Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Wednesday, December 03, 2008 11:14 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call You’ll have to recheck your facts...MS does include a SIP client in WM6. And it works great ☺ Carriers/brands can remove items from ROM, but the SIP client is in by default. Have a look on XDA developers web site for details Jason, here’s what you need: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk/ OCG; Have you managed to get this working on the front speaker? Or still the back speaker only? Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
Now, I have : Client 1 -Asterisk1--Asterisk2 Client 2 I need that sip sign go to Asterisk2 But RTP go to Asterisk1 and no more. Where have I to insert canreinvite ? Thank you -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric ManxPower Wieling Envoyé : mercredi 3 décembre 2008 19:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] canreinvite=yes problem canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a reinvite feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
On 3 Dec 2008, at 17:38, BERGANZ François wrote: Someone have a solution for me ? De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] De la part de BERGANZ François Envoyé : mercredi 3 décembre 2008 18:24 À : asterisk-users@lists.digium.com Objet : [asterisk-users] canreinvite=yes problem Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outbridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you Yes. 1. POST ONCE 2. If no one replies within 20 mins, don't start chasing 3. If its that important pay for support 4. Read documentation ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday, Asterisk is 9 years old!
Hi, December 5th, 1999 was the initial release of Asterisk by Mark Spencer. We'll be celebrating this by gathering as usual at 12 Noon Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for the VoIP Users Conference. You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems. IRC is Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: [EMAIL PROTECTED] (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: [EMAIL PROTECTED] (thanks to IdeaSIP.com) or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record) We start about 15 minutes to the hour with an informal chat. Join us anytime, but especially, grab a virtual beer and join us Friday the 5th. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set monitor_filename
Hi
I have this in my queue extension and I see this in asterisk when I call to the
queue, but no file is created in the directory any ideas?
exten =
s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [EMAIL PROTECTED]:1] Set(SIP/0850001175-b7942770,
MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12)
Regards
/ralf
Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707458074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
[EMAIL PROTECTED]mailto:[EMAIL PROTECTED]
www.adlibris.comhttp://www.adlibris.com/
P Please consider the environment before printing this e-mail
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Re: [asterisk-users] asterisk ooh323 avaya (URGENT!!!)
You can check the notes/links on the following post, it has a link to an avaya-asterisk ip trunk setup. http://www.tek-tips.com/viewthread.cfm?qid=1431673 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: 03 December 2008 22:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk ooh323 avaya (URGENT!!!) gracias!!! (thanks) 2008/12/3 Guillermo V. Salas [EMAIL PROTECTED] - [EMAIL PROTECTED] escribió: I would go for chan_h323. Much more stable since 1.4 and the config more close to the other channel configs too. We used it on production for a long time and it worked well although a little heavy cpu-wise. To get started you need to install openh323 and pwlib from here http://sourceforge.net/project/showfiles.php?group_id=80674 and the ./configure and make menuselect will detect it and let you build it along with asterisk. Be careful with the paths when installing them though. And watch the output of the asterisk configure command for possible errors. Here is a small chan_h323.so install guide: http://www.ecualug.org/?q=2008/04/18/comos/asterisk_14_agregando_soporte_h323_chan_h323so_en_asterisk_14 Saludos, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.manta.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Is sourceforge OpenH323 active ?
Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Deadlock ? I hope i am wrong
I have thousands if this messages in the logs: Dec 4 10:53:43 NOTICE[26310]: app_queue.c:1980 wait_for_answer: No one is answering queue 'COMMERCIAL-WT' (2/0/0) Dec 4 10:53:43 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Dec 4 10:53:43 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Dec 4 10:53:44 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Dec 4 10:53:44 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Can someone tell me to what it is related ? asterisk 1.4 freepbx Thank you Grygoriy Dobrovolskyy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Is sourceforge OpenH323 active ?
As I recall, when openh323.org because obsolete people could download the PWLib OpenH323 libraries from http://www.voxgratia.org/ Openh323 has moved to become OPAL (supporting SIP, H323 and IAX) and can be downloaded from http://www.opalvoip.org H323Plus is also a continuation of OpenH323 supporting only H323. If you need to download OpenH323 and PWLib version suitable for Asterisk's chan_h323 you can follow the OpenH323 downloads link at the Voxgratia site. I hope this helps. Best regards, Vlasis Hatzistavrou. Kinetix Tele.com International Inc. 306 Victoria House, Victoria, Mahe, Seychelles Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: [EMAIL PROTECTED] http://www.kinetixtele.com Postal address: Monastiriou 9 Enotikon 54627 Thessaloniki Greece Olivier wrote: Hi, A glance at sourceforge.net/projects/openh323 http://sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing the callerid of a mobile
Does anyone know of any UK mobile provider who can either provide a single number for a range of sims, or allow us to change the callerid of a sim dynamically ? We are looking at between 20-30 sims, perhaps more next year. Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :(
Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for the call • National is 020 will result in 20 being sent and dialled, which works • Mobile is 07x will result in 7x being sent and dialled, which works • International 00x[any number of digits] will result in 00x[any number of digits] which does not work I do not see why this does not work. I do know that for every call, the flag sent is national - how can I make sure the correct flag is sent for the call? By flag, I mean the TON, (type of number) Any assistance will be greatly appreciated. Thank you Mr Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Design
On Wed, Dec 3, 2008 at 10:47 PM, JD [EMAIL PROTECTED] wrote: Grey man: you are right. The direction of a call leg is easy to determine from the point of view of asterisk. I suspect other folks however, think of it differently. Some would think of a call coming from a customer CPE to asterisk as an outbound call. Inbound to asterisk, yes. But outbound for billing purposes. But if you are using Asterisk for xyz, then the pattern of logic may differ yet again. So, while AMA CDRs may look at it in a specific way, Steve's flexible system should probably leave it up to each programmer writing their unique billing analysis software. I knew I shouldn't have mentioned inbound and outbound calls :-). I don't have any issues distinguishing between outbound and inbound calls in the current system all I was attempting to do was point out it's not very difficult to manage. However it's not that relevant an argument for the CDR design so should be disregarded. To sum up, personally I still think the current AMA-style CDRs should remain in place. Perhaps touched up a bit. Or not. Steve's system should be in addition to the AMA-style CDRs and called something other than CDRs. (CDRplus?) Well if you are in a situation where you have to bill users and those users are able to make transfers you'd be just as keen for the system to get the system changes as a lot of the rest of us. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :(
On Thu, Dec 04, 2008 at 11:49:50AM +, Mr Gabriel wrote: Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for the call • National is 020 will result in 20 being sent and dialled, which works • Mobile is 07x will result in 7x being sent and dialled, which works • International 00x[any number of digits] will result in 00x[any number of digits] which does not work I do not see why this does not work. I do know that for every call, the flag sent is national - how can I make sure the correct flag is sent for the call? By flag, I mean the TON, (type of number) Any assistance will be greatly appreciated. Look into pridialplan in zapata.conf / chan_dahdi.conf . I'm not sure if 'pridialplan = unknown' is applicable. If not: something of the sort of internationalprefixx should help. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :(
- Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, 4 December, 2008 12:01:54 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :( On Thu, Dec 04, 2008 at 11:49:50AM +, Mr Gabriel wrote: Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for the call • National is 020 will result in 20 being sent and dialled, which works • Mobile is 07x will result in 7x being sent and dialled, which works • International 00x[any number of digits] will result in 00x[any number of digits] which does not work I do not see why this does not work. I do know that for every call, the flag sent is national - how can I make sure the correct flag is sent for the call? By flag, I mean the TON, (type of number) Any assistance will be greatly appreciated. Look into pridialplan in zapata.conf / chan_dahdi.conf . I'm not sure if 'pridialplan = unknown' is applicable. If not: something of the sort of internationalprefixx should help. Gabriel Says The pridialplan is set to pridialplan=unknown, and internationalprefix=00, I have rebooted a few times, so I know this is what is currently loaded. I am using freepbx as a web interface, is it possible that there are conflicting settings? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
- On 12/4/08, Uros Djokic [EMAIL PROTECTED] wrote: Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but they think they are cpe too ERROR: got ! frame in state 8 and soon after something like no Dchannel using 16 anyway.Is it normal ? Don't I need to see reconfiguring or reset channel 1 to 30 ? My zaptel.conf and zapata.conf are probably ok because i copied them from another system where everything works fine. I installed asterisk through book on my Ubuntu 8.04 server system.Only thing I didn't install before libpri and zaptel was libtermcap because there is no such package on Ubuntu. Thanks, Uros I had the same issue with a Qwest circuit. Rather than trying to work with Quest and burn up time on my cell, I just made my machine the the net (pri_net) and everything worked perfectly. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call
I had front speaker working initially - but have lost that (now back only). Something isn't quite right - but still workable... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: December 4, 2008 3:10 AM To: Asterisk Users List Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call Importance: High From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Wednesday, December 03, 2008 11:14 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call You’ll have to recheck your facts...MS does include a SIP client in WM6. And it works great ☺ Carriers/brands can remove items from ROM, but the SIP client is in by default. Have a look on XDA developers web site for details Jason, here’s what you need: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk/ OCG; Have you managed to get this working on the front speaker? Or still the back speaker only? Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possible to get Courtesy Tone on attended transfer?
Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when picking up a parked call, but I haven't found any information re: transferred calls. My users are finding that they don't know when to begin talking to the person on the other end of the call. Similarly, but less important, is there any way to push the original caller ID over to the extension that receives the transferred call? Thanks in advance, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
I still have: Client 1 -Asterisk1--Asterisk2 Client 2 When client1 do a call, asterisk1 forward to asterisk2, asterisk2 forward to Asterisk1 At this moment, asterisk1 say : 404Not found But I have insecure=very This is the sip debug at that moment: - --- (11 headers 0 lines) --- --- SIP read from UDP://192.168.1.151:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;rport Max-Forwards: 70 From: 103 sip:[EMAIL PROTECTED];tag=as636875d3 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Date: Thu, 04 Dec 2008 14:55:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1545198644 1545198644 IN IP4 192.168.1.151 s=Asterisk PBX 1.6.0.1 c=IN IP4 192.168.1.151 t=0 0 m=audio 12272 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (14 headers 13 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.1.151 : 5060 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user '103' in SIP users list Found peer 'media' for '103' from 192.168.1.151:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.151:12272 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.151:12272 Looking for 33170725012 in media (domain 192.168.1.153) --- Reliably Transmitting (no NAT) to 192.168.1.151:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;received=192.168.1.151;rport=5060 From: 103 sip:[EMAIL PROTECTED];tag=as636875d3 To: sip:[EMAIL PROTECTED];tag=as242de969 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Have you an idea why ? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de BERGANZ François Envoyé : jeudi 4 décembre 2008 09:15 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] canreinvite=yes problem Now, I have : Client 1 -Asterisk1--Asterisk2 Client 2 I need that sip sign go to Asterisk2 But RTP go to Asterisk1 and no more. Where have I to insert canreinvite ? Thank you -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric ManxPower Wieling Envoyé : mercredi 3 décembre 2008 19:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] canreinvite=yes problem canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a reinvite feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] cepstral vs festival
On December 2, 2008 07:55:00 pm Erik (Caneris) wrote: Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a That's pretty cool! Is there any SIP or IAX access to this (aside from dialing a POTS number) ? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN PRI settings for Telus BC network
Hi there! Does anyone deal with Telus in BC ? We have some PRI lines that were used for dialup, would like to convert them for pbx system, talked with some technicians @ Telus, but the information given was not clear, kind of: try this see if it works Does anyone here have the settings required to talk to there equipment ? Thanks for your help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy strangeness...
Doesn't look like anyone has any suggestions though, guess it's time to play until it's fixed then :) 2008/12/2 Thomas Kenyon [EMAIL PROTECTED] Geraint Lee wrote: Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on at the time. This is happening on mixmonitor recordings even if you're not listening in on chanspy too. Any suggestions? I don't have any suggestions, but this is similar to something I am experiencing with Chanspy in 1.4.21.1. If I spy on a call, then progressively throughout the call a delay is introduced. By the end of the call I can be listening to sound that is 10 seconds out of sync. (Then I don't get to hear the end of the call when the call is finished). This also leaves stale channels open. (the entry in show channels doesn't go away until the asterisk process is restarted). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.0.1, IMAP Voicemail storage and temporary greetings.
On Wednesday 03 December 2008 19:22:09 Barry L. Kline wrote: It appears as though * is looking for the temporary greeting on the local box, which is what I'd expect because of my configuration option. It also appears that * isn't deleting the file(s) when I ask it to. It also isn't taking them out of IMAP, which is compounded by the fact that I don't think that based on my configuration there should be an email anyway. You might want to try 1.6.0.3-rc1, released yesterday. In the fixes were included something similar to this. In case 1.6.0.3-rc1 does not fix this issue, please open a report on http://bugs.digium.com. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :(
In article [EMAIL PROTECTED], Mr Gabriel [EMAIL PROTECTED] wrote: Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for the call - National is 020 will result in 20 being sent and dialled, which works - Mobile is 07x will result in 7x being sent and dialled, which works - International 00x[any number of digits] will result in 00x[any number of digits] which does not work I do not see why this does not work. I do know that for every call, the flag sent is national - how can I make sure the correct flag is sent for the call? By flag, I mean the TON, (type of number) Not sure from your description of the numbers whether you are referring to the dialling rules in FreePBX or to what happens between the dialplan and the PRI. What I am about to say is applicable to the latter, ignoring whatever FreePBX does. I believe there are two possibilities: 1. Use pridialplan=unknown, leave nationalprefix and internationalprefix both empty, and give the full number to Dial(), as you would dial it, i.e. 020, 07x, 00x[whatever] The numbers will be sent over the PRI as TON unknown, without stripping any digits. 2. Use pridialplan=dynamic, nationalprefix=0, internationalprefix=00, and still give the full number to Dial(). In this case, the dynamic setting will match a leading 00, strip it and set TON=International, or else match a leading 0, strip it and set TON=National, or else it will set TON=Local and not strip any digits. In both cases, make sure your FreePBX dialling rules are set NOT to strip the 0 or 00 prefixes. Actually, I think if pridialplan is NOT dynamic, the national and international prefixes are ignored for dialling, but can still be usefully set to 0 and 00 to get incoming CallerId into the correct format. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.0.1, IMAP Voicemail storage and temporary greetings.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tilghman Lesher wrote: You might want to try 1.6.0.3-rc1, released yesterday. In the fixes were included something similar to this. In case 1.6.0.3-rc1 does not fix this issue, please open a report on http://bugs.digium.com. I don't have the email in production yet (I'm still using the legacy box), so I'll await the 1.6.0.3 release. If that fails to fix the problem I'll report the bug and perhaps see if I can track it down myself. My C skills have atropheed over the last couple of decades but... Thanks Tilghman. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJOAnWCFu3bIiwtTARAq1EAJ4zfzT+uP8WLA5ZiJFucMUYm1yJwwCeI8qZ EWAegiPxbLTUCCEavSstl+0= =iiC/ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 407 Proxy Authentication Required
Hello, I'm receiving some traffic from a Softwitch to Asterisk When I'm hiding the CallerID in the softwitch, everything is all right. When I allow to send the callerid from softwitch to Asterisk (actually, I would like to have it) Asterisk rejects the call with a 407 Proxy Authentication SIP packet. I copy-paste the SIP Invitation: -- Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK-27003710ff15ff5fff41 Transport: UDP Sent-by Address: 2.2.2.2 Sent-by port: 5060 Branch: z9hG4bK-27003710ff15ff5fff41 From: sip:[EMAIL PROTECTED];user=phone;tag=27003710ff15ff5fff41 SIP from address: sip:[EMAIL PROTECTED] SIP tag: 27003710ff15ff5fff41 To: sip:[EMAIL PROTECTED]:5060;user=phone SIP to address: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Contact: sip:[EMAIL PROTECTED];user=phone Contact Binding: sip:[EMAIL PROTECTED];user=phone URI: sip:[EMAIL PROTECTED];user=phone SIP contact address: sip:[EMAIL PROTECTED] Max-Forwards: 10 User-Agent: x Cisco-Guid: 406000640-2566207248-2147483669-3311398977 Content-Type: application/sdp Content-Length: 164 -- sip.conf section: [2.2.2.2] host=2.2.2.2 type=friend insecure=yes context=test canreinvite=no (and calls goes to test context) Which header is forcing Asterisk to ask for authentication, and if I hide the callerid it's not asking it? Thanks, -- Carles Pina i EstanyGPG id: 0x17756391 http://pinux.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
There is a loopback somewhere on the line. Contact your telco and say I see a loopback on the line. Please remove it. Uros Djokic wrote: Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but they think they are cpe too ERROR: got ! frame in state 8 and soon after something like no Dchannel using 16 anyway.Is it normal ? Don't I need to see reconfiguring or reset channel 1 to 30 ? My zaptel.conf and zapata.conf are probably ok because i copied them from another system where everything works fine. I installed asterisk through book on my Ubuntu 8.04 server system.Only thing I didn't install before libpri and zaptel was libtermcap because there is no such package on Ubuntu. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
Reinvites will happen by default. Post your sip.conf [general] and the peers in sip.conf masking only the passwords. Also paste the part of extensions.conf that you use to Dial. BERGANZ François wrote: Now, I have : Client 1 -Asterisk1--Asterisk2 Client 2 I need that sip sign go to Asterisk2 But RTP go to Asterisk1 and no more. Where have I to insert canreinvite ? Thank you -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric ManxPower Wieling Envoyé : mercredi 3 décembre 2008 19:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] canreinvite=yes problem canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a reinvite feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
In article [EMAIL PROTECTED], Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: There is a loopback somewhere on the line. Contact your telco and say I see a loopback on the line. Please remove it. I don't think this is correct. The OP below said that he put the loopback on himself, as a test. Uros Djokic wrote: Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but they think they are cpe too ERROR: got ! frame in state 8 and soon after something like no Dchannel using 16 anyway.Is it normal ? No, you can't loopback a port to itself with Asterisk running, because it will always appear to be talking to the same kind of device as itself, whether it is NET or CPE. What is more likely is that you need a crossover cable between your Digium card and the modem. Make up a crossover cable according to the instructions at http://wiki.sangoma.com/Cablepinouts Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
I would try setting zaptel to pri_net before calling the telco. If it works, you just saved yourself (possibly) hours of being bounced around from person to person and sitting on hold, not to mention being on hold or transfered and getting dropped and having to start all over again. Path of least resitance Thanks, Steve Totaro On 12/4/08, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: There is a loopback somewhere on the line. Contact your telco and say I see a loopback on the line. Please remove it. Uros Djokic wrote: Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but they think they are cpe too ERROR: got ! frame in state 8 and soon after something like no Dchannel using 16 anyway.Is it normal ? Don't I need to see reconfiguring or reset channel 1 to 30 ? My zaptel.conf and zapata.conf are probably ok because i copied them from another system where everything works fine. I installed asterisk through book on my Ubuntu 8.04 server system.Only thing I didn't install before libpri and zaptel was libtermcap because there is no such package on Ubuntu. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
Some telco switches just behave this way, either by design or misconfiguration. It is much easier to reconfigure your switch ;-) than the telco's. Not sure what switch Quest had me on, it may have been a DMS100 but I don't recall. Anyways, pri_net worked and has been working for over two years now. Thanks, Steve On 12/4/08, Steve Totaro [EMAIL PROTECTED] wrote: I would try setting zaptel to pri_net before calling the telco. If it works, you just saved yourself (possibly) hours of being bounced around from person to person and sitting on hold, not to mention being on hold or transfered and getting dropped and having to start all over again. Path of least resitance Thanks, Steve Totaro On 12/4/08, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: There is a loopback somewhere on the line. Contact your telco and say I see a loopback on the line. Please remove it. Uros Djokic wrote: Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but they think they are cpe too ERROR: got ! frame in state 8 and soon after something like no Dchannel using 16 anyway.Is it normal ? Don't I need to see reconfiguring or reset channel 1 to 30 ? My zaptel.conf and zapata.conf are probably ok because i copied them from another system where everything works fine. I installed asterisk through book on my Ubuntu 8.04 server system.Only thing I didn't install before libpri and zaptel was libtermcap because there is no such package on Ubuntu. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
On Thu, 4 Dec 2008, Steve Totaro wrote: I would try setting zaptel to pri_net before calling the telco. If it works, you just saved yourself (possibly) hours of being bounced around from person to person and sitting on hold, not to mention being on hold or transfered and getting dropped and having to start all over again. Path of least resitance Unfortunately, some helpful telco tech may notice the misconfiguration and fix it at the worst possible time :( Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
On 12/4/08, Steve Edwards [EMAIL PROTECTED] wrote: On Thu, 4 Dec 2008, Steve Totaro wrote: I would try setting zaptel to pri_net before calling the telco. If it works, you just saved yourself (possibly) hours of being bounced around from person to person and sitting on hold, not to mention being on hold or transfered and getting dropped and having to start all over again. Path of least resitance Unfortunately, some helpful telco tech may notice the misconfiguration and fix it at the worst possible time :( Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Point taken, but while true, unless you are the only customer on that switch, that tech may take down quite a few customers in the process, requiring the customer to reconfigure their CPE systems I seriously doubt they will touch a misconfiguration of such magnitude on a production switch. That is why I included the fact that in my situation, it had been two plus years with no trouble. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] We think we are cpe but they think they are cpe too
Next time I'll be sure to finish my morning coffee before posting. 8-) Tony Mountifield wrote: In article [EMAIL PROTECTED], Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: There is a loopback somewhere on the line. Contact your telco and say I see a loopback on the line. Please remove it. I don't think this is correct. The OP below said that he put the loopback on himself, as a test. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disable database
Hi, How do I disable asterisk to use database and storage voicemail in directory? Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2399 message_exists: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2429 delete_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2600 store_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2305 retrieve_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:3474 messagecount: Failed to obtain database object for 'asterisk'! Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival (MRCP)
John: However, that doesn't mean that it shouldn't be implemented. This is an area in which I think there is a disproportionate amount of non- discussion, since many people who would use or be interested in MRCP simply don't participate in the Asterisk project because it doesn't meet their needs out of the gate. Therefore, we see few people asking for it, in a self-fulfilling loop. Is MRCP something that is significantly lacking in Asterisk? Is it a difficult protocol to implement? Is there anyone here on -dev with the experience to do it? I don't know whether it's significantly lacking nor how difficult it is to implement, but it's certainly nice to have. It would increase the appeal of Asterisk to those used to working with MRCP-compatible resources in other platforms. That said, it can be argued that it's best to keep Asterisk simple and free of extra features. If its core purpose does not consist of interfacing with ASR and TTS engines, then some would argue that it's best to keep such features to a separate platform. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disable database
I think its the entries in the extconfig.conf file Mos 2008/12/4 Geraldo Coelho [EMAIL PROTECTED] Hi, How do I disable asterisk to use database and storage voicemail in directory? Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2399 message_exists: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2429 delete_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2600 store_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2305 retrieve_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:3474 messagecount: Failed to obtain database object for 'asterisk'! Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one of the TNs. However, I'll bring it up with the client and see if they'd want us to configure that. Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith (lists) [EMAIL PROTECTED] Sent: Thursday, December 04, 2008 10:43 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cepstral vs festival On December 2, 2008 07:55:00 pm Erik (Caneris) wrote: Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a That's pretty cool! Is there any SIP or IAX access to this (aside from dialing a POTS number) ? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
On December 4, 2008 02:14:52 pm Erik (Caneris) wrote: Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one of the TNs. However, I'll bring it up with the client and see if they'd want us to configure that. Definitely would be cool, you don't lose any ad revenue and I don't have to use up my minutes. Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. You have Kitchener/Waterloo! Yay dials Oh. No traffic. Boo-urns. I'd definitely like to know when you start populating the traffic part of K/W (and separate out london, it's a poor choice to group. Kitchener/Wwaterloo/Cambridge sure... but London? That's a common Torontonian thing to do. :-) -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic loading changed in asterisk 1.4
Thanks Kevin, After looking at the skeleton application, I inserted the line *AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY,My Prepaid Application);* To the end of the application. I also changed all references to pbx_exec(chan,app,data) to include three parameters instead of four (I guess the APIs have changed, the last parameter was an int). I no longer get the original message about failure to load, however, I do get a new message; *asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_prepaid.so: undefined symbol: mysql_init * Just so you know, I have res_config_mysql.so loaded already and I thought this would provide this symbol. I have played around with my Makefile in case it was a linking issue but cant seem to get it right. I enabled the NOISY_BUILD var in the Makefile in the asterisk root directory, so the actual gcc build command is displayed, which is: *gcc -o app_prepaid.o -c app_prepaid.c -pthread -I/opt/bristuff-0.4.0-RC3c/asterisk-1.4.21.2/include -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -include /opt/bristuff-0.4.0-RC3c/asterisk-1.4.21.2/include/asterisk/autoconfig.h -march=i686 -O6 -fPIC -DAST_MODULE=\app_prepaid\ -MD -MT app_prepaid.o -MF .app_prepaid.o.d -MP* Any ideas? Thanks, Mos 2008/12/3 Kevin P. Fleming [EMAIL PROTECTED] Mosiuoa Tsietsi wrote: I have browsed through a couple of posts that deal with the failure of applications that originally worked on asterisk 1.2 but fail on asterisk 1.4, but can't seem to understand what I need to change in my installation. I also went through the CHANGES.txt file in my asterisk source directory, and still was none the wiser. The simplest thing for you to do is to review the changes the sample application (app_skel) that we ship with Asterisk; the changes made in app_skel.c between 1.2 and 1.4 are the changes that need to be made to all loadable modules to work with the 1.4 module loader. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: disable database
Not work!!! I am receiving message in my email, but I can't heard from phone because the asterisk isn't salve message in directory. thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Mosiuoa Tsietsi Enviada em: quinta-feira, 4 de dezembro de 2008 17:09 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] disable database I think its the entries in the extconfig.conf file Mos 2008/12/4 Geraldo Coelho [EMAIL PROTECTED] Hi, How do I disable asterisk to use database and storage voicemail in directory? Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2399 message_exists: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2429 delete_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2600 store_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2305 retrieve_file: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:3474 messagecount: Failed to obtain database object for 'asterisk'! Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Realtime
Someone could make it work??? I tried everything and there's no way I can make it work! Someone can help me? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disable database
On Thursday 04 December 2008 12:46:53 Geraldo Coelho wrote: How do I disable asterisk to use database and storage voicemail in directory? Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! Recompile app_voicemail, turning off ODBC_STORAGE in the Voicemail options. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RES: disable database
So your sendmail is working fine. Asterisk should plug the voicemail in the
/var/spool/asterisk/voicemail/${yourcontext} directory. I would check the
permissions to make sure asterisk has rights to write here.
Mos
2008/12/4 Geraldo Coelho [EMAIL PROTECTED]
Not work!!!
I am receiving message in my email, but I can't heard from phone because
the asterisk isn't salve message in directory.
thanks
*De:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *Em nome de *Mosiuoa Tsietsi
*Enviada em:* quinta-feira, 4 de dezembro de 2008 17:09
*Para:* Asterisk Users Mailing List - Non-Commercial Discussion
*Assunto:* Re: [asterisk-users] disable database
I think its the entries in the extconfig.conf file
Mos
2008/12/4 Geraldo Coelho [EMAIL PROTECTED]
Hi,
How do I disable asterisk to use database and storage voicemail in
directory?
Im getting the below error
[Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to
obtain database object for 'asterisk'!
[Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index:
Failed to obtain database object for 'asterisk'!
[Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to
obtain database object for 'asterisk'!
[Dec 3 19:09:04] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to
obtain database object for 'asterisk'!
[Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2399 message_exists:
Failed to obtain database object for 'asterisk'!
[Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2429 delete_file: Failed
to obtain database object for 'asterisk'!
[Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2600 store_file: Failed to
obtain database object for 'asterisk'!
[Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:2305 retrieve_file: Failed
to obtain database object for 'asterisk'!
[Dec 3 19:09:06] WARNING[5129]: app_voicemail.c:3474 messagecount: Failed
to obtain database object for 'asterisk'!
Thanks in advance
___
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[asterisk-users] Packet size limit for HDLC?
Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with the analogue modems. Then, everything went fine (for a while). Unfortunately, PPP via ISDN is typically using a MTU and a MRU of 1500, and I found, that some commercial ISDN routers do not allow negotiating MTU and MRU. They insist to use a size of 1500. Since, using CAPI or ISDN4Linux (not via asterisk), pppd is working well with the MTU/MRU value of 1500, I assume, there is some packet size limitation in the asterisk part (including app_pppd). I tried to find any too small buffer or similar, but successless. May I ask you, where do you think, the limitation does come from: - from app_pppd (I don't think so) - from libpri - from chan_dahdi - from the dahdi kernel modules - from the asterisk kernel Any hint is welcome! Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: disable database
Thanks Mos and Tilghman, I recompiled it and now is working fine -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Tilghman Lesher Enviada em: quinta-feira, 4 de dezembro de 2008 18:52 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] disable database On Thursday 04 December 2008 12:46:53 Geraldo Coelho wrote: How do I disable asterisk to use database and storage voicemail in directory? Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! Recompile app_voicemail, turning off ODBC_STORAGE in the Voicemail options. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
Welcome to the world of FreePBX. It would save me quite a bit of time if you could list what ports (port number and signaling) you have on the card and what context you want each port to go into. When I manually merge the two files (after stripping out 37 billion comment lines) I see that channel 1 is defined twice, channel 4 is defined once and and channels 2 and 3 are not defined at all. John covici wrote: Sorry about that -- I think I have things in two places -- they do things in a different order than one would expect. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet size limit for HDLC?
ICMP is used to determine maximim packet size. If you or the other end are blocking all ICMP then MTU Path Discovery will not work. It's a classic newbie network admin mistake. Symptoms of this problem would be exactly like you describe. Typically I see this on PPPoE connections. More info: http://www.znep.com/~marcs/mtu/ Roger Schreiter wrote: Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with the analogue modems. Then, everything went fine (for a while). Unfortunately, PPP via ISDN is typically using a MTU and a MRU of 1500, and I found, that some commercial ISDN routers do not allow negotiating MTU and MRU. They insist to use a size of 1500. Since, using CAPI or ISDN4Linux (not via asterisk), pppd is working well with the MTU/MRU value of 1500, I assume, there is some packet size limitation in the asterisk part (including app_pppd). I tried to find any too small buffer or similar, but successless. May I ask you, where do you think, the limitation does come from: - from app_pppd (I don't think so) - from libpri - from chan_dahdi - from the dahdi kernel modules - from the asterisk kernel Any hint is welcome! Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet size limit for HDLC?
Eric \ManxPower\ Wieling schrieb: ICMP is used to determine maximim packet size. If you or the other end are blocking all ICMP then MTU Path Discovery will not work. It's a Hi, the problem is, the other side (ISDN-router) does not negotiate the MTU while setting up PPP. I can see this in the log file: Our side is proposing 296, but the other answers with NACK and tells 296. I think, my side is doing something according RFC, when proposing a smaller MTU than usual, but this does not solve my problem, because: More info: http://www.znep.com/~marcs/mtu/ A MTU of 1500 is typical for PPP over HDLC, and when my solution does not do, what is typical, it is not compatible enough. Now I want bring asterisk and app_pppd also to work with a MTU of 1500 (like native linux ippp also does). I want to understand, why PPP via asterisk is failing, when MTU is 1500. Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom no menu
Was messing with a polycom 501 and changed the IP from dhcp to static. Working with a user remotely. Now, the user says the phone does not show anything on the LCD and does not respond to any buttons. When rebooting, there is text shown as it proceeds. ?? Is there a way to reset this to a default? Does not respond to ping on the address we set. joe a. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Web front end for Meetme?
Is there another web front end for meetme apart from Web-MeetMe? Since it keeps crashing I need a stable solution for a customer. Any recommendations? Even a commercial app would be acceptable as long as it is stable and uses Asterisk. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Low RX volume and half duplex/walkie-talkie on AEX-804E
On 21/11/2008 6:47 a.m., Lincoln King-Cliby wrote: Hi All, I have a ticket open with Digium, but based on their previous lack of support for the Asterisk Appliance, I'm not really holding my breath - and, honestly, I'm not 100% convinced it's a Digium issue in the first place (but I don't know where else to point fingers). We have an AEX-804E (PCI Express, 4 FXO ports, Hardware Echo Cancellation) in a Dell PowerEdge 1950 with four straight analog telephone lines, and running asterisk 1.4.22. All of the local phones are Cisco 7961G with the SIP firmware. Calls between SIP sets, across our SIP trunk on a VPN to a remote office, or calls to or from the remote office's PSTN lines (over the aforementioned SIP trunk) are all fine. On many [but not all] calls to or from the PSTN, I'm getting two complaints - #1 is low receive (i.e. from the PSTN) volume #2 (which seems to get significantly worse if I try tweaking bumping up the tx/rx gain in Zapata.conf) is that if the person in our office is talking all inbound audio is muted, but not the other way around (i.e. half duplex, but not half duplex both directions if that makes any sense) Further compromising my sanity is that #1 seems hard for me to duplicate - calls to or from my cell phone, for example, always sound fine. Local calls are mostly fine, and long distance calls are hit-or-miss, calls to a Hawaiian (how's that for Long Distance from Ohio) 1004 Hz test number are fine - in fact, subjectively, borderline too loud which makes no sense since before going live with Asterisk, we had a legacy Panasonic KSU/PBX on the same lines - on the same punchdown blocks - and no one ever complained about these issues. If I turn off the echo canceller there's a modest (may even just be psychological) improvement in line gain, but the echo is so horrendous (actually the echo sounds louder than the inbound call volume) as to make things unusable. Any ideas? At all? I'm still relatively new to the Asterisk-interconnected-to-PSTN side of things, and it seems like there are dozens of config files and tools so explicit instructions are appreciated! Try adding these to the modprobe line: vpmnlptype=4 vpmnlpmaxsupp=11 -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom no menu
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, December 04, 2008 7:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] polycom no menu Was messing with a polycom 501 and changed the IP from dhcp to static. Working with a user remotely. Now, the user says the phone does not show anything on the LCD and does not respond to any buttons. When rebooting, there is text shown as it proceeds. ?? Is there a way to reset this to a default? Does not respond to ping on the address we set. joe a. You could try lobotomizing it by pressing and holding 468* You'll need to enter the password (456 if you haven't changed it) and then hit the leftmost softkey. Obviously, this will all be blind since it doesn't display anything. But it's worth a shot. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rate My Dialplan Contest Announced - Win a Phone or Copies of APSTel Visual Dialplan Std or Pro!
We often find ourselves reading through all sorts of contests on the Internet that never seem to echo our own personal skill set or interests. Perhaps you've even fantasized about a type of contest with the types of prizes and goodies that YOU'D actually enjoy. Maybe you've wished there were something along the lines of a asterisk phone system diagram contest? With prizes ranging from APSTel visual dial plan software licenses to an Aastra 57I IP telephone? Does that sound like you? It does!? Well it's your lucky day!! Flewid Inc, Ottawa Phone Systems, APSTel and VOIP Phreak have all been kind and generous enough to sponsor just such a contest! Not only does it include some prizes that you're sure to salivate over, but it also offers you a way to showcase your elite and skillful phone system diagram designing skills! All you'll need to do is submit your best, most organized and beautifully crafted IVR diagram for an Asterisk or similar VOIP PBX phone system. The top three overall diagrams with the most overall votes will be selected as the winners and will be awarded the following prizes: 1st place: An APSTel dial plan (professional license) donated by -- you guessed it - APSTel! 2nd place: An Aastra 57I IP telephone donated by Ottawa Phone Systems and Flewid Inc! 3rd place: An APSTel dial plan (standard license) donated by APSTel! The contest is going to be running from December 5th 2008 through February 5th 2009. You can submit your phone system IVR diagram entries either at the Rate My Dial Plan website (http://www.ratemydialplan.com) or by e-mail at [EMAIL PROTECTED] Again, the prizes will be awarded to the IVR diagrams with the most votes by February 5th 2009, just in time for you to enjoy your wonderful prize in the New Year! So get planning, get thinking and get working, because the competition will be bountiful and high in quality! We would expect nothing less from you all anyway and know we can expect some incredibly and inventive diagrams! Make sure you get your submission in before the deadline on FEBRUARY 5TH, 2009! We'd hate to see a quality diagram get disqualified because you procrastinated and didn't submit it to us in time! Make sure you think out of the box and with the fact that the plans with the most votes will be declared the victors. While you could definitely play it safe and close to the vest and still get votes, it's highly likely the winner will be one that presents us ideas on things that haven't been done before or even a better way to accomplish existing methods. While we'd like to offer everyone a chance to participate in this contest, you do have to be at LEAST 13 years of age and able to utilize the prizes effectively. What this means is in consideration of the fact that the prizes are powered capable for North American markets only. So take that into account if you're in a different country and if you have any questions, please don't hesitate to contact us to address them. Now go get designing! We wish you all the best of luck and look forward to seeing all of the great designs! Submit your phone system IVR diagram entries either at the Rate My Dial Plan website (http://www.ratemydialplan.com) or by e-mail at [EMAIL PROTECTED] Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] remote phones, no audio to PSTN
Odd problem, where some remote phones, at users homes, dial and connect fine, no matter what the destination is. Bad phones, SIP to SIP, between remote and office, or remote to remote, work and have good audio, but no audio, at all, to PSTN or Cell phones.Phone can be moved to office and work fine. I'm perplexed, at this hour. OK, ok, at most hours. joe a. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set monitor_filename
Ralf Träskman wrote:
Hi
I have this in my queue extension and I see this in asterisk when I call
to the queue, but no file is created in the directory any ideas?
exten =
s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [EMAIL PROTECTED]:1] Set(SIP/0850001175-b7942770,
MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12)
Regards
/ralf
The basics. Does the queuecalls directory exist? Does the user that *
runs under have write permission in that directory?
The monitor-format parameter MUST be set in queues.conf to enable
recording and to select the format of the recording. In addition, I
believe this is still true in 1.4, if you don't set the monitor-join
parameter to yes you will end up with two files (in out) instead of a
single file with both legs of the call.
Alex
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Re: [asterisk-users] Friday, Asterisk is 9 years old!
randulo wrote: Hi, December 5th, 1999 was the initial release of Asterisk by Mark Spencer. We'll be celebrating this by gathering as usual at 12 Noon Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for the VoIP Users Conference. You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems. IRC is Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: [EMAIL PROTECTED] (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: [EMAIL PROTECTED] (thanks to IdeaSIP.com) or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record) We start about 15 minutes to the hour with an informal chat. Join us anytime, but especially, grab a virtual beer and join us Friday the 5th. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users December 5th, 1879 is also the date when the first automatic telephone switch was patented. A good day for telecom all-round. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Is sourceforge OpenH323 active ?
2008/12/4 Vlasis Hatzistavrou (KTI) [EMAIL PROTECTED] As I recall, when openh323.org because obsolete people could download the PWLib OpenH323 libraries from http://www.voxgratia.org/ Openh323 has moved to become OPAL (supporting SIP, H323 and IAX) and can be downloaded from http://www.opalvoip.org H323Plus is also a continuation of OpenH323 supporting only H323. If you need to download OpenH323 and PWLib version suitable for Asterisk's chan_h323 you can follow the OpenH323 downloads link at the Voxgratia site. I hope this helps. Sure it does ! Thank you very much : opalvoip.org seems much more active and up to date ! If my trials are successfull, I'll update voip-info.org accordingly. Cheers Best regards, Vlasis Hatzistavrou. Kinetix Tele.com International Inc. 306 Victoria House, Victoria, Mahe, Seychelles Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: [EMAIL PROTECTED] http://www.kinetixtele.com Postal address: Monastiriou 9 Enotikon 54627 Thessaloniki Greece Olivier wrote: Hi, A glance at sourceforge.net/projects/openh323 http://sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
