUTE, SURCHARGE, THEME, etc) at each major step in my dialplan to MySQL and then the web pages are created dynamically using PHP (to read from the database) and Smarty (to format for presentation). On Thu, 31 Aug 2006, Marco Mouta wrote: Hi, As far as I know you must have a look on Asterisk Manager Interface, the HTTP way to communicate with asterisk and send and receive commands/call states etc Have a look on wiki for AMI, or Asterisk Manager Interface. On 8/31/06, David R. [EMAIL PROTECTED] wrote: How do I get Asterisk to send streaming data, such as incoming calls, call times, etc. to a web page? I have a web app that I'm trying to use as a call manager. Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quadbri TDM400P on same pbx ?
Dear list, it is possible to have one quadbri (with only two ports connected) and one TDM400P card with only one FXO module connected, coexist on the same machine ? googling, and voip-infoing (tm) the answer seems no, anyway, maybe lately something has changed ? thanks so much for your support .mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
Aaron, was the MWI working for you on 8.0.2? I've got a 7970 and 7961 sitting on a shelf because the MWI doesn't work. On the 8.0.4, it never registered, but I was able to make calls with it. I didn't try calling it, since I never saw it register. It appeared it was authenticating for outgoing calls. On 8/31/06, Aaron Daniel [EMAIL PROTECTED] wrote: I tried that image for about 5 minutes.Kept getting errors in asteriskfrom the phone and it wouldn't stay registered.Rolled back to 8.0.2and that works fine for us for now.On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Aaron DanielComputer Systems TechnicianSam Houston State University [EMAIL PROTECTED](936) 294-4198___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 8.0.4 SIP firmware
MWI has been working on our (2) 7970's, as far as I can tell. My boss usually complains when his doesn't work, so it seems to be working fine as far as that's concerned. The 8.0.4 firmware attempted to register, but asterisk threw an error on a response it got back from the phone (I don't remember exactly which one), but I could make calls from it, just not to it. Aaron On Thu, 2006-08-31 at 14:33 -0500, Lacy Moore - Aspendora wrote: Aaron, was the MWI working for you on 8.0.2? I've got a 7970 and 7961 sitting on a shelf because the MWI doesn't work. On the 8.0.4, it never registered, but I was able to make calls with it. I didn't try calling it, since I never saw it register. It appeared it was authenticating for outgoing calls. On 8/31/06, Aaron Daniel [EMAIL PROTECTED] wrote: I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote: Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missing Agent Function
When the indicators are blinking it means that the op_panel.pl job isn't running on the server. There are some init scrips located in the init/ directory of the tar-ball for Flash Operator Panel. You can copy the appropriate script to your server's init directory (/etc/init.d on a RedHat system, for example) and then the Panel will start automatically at boot-up. Delca wrote: Hi, Seems that FOP is a great tool and the person who made it is from my country :). But I'm having some problems configuring it. I made it possible to connect to the Asterisk as manager. Also I see a lot of output/input when I set debug=1. But, at the flash interface, the button that is under the arrow it's blinking... and as I can see in the official page demo, it isn't normal and I don't really know what could it be causing it. Cheers, Santiago On 8/31/06, Joe Dennick [EMAIL PROTECTED] wrote: The Flash Operator Panel (http://www.asternic.org/) can be configured to change the color of a phone's icon to indicate whether that agent is logged in or not. I've found it to be very useful and the agents don't mind using that to check their status as well as the queue status (how many callers are in the queue, etc.). Delca wrote: Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT function since i need something to offer the agents a way to check if they are logged in or not. i was specting to use AGENT function for this. and i found out this: asterisk*CLI show function AGENT No function by that name registered. As i read here http://www.voip-info.org/wiki/view/Asterisk+functions . AGENT should be available for 1.2.x.x and i don't have it :( (chan_agent.so is loaded). Do i have to enable something else in order to use this function? or anyone else knows any other way to offer a way to check if an agent is logged in or not? (without using show agents, since it must be used phone-side and by agents). Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call to a queue killing Asterisk?
Avi Miller wrote: Hey guys, Last week I changed my queues from using proper agents and AgentCallbackLogin() to using the the FreePBX default with fixed agents (which uses the Local/[EMAIL PROTECTED] style for the member= field). I've also upgraded to Asterisk 1.2.10 and FreePBX 2.2.0 Beta 1. Since then, I noticed that my FOP would sometimes get stuck when a call hit the queue (showing all the agents being busy forever, until a op_server.pl reload). I started to track it this morning and actually saw Asterisk shutdown as the call got answered (and get restarted by safe_asterisk, of course). This accounts for the stuck FOP, but now I have the joy of working out why Asterisk is shutting down. I don't see anything in /var/log/asterisk/full -- I see the mysql CDR being recorded and then 4 seconds later, I see the Asterisk startup sequence happening. Anyone have any suggestions on where to start debugging this? Thanks, Avi Check http://bugs.digium.com/view.php?id=6626 cheers Terry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
