Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help
Hi, I switched to Wengo and solved the one beatproblem. However, I am still not able to listen to the recorded .wav sound. Can anyone please point me to the right direction? How to listen to the .wav sound? Thanks, Pete On Fri, Mar 21, 2008 at 9:34 AM, Carlos Rojas [EMAIL PROTECTED] wrote: Hello, Do your verify, the codecs, of both clients, in your sip.conf? What codec do you use? Best Regards On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a beat sounds, and then nothing else. In the console, I can see: *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2001-081dd6e0, ) in new stack -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2001-081dd6e0, 2000) in new stack Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts 000160, len 000160) -- SIP/2001-081dd6e0 Playing 'vm-intro' (language 'en') Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts 000320, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts 000480, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037721, ts 000640, len 000160) Got RTP packet from192.168.1.102:8000 (type 00, seq 06, ts 1373137124, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037722, ts 000800, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037723, ts 000960, len 000160) Is it the prolem? First it sends to the public address of the the router, then it sends to the virtual IP. Is this the problem that causing my to hear just one beat sound and then no audio? Problem 2: The problem is isolated from Problem 1, cuz I run the SIP client on the same machine as the server, so there should not be network problem. I recorded some voice mails and they are stored as .wav files ok. When I tried to hear back the message, It does not work. Is there any configuration that I have to go through to have Asterisk to play .wav file? Thank you very much in advance for all your kind help. Pete ___ -- 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
Re: [asterisk-users] Hardphone SIP phone costs
On Wed, 19 Mar 2008, Norman Franke wrote: On Mar 19, 2008, at 2:48 PM, [EMAIL PROTECTED] wrote: My mobile does not sound terrible, does not have echo, does not fade in or out, and the last time I used it to call the emergency services, I got through straight away. I've not had a dropped call for a long time either (going through tunnels on the train, or over Dartmoor excepted) I've never heard a cell phone on the other end that I couldn't tell was a cell phone, even on a good day. Over here it's GSM. nothing more nothing less. Yes, it's noticable, but it's not terrible and it is consistent. I'm not aware of the networks imposing more compression on top of what the handset itself does. They compress the audio so much it's rather obvious. That may vary by carrier, ATT and Verizon being the largest in the US are both pretty awful. I'm getting the impression that the telcos in the US are basically shafting you because of the monopoly they have. More intersted in keeping themselves happy than their customers. I think it's nice I have a choice of 5 major mobile phone carriers in the UK, and well over 100 ISPs for broadband via the BT Wholesale network. A fun test is to call a landline from your cell in the same room and note now long the delay is. I find it long enough to interfere with conversations, people talking over each other (especially when both are on cells from different carriers.) There is a delay - but I've never really noticed it unless I play tricks on the network like that. It's certianly nothing like making a call to Austrailia! None of the carriers really offer a phone that can do SIP, as far as I've seen. As soon as the iPhone software 2.0 is out, there will be one for that. Don't rely on the carriers to provide you anything - there are plenty of phones on the market which do SIP now - most modern Nokias do. I use an E90 Communicator, but the E95 is popular too, so I'm experimenting with using my mobile as my one phone, via Wi-Fi/SIP when I'm in the home/office and GSM/3G when out and about. It's not perfect yet, but getting there. (And 10:1 gives you a SIP service on the iPhone that's locked into their own service ;-) Cheers, Gordon ___ -- 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] DRUID/Voiceroute on VoIP Users COnference today Friday 2008-03-21 at 11:58 AM EDT
FRIDAY March 21 2008 at 12 Noon Eastern Daylight Time ( 4PM UTC ) See http://VoipUsersConference.org for SIP or PSTN dial in information This week, Voiceroute.net will be joining our conference to talk about DRUID. People from Voiceroute on the call: Vikram Rangnekar, COO Voiceroute and Druid Community Honcho Navin Kumar, CTO Voiceroute Ming Yong, CEO Voiceroute They propose to 1) Introduce Voiceroute to the Asterisk IP Telephony Community. What we do? Why? 2) Announce that we have a open source Unified Communications project called Druid. What is Druid OSE (Open Source Edition) and the licensing for it? 3) Vision for Druid OSE project: It aims to be the de factor Open Source Unified Communications Software for the enterprise IP communications space. Why? 4) Discussion on the core features of the Druid Open Source Edition (OSE) 5) Discussion of the community aspect of the Druid OSE http://voipusersconference.org/topics.php for more on DRUID /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] Which command line is used to send emails to notify incoming voicemail ?
Hi, In exim4, I can see lines such as : mainlog.9:2008-03-12 08:53:28 1JZLmC-E7-0A = [EMAIL PROTECTED] U=root P=local S=43802 [EMAIL PROTECTED] In my voicemail.conf, I see : ; If you need to have an external program, i.e. /usr/bin/myapp called when a ;externnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp called when a ;externpass=/usr/bin/myapp So, I guess this line (from app_voicemail.c) is used and somehow Sendmail is used #define SENDMAIL /usr/sbin/sendmail -t I want to know which command line is used, in general, to send emails that include this id field so that I could also use this id field when emailing incoming faxes. My question are either : Which debug or log option can I turn on to find out which command Asterisk is exactly sending to Sendmail/Exim4 to notify incoming voicemail ? Which sendmail command line option can I use to include a populated id field in log files ? Regards ___ -- 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] Which command line is used to send emails to notify incoming voicemail ?
2008/3/21, Olivier [EMAIL PROTECTED]: Hi, In exim4, I can see lines such as : mainlog.9:2008-03-12 08:53:28 1JZLmC-E7-0A = [EMAIL PROTECTED] U=root P=local S=43802 [EMAIL PROTECTED] In my voicemail.conf, I see : ; If you need to have an external program, i.e. /usr/bin/myapp called when a ;externnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp called when a ;externpass=/usr/bin/myapp So, I guess this line (from app_voicemail.c) is used and somehow Sendmail is used #define SENDMAIL /usr/sbin/sendmail -t I want to know which command line is used, in general, to send emails that include this id field so that I could also use this id field when emailing incoming faxes. My question are either : Which debug or log option can I turn on to find out which command Asterisk is exactly sending to Sendmail/Exim4 to notify incoming voicemail ? Which sendmail command line option can I use to include a populated id field in log files ? Regards In Asterisk full log, I can see Mar 20 14:36:41 DEBUG[29025] app_voicemail.c: Sent mail to [EMAIL PROTECTED] command '/usr/sbin/sendmail -t' But when I type /usr/sbin/sendmail [EMAIL PROTECTED] I can't see the same log lines with this id field. ___ -- 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] Newbie: Two problems with Asterisk Config, Please Help
Probably a codec issue. SIP debug while making a call would be helpful. Thanks, Steve Totaro On Fri, Mar 21, 2008 at 4:06 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I switched to Wengo and solved the one beatproblem. However, I am still not able to listen to the recorded .wav sound. Can anyone please point me to the right direction? How to listen to the .wav sound? Thanks, Pete On Fri, Mar 21, 2008 at 9:34 AM, Carlos Rojas [EMAIL PROTECTED] wrote: Hello, Do your verify, the codecs, of both clients, in your sip.conf? What codec do you use? Best Regards On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a beat sounds, and then nothing else. In the console, I can see: *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2001-081dd6e0, ) in new stack -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2001-081dd6e0, 2000) in new stack Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts 000160, len 000160) -- SIP/2001-081dd6e0 Playing 'vm-intro' (language 'en') Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts 000320, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts 000480, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037721, ts 000640, len 000160) Got RTP packet from192.168.1.102:8000 (type 00, seq 06, ts 1373137124, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037722, ts 000800, len 000160) Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037723, ts 000960, len 000160) Is it the prolem? First it sends to the public address of the the router, then it sends to the virtual IP. Is this the problem that causing my to hear just one beat sound and then no audio? Problem 2: The problem is isolated from Problem 1, cuz I run the SIP client on the same machine as the server, so there should not be network problem. I recorded some voice mails and they are stored as .wav files ok. When I tried to hear back the message, It does not work. Is there any configuration that I have to go through to have Asterisk to play .wav file? Thank you very much in advance for all your kind help. Pete ___ -- 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 ___ -- 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] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
Thanks Atis, On Tue, Mar 18, 2008 at 3:50 AM, Atis Lezdins [EMAIL PROTECTED] wrote: As for current problem - i suspect that device state don't get updated correctly for Queue application, so Queue tries to dial device, and call-limit blocks it from doing so. There's a patch, currently in testing (issue 12127), it should fix this, however if you intend to keep incominglimit to 1, and don't use local channels - there's nothing to worry about. I had gone through bug 12127. Currently I am testing with 1.4 Trunk, dated 20th. so the 12127 patch is applied. But even in trunk the behavior does not change. I still get the [Mar 21 18:18:59] ERROR[29689]: chan_sip.c:3266 update_call_counter: Call to peer '2501' rejected due to usage limit of 1 But some times, usually when I start testing, I get this new message, when a call is picked up by agent. [Mar 21 18:18:28] WARNING[29684]: app_queue.c:3002 try_calling: The device state of this queue member, Agent/2503, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. I had gone through the UPGRADE.txt and now my sip.conf is like the following: [general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes limitonpeer = yes [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw call-limit=1 nat=1 Also the queue show command shows that the agent is Not in use, though the call is being taken. Agent/2503 (dynamic) (Not in use) has taken 3 calls (last was 26 secs ago) sip show inuse command shows the following output for SIP/2501 (the phone of Agent/2503) asterisk:/etc/asterisk# asterisk -rx sip show inuse | grep 2501 2501 0 1 2501 1/0 1 To me it seems asterisk (or my configurations) is still not recognising the fact that SIP peers are busy when attending calls from queues. Thanks in advance for any assistance in resolving this, raj ___ -- 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] Hardphone SIP phone costs
are plenty of phones on the market which do SIP now - most modern Nokias do. I use an E90 Communicator, but the E95 is popular too, so I'm experimenting with using my mobile as my one phone, via Wi-Fi/SIP when I'm in the home/office and Out of curiosity, how do these phones handle the transition from Wi-Fi to GSM? Is it seamless? Can the transition occur when on a call? Thanks, John ___ -- 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] Hardphone SIP phone costs
John Faubion wrote: are plenty of phones on the market which do SIP now - most modern Nokias do. I use an E90 Communicator, but the E95 is popular too, so I'm experimenting with using my mobile as my one phone, via Wi-Fi/SIP when I'm in the home/office and Out of curiosity, how do these phones handle the transition from Wi-Fi to GSM? Is it seamless? Can the transition occur when on a call? Not seamless unless the cell phone provider offers such a service. You won't find that available in the US. So even though it's one phone, you'd have 2 numbers. Cell phone providers have no incentive to offer such a hand-off because they wouldn't make any money on the calls after they are handed over to the voip system. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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] Digium registration utility version 3.0.3 released
Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. The new versions of 'register' and 'asthostid' can be found at: http://downloads.digium.com/pub/register/linux If you experience any issues with this release, please contact Digium technical support.___ -- 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] Digium registration utility version 3.0.3 released
Terry Wilson wrote: Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. Great! What will it take to get the g729 codec module compiled against uClibc? Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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] TxFax in asterisk 1.4
Hi guys, I installed asterisk 1.4.17 and mISDN 1.1.7 with AGX addons to test the faxs. If I receive i do not have any problem, but i'm not able to send put any fax, i get always the same error: txfax_exec: transmission done with ast_read(chan) == NULL Anyone has txfax working with asterisk 1.4? I try to download app_rxfax.c and app_txfax.c with the asterisk.patch file but without success. On HYPERLINK http://www.soft-switch.orgwww.soft-switch.org is not longer available. Anyone can help me please ? TIA Giordano No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1335 - Release Date: 19/03/2008 9.54 ___ -- 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] Hardware supporting groundstart signalling
Hello! Recently I posted a question about an installation I have that was experiencing glare problems. The solutions presented were to use inverse inbound and outbound line groups and to use groundstart signalling. As it turns out, the Sangoma A400D card that is in use does NOT support groundstart. I've confirmed this with a Sangoma engineer and their support staff. I've also read that Digium products do not support groundstart signalling. Since glare is a common problem with analog PBX systems, it would make sense that groundstart is a common signalling type. Why do the major manufacturers not support this? If you're using groundstart, what hardware are you using? Thank you! Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- 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] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min
starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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] Hardphone SIP phone costs
On Fri, 21 Mar 2008, John Faubion wrote: are plenty of phones on the market which do SIP now - most modern Nokias do. I use an E90 Communicator, but the E95 is popular too, so I'm experimenting with using my mobile as my one phone, via Wi-Fi/SIP when I'm in the home/office and Out of curiosity, how do these phones handle the transition from Wi-Fi to GSM? Is it seamless? Can the transition occur when on a call? The ones I've used don't. You make/take a SIP call or a GSM call, but the 2 don't mix. I can live with that - for now. I have my Nokia set to default to making a SIP call, but it falls-back to using the network when that fails. (eg. when I go out of range). It goes back to WiFi mode when it comes into range of an access point it knows about and I've got Wi-Fi search on. I think there's still some resistance from the (UK) mobile telcos about SIP/VoIP on the phones as it's a competing technology, so there were reports of early phones being crippled by the network operators, but there's never been anything to stop you buying an un-branded phone and putting your own SIM card in. I think the operators are giving in though Some even offer free Skype software on the phones and calls (but not Skype out!) Femto cells might be the next best thing though as I'm really not a fan of VoIP over Wi-Fi. I'd want one with an Ethernet port and multiple SIP accounts, so my mobile could roam to my home/office cell and the cell could then either contact my mobile telco, or my VoIP provider, as required/desired. Something like this maybe: http://www.3g.co.uk/PR/Feb2007/4221.htm but a bit more open to let you use your own SIP service rather than the mobile telcos.. Gordon ___ -- 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] Digium registration utility version 3.0.3 released
On Fri, 21 Mar 2008 10:10:22 -0500, Darrick Hartman (lists) wrote: Terry Wilson wrote: Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. Great! What will it take to get the g729 codec module compiled against uClibc? Let me add my voice to the cry! This would be tremendously usefull for me! Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- 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] TxFax in asterisk 1.4
No, Steve removed it from his site. I have a copy of it @ http://www.taylortelephone.com/asterisk/ Jonn _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Friday, March 21, 2008 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TxFax in asterisk 1.4 Hi guys, I installed asterisk 1.4.17 and mISDN 1.1.7 with AGX addons to test the faxs. If I receive i do not have any problem, but i'm not able to send put any fax, i get always the same error: txfax_exec: transmission done with ast_read(chan) == NULL Anyone has txfax working with asterisk 1.4? I try to download app_rxfax.c and app_txfax.c with the asterisk.patch file but without success. On www.soft-switch.org is not longer available. Anyone can help me please ? TIA Giordano No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1335 - Release Date: 19/03/2008 9.54 ___ -- 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] R: TxFax in asterisk 1.4
Thanks very much, i will test it. Hi and thanks again Giordano Grandis e-mail : HYPERLINK mailto:[EMAIL PROTECTED][EMAIL PROTECTED] VoIP: HYPERLINK mailto:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] _HYPERLINK http://%5c%5cwww.invidea.it/www.invidea.it TecnoJest Srl Verrotti c/o Centro attività Espansione II, int 4, 65016 Montesilvano (PE) Tel [+39] 085 4450011- Fax [+39] 085 4459477 - PI 01635460684 Le informazioni contenute nella presente e-mail e nei documenti eventualmente allegati possono essere confidenziali e sono comunque riservate al destinatario della stessa. La loro diffusione, distribuzione e/o copiatura da parte di terzi è proibita. Se avete ricevuto questa comunicazione per errore, Vi preghiamo di informare immediatamente il mittente del messaggio e di distruggere questa e-mail. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. _ Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Jonn R Taylor Inviato: venerdì 21 marzo 2008 16.32 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] TxFax in asterisk 1.4 No, Steve removed it from his site. I have a copy of it @ HYPERLINK http://www.taylortelephone.com/asterisk/http://www.taylortelephone.com/asterisk/ Jonn _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Friday, March 21, 2008 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TxFax in asterisk 1.4 Hi guys, I installed asterisk 1.4.17 and mISDN 1.1.7 with AGX addons to test the faxs. If I receive i do not have any problem, but i'm not able to send put any fax, i get always the same error: txfax_exec: transmission done with ast_read(chan) == NULL Anyone has txfax working with asterisk 1.4? I try to download app_rxfax.c and app_txfax.c with the asterisk.patch file but without success. On HYPERLINK http://www.soft-switch.orgwww.soft-switch.org is not longer available. Anyone can help me please ? TIA Giordano No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1335 - Release Date: 19/03/2008 9.54 No virus found in this incoming message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1335 - Release Date: 19/03/2008 9.54 No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1335 - Release Date: 19/03/2008 9.54 image002.jpg___ -- 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] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min
Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min
My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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] Problem with user regsitration and ldap on SVN version
Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a sip show peers, no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I see this trace when trying to register: [Mar 21 16:53:54] DEBUG[12002] acl.c: Found IP address for this socket [Mar 21 16:53:54] DEBUG[12002] chan_sip.c: Allocating new SIP dialog for OWY3OTAwNzFhNDZhYWU5NTU0YTU1MGY4MzYwOTdlZjQ. - REGISTER (No RTP) [Mar 21 16:53:54] DEBUG[12002] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: substituted: string: 'dc=example, dc=com' = 'dc=example, dc=com' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: basedn: 'dc=example, dc=com' = 'dc=example, dc=com' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: Everything seems fine. [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: name='name' value='Pierre' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: name='host' value='dynamic' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: canreinvite LDAP value: no [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: regserver LDAP value: 0 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: objectClass LDAP value: AsteriskObject [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: objectClass LDAP value: AsteriskExtension [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: objectClass LDAP value: AsteriskSIPUser [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: objectClass LDAP value: top [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: context LDAP value: from-sip [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: context LDAP value: internal [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: type LDAP value: friend [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: callerid LDAP value: Pierre Bachelet 2001 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: fullcontact LDAP value: Pierre Bachelet [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: AstAccountSecret LDAP value: 1234 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: host LDAP value: dynamic [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: name LDAP value: Pierre [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: qualify LDAP value: no [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: language LDAP value: fr [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: ipaddr LDAP value: 0.0.0.0 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: port LDAP value: 0 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: regseconds LDAP value: 1206118346 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: defaultuser LDAP value: 0 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: canreinvite value: no [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: regserver value: 0 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: objectClass value: AsteriskObject [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: objectClass value: AsteriskExtension [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: objectClass value: AsteriskSIPUser [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: objectClass value: top [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: context value: from-sip [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: context value: internal [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: type value: friend [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: callerid value: Pierre Bachelet 2001 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: fullcontact value: Pierre Bachelet [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: AstAccountSecret value: 1234 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: host value: dynamic [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: name value: Pierre [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: qualify value: no [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(482) attribute_name: language value: fr [Mar 21 16:53:54] DEBUG[12002]
Re: [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min
The template website, page titles, and Gmail contact address surely aren't very convincing. Another crappy VoIP reseller that will fail in a few months taking a handful of customers down... assuming they're legit to begin with. --Tim - Original Message - From: Outback Dingo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 21, 2008 11:06:31 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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] Hardphone SIP phone costs
2008/3/21, Darrick Hartman (lists) [EMAIL PROTECTED]: John Faubion wrote: are plenty of phones on the market which do SIP now - most modern Nokias do. I use an E90 Communicator, but the E95 is popular too, so I'm experimenting with using my mobile as my one phone, via Wi-Fi/SIP when I'm in the home/office and Out of curiosity, how do these phones handle the transition from Wi-Fi to GSM? Is it seamless? Can the transition occur when on a call? Not seamless unless the cell phone provider offers such a service. If you're on call using GSM band, it is seamless. If you're on call using SIP/WiFi, it's up to SIP server to dial a new call to your mobile number and blind transfert previous call to it. Maybe some dual band phones are able to automatically accept some incoming GSM calls, put them in 3-way conference (of some kind) and wait for SIP server to end WiFi call without asking anything to user. Parts of this puzzle are here but integration should be rather hard. You won't find that available in the US. So even though it's one phone, you'd have 2 numbers. Cell phone providers have no incentive to offer such a hand-off because they wouldn't make any money on the calls after they are handed over to the voip system. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min
I think this type of abuse is well deserved due to the way he intended to advertise his business, so I'll add a bit of wood to the fire. How about the sign-up form?? Some serious HTML design work going on there. - Gonzalo On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson [EMAIL PROTECTED] wrote: The template website, page titles, and Gmail contact address surely aren't very convincing. Another crappy VoIP reseller that will fail in a few months taking a handful of customers down... assuming they're legit to begin with. --Tim - Original Message - From: Outback Dingo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 21, 2008 11:06:31 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] ----www.cdsportal.net---- wholesale voip provider --starting at 1.1 cent per min
We are terribly sorry is we insult anybody for posting this message here we only want to offer good prices with good quality to the ones who need it that's all We have a strong voip platform just try it. Thanks -Original Message- From: Gonzalo Servat [EMAIL PROTECTED] Sent: Friday, March 21, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min I think this type of abuse is well deserved due to the way he intended to advertise his business, so I'll add a bit of wood to the fire. How about the sign-up form?? Some serious HTML design work going on there. - Gonzalo On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson [EMAIL PROTECTED] wrote: The template website, page titles, and Gmail contact address surely aren't very convincing. Another crappy VoIP reseller that will fail in a few months taking a handful of customers down... assuming they're legit to begin with. --Tim - Original Message - From: Outback Dingo [EMAIL PROTECTED] [The entire original message is not included] ___ -- 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] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min
Piling on... InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic: C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: CDSPORTAL.NET Created on: 14-Mar-08 Expires on: 15-Mar-09 Last Updated on: 14-Mar-08 Administrative Contact: Almonte, Juan [EMAIL PROTECTED] JHALMONTE C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic (809) 220-3278 Judging by the site's purported function, it's nothing more than a front for telemarketers, autodialers, and other ilk of the telephony industry to annoy normal people with. How can you claim five 9's uptime when your domain isn't barely over a week old? Well, I guess if the system hasn't crashed within that first week. But that's hardly a valid measurement, unless you're comparing against Windows Millenium systems. I call scam. --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gonzalo Servat Sent: Friday, March 21, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] www.cdsportal.net wholesale voipprovider --starting at 1.1 cent per min I think this type of abuse is well deserved due to the way he intended to advertise his business, so I'll add a bit of wood to the fire. How about the sign-up form?? Some serious HTML design work going on there. - Gonzalo On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson [EMAIL PROTECTED] wrote: The template website, page titles, and Gmail contact address surely aren't very convincing. Another crappy VoIP reseller that will fail in a few months taking a handful of customers down... assuming they're legit to begin with. --Tim - Original Message - From: Outback Dingo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 21, 2008 11:06:31 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] Hardphone SIP phone costs
At 09:27 AM 3/21/2008, you wrote: If you're on call using SIP/WiFi, it's up to SIP server to dial a new call to your mobile number and blind transfert previous call to it. Maybe some dual band phones are able to automatically accept some incoming GSM calls, put them in 3-way conference (of some kind) and wait for SIP server to end WiFi call without asking anything to user. Parts of this puzzle are here but integration should be rather hard. T-mobile has a service in the USA for a few of their phones that will seamlessly move between WIFI and GSM. You pay based on where the call starts, if it starts GSM you pay for the minutes, if it starts WIFI, the call does not count against your minutes. But it's proprietary and costs $10/month. Ira ___ -- 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] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min
Better yet. They claim SIX 9's uptime :) On Mar 21, 2008, at 1:37 PM, Joshua Kinard wrote: Piling on... InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic: C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: CDSPORTAL.NET Created on: 14-Mar-08 Expires on: 15-Mar-09 Last Updated on: 14-Mar-08 Administrative Contact: Almonte, Juan [EMAIL PROTECTED] JHALMONTE C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic (809) 220-3278 Judging by the site's purported function, it's nothing more than a front for telemarketers, autodialers, and other ilk of the telephony industry to annoy normal people with. How can you claim five 9's uptime when your domain isn't barely over a week old? Well, I guess if the system hasn't crashed within that first week. But that's hardly a valid measurement, unless you're comparing against Windows Millenium systems. I call scam. --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ]On Behalf Of Gonzalo Servat Sent: Friday, March 21, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] www.cdsportal.net wholesale voipprovider --starting at 1.1 cent per min I think this type of abuse is well deserved due to the way he intended to advertise his business, so I'll add a bit of wood to the fire. How about the sign-up form?? Some serious HTML design work going on there. - Gonzalo On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson [EMAIL PROTECTED] wrote: The template website, page titles, and Gmail contact address surely aren't very convincing. Another crappy VoIP reseller that will fail in a few months taking a handful of customers down... assuming they're legit to begin with. --Tim - Original Message - From: Outback Dingo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 21, 2008 11:06:31 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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 ___ -- 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] Calls to sip extensions not defined
Hi all, new to the list and this is probably a basic question and
couldn't find anything clear googling around but I don't know how to
handle calls to sip extensions not defined on sip.conf while using
pattern matching. On my example I have sip extensions 10, 11, 12, and 13
on sip.conf. On a basic extension.conf I set up a pattern starting with
1 and a second digit should dial the sip extension entered by the user
and if the user don't pick up or is unavailable the call goes to the
user voicemail and then hangup. This basic setup can be seen next:
[default]
exten = _1X,1,Dial(SIP/${EXTEN},10)
exten = _1X,2,VoiceMail([EMAIL PROTECTED],u)
exten = _1X,3,HangUp()
Now, what happens if the user dials 15? Then the pattern is applied and
the asterisk tries to dial that sip extension that doesn't exist, the
next step that is the voicemail also fails as 15 is not defined on
voicemail.conf and finally reaches the last step where it hang ups. This
can be seen on the cli output copied below:
astbox*CLI
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/10-0820d8e0, SIP/15|10) in new
stack
[Mar 21 19:57:48] WARNING[14321]: chan_sip.c:2860 create_addr: No such
host: 15
[Mar 21 19:57:48] WARNING[14321]: app_dial.c: dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/10-0820d8e0, [EMAIL
PROTECTED]|u)
in new stack
[Mar 21 19:57:48] WARNING[14321]: app_voicemail.c:2808 leave_voicemail:
No entry in voicemail config file for '15'
-- Executing [EMAIL PROTECTED]:3] Hangup(SIP/10-0820d8e0, ) in new stack
== Spawn extension (default, 15, 3) exited non-zero on 'SIP/10-0820d8e0'
astbox*CLI
What I am looking for is to play Playback(pbx-invalid) if a user enters a
sip extension not created. I've been testing a few options using
DIALSTATUS, AVAILSTATUS and their values but without luck as if the sip
phone 11 is not registered the pbx-invalid message.
Thansk for reading and any suggestion will be welcome.
Richard
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Re: [asterisk-users] Calls to sip extensions not defined
On Fri, 21 Mar 2008, Ricardo B. wrote:
On my example I have sip extensions 10, 11, 12, and 13
on sip.conf. On a basic extension.conf I set up a pattern starting with
1 and a second digit should dial the sip extension entered by the user
and if the user don't pick up or is unavailable the call goes to the
user voicemail and then hangup. This basic setup can be seen next:
[default]
exten = _1X,1,Dial(SIP/${EXTEN},10)
exten = _1X,2,VoiceMail([EMAIL PROTECTED],u)
exten = _1X,3,HangUp()
Now, what happens if the user dials 15? Then the pattern is applied and
the asterisk tries to dial that sip extension that doesn't exist, the
next step that is the voicemail also fails as 15 is not defined on
voicemail.conf and finally reaches the last step where it hang ups.
What I am looking for is to play Playback(pbx-invalid) if a user enters a
sip extension not created.
While I didn't take the time to test it, the following should be close:
[default]
exten = _1[1-3],1, dial(sip/${EXTEN},10)
exten = _1[1-3],n, voicemail([EMAIL PROTECTED],u)
exten = _1[1-3],n, hangup
exten = i,1,playback(pbx-invalid)
exten = i,n,hangup
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Want to know Frequency and lenght of Frame
Zoa wrote: Mojo with Horan Company, LLC wrote: Aren't all the frames in asterisk 20ms long, no exceptions? Isn't ilbc the exception ? Even though the ilbc codec likes multiples of 50 for its frame size (Is this right?), I was under the impression that asterisk broke everything down to 20ms slin samples internally, unless it was just directly bridging two similarly-codeced channels. I would imagine that Sanjay meant zaptel hardware anyway, as the SIT is an in-band pattern meant for our ears. (I think that SIP would simply return a cause code out-of-band describing WHY the call failed, but would not pass any RTP audio.) Moj ___ -- 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] ----www.cdsportal.net---- wholesale voip provider
Why pay 1.1 cent's a minute for interconnecting to another Asterisk server for a high volume call center. Do people really understand what they are trying to sale and take an honest look into what they advertise. As a high volume user like a call center I would not connect my Asterisk Box to there Asterisk Box to a third Sip provider who then hands of to the Level 3 and so forth. With LD PRI's at sub penny rates, cutting out 2 or 3 other points of failure and added latency only make since. Also if your doing termination why are you worried about having all these other providers typically used for Origination. If you are going to be a provider you need to fork over the dough and do it right not buy something from someone, stick a device in the midle and resale it. This is looks like a kid who set up a Trixbox pc and trying to make a buck. http://www.ru-intouch.com/ruwho.php?action=detailsddomain=cdsportal.netserver=whois.opensrs.net Sure he has 99.9% uptime since he just this purchased site 3/14/2008. Matthew Warren My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- 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 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080321/1a72a82b/attachment-0001.htm -- Message: 2 Date: Fri, 21 Mar 2008 17:08:29 +0100 From: [EMAIL PROTECTED] Subject: [asterisk-users] Problem with user regsitration and ldap on SVN version To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a sip show peers, no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I see this trace when trying to register: [Mar 21 16:53:54] DEBUG[12002] acl.c: Found IP address for this socket [Mar 21 16:53:54] DEBUG[12002] chan_sip.c: Allocating new SIP dialog for OWY3OTAwNzFhNDZhYWU5NTU0YTU1MGY4MzYwOTdlZjQ. - REGISTER (No RTP) [Mar 21 16:53:54] DEBUG[12002] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: substituted: string: 'dc=example, dc=com' = 'dc=example, dc=com' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: basedn: 'dc=example, dc=com' = 'dc=example, dc=com' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: Everything seems fine. [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: name='name' value='Pierre' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: name='host' value='dynamic' [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: canreinvite LDAP value: no [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: regserver LDAP value: 0 [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: objectClass LDAP value: AsteriskObject [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289) attribute_name: objectClass LDAP value: AsteriskExtension [Mar 21 16:53:54] DEBUG[12002] res_config_ldap.c: LINE(289
Re: [asterisk-users] ----www.cdsportal.net---- wholesale voip provider
Let me tell you something i own 200 seats call center, besides have an IT company who develops applications based on asterisk, we are not kid just playing to get some money... so i move millions of min i also resell min to others call centers since 9 months ago now we just open to the public, so yes we bougth the domain a week ago but this not indicates we are trying to steal or someting that`s why we give a free test so if you liked you take it that`s it. Again CDS want to apologise to all users to use this channel to send this message it seems to be a improper channel for that. Regards On Fri, Mar 21, 2008 at 7:36 PM, Matthew Warren [EMAIL PROTECTED] wrote: Why pay 1.1 cent's a minute for interconnecting to another Asterisk server for a high volume call center. Do people really understand what they are trying to sale and take an honest look into what they advertise. As a high volume user like a call center I would not connect my Asterisk Box to there Asterisk Box to a third Sip provider who then hands of to the Level 3 and so forth. With LD PRI's at sub penny rates, cutting out 2 or 3 other points of failure and added latency only make since. Also if your doing termination why are you worried about having all these other providers typically used for Origination. If you are going to be a provider you need to fork over the dough and do it right not buy something from someone, stick a device in the midle and resale it. This is looks like a kid who set up a Trixbox pc and trying to make a buck. http://www.ru-intouch.com/ruwho.php?action=detailsddomain=cdsportal.netserver=whois.opensrs.net Sure he has 99.9% uptime since he just this purchased site 3/14/2008. Matthew Warren ___ -- 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] --POSIBLE SPAM-- Peticion on line Parque de atracciones mcdonals
Buenas noches , os envió esta petición on line para que la firméis y
la reenviéis ya que la considero muy importante,
Un saludo , gracias por apoyarme y disculpar las molestias.
http://www.firmasonline.com/1firmas/camp1.asp?C=1285
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Re: [asterisk-users] Want to know Frequency and lenght of Frame
Mojo wrote: [EMAIL PROTECTED] wrote: I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Aren't all the frames in asterisk 20ms long, no exceptions? 1.0 1.2 Yes (with the possible exception fo iLBC) 1.4 1.6 No The default in 20ms, but can be changed per user/peer/codec ___ -- 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
