Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
Shift + Page Up and Shift + Page Down. Leif Madsen told me this in 2005 when I was new to Linux and Asterisk, at an Asterisk seminar in Mississauga. Thanks Leif, it made my life easier to scroll through the logs. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-04 11:36 PM, bruce bruce bruceb...@gmail.com wrote: And the 20k+ lines is where it's really hard to handle. The scroll bar is too small and I was wishing there was an easy page up or page down function maybe to it rather than using the mouse. Thanks for the input. On Tue, Jun 29, 2010 at 11:13 AM, Danny Nicholas da...@debsinc.com wrote: I use PUTTY 0.58 a... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hold and Retrieve the call through AGI
Dear All, Is there anyway to put the call on Hold and Retrieve the call based on external configurations through AGI? Please help me... Regards, Velusamy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On Sun, 4 Jul 2010, bruce bruce wrote: And the 20k+ lines is where it's really hard to handle. The scroll bar is too small and I was wishing there was an easy page up or page down function maybe to it rather than using the mouse. No-one's mentioned 'screen' yet. Use putty to connect to a *ix host and run screen... Gordon Thanks for the input. On Tue, Jun 29, 2010 at 11:13 AM, Danny Nicholas da...@debsinc.com wrote: I use PUTTY 0.58 and have Window title and scroll control for 20K+ lines. It could use some improvements, but it is more than adequate for green screen control. The quality of Putty and many other applications depends on how you choose to control it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A. Anderson Sent: Tuesday, June 29, 2010 10:08 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY? On 06/29/2010 06:53 AM, bruce bruce wrote: Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etcbut it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the IP address as the title of the window would help a lot if you have many different servers opened at the same time. I haven't used Putty for several months and that was with a setup I'd made several years ago so I can't, off the top of my head, tell you how I did it; but I had the remote system's name or IP in the window title bar. It might nave been the name I saved the connection as. Look in the configuration under Terminal. Something like a %s and make sure the terminal type is either Linux or ANSI. Again too long ago. Rod -- Can you please weigh in and tell me what your favorite terminal software is and why? Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo problem in VoIP-calls
Hello Gareth, echo also appears when making calls with a SIP phone. These are outgoing calls. Another site now also gives feedback on echo, telling they sometimes also have echo on outgoing calls and if they recall right then sometimes also on incoming calls (coming from a queue). This one site that now also gives feedback on echo has a fiber optic internet connection, so I don't think the latency plays a role here. I will now turn off the buffer in sip.conf and see how this goes... I hope I can resolve this echo-problem. Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does my IAX2 trunk between two office hangup a channel after 30 seconds? Can you share your IAX2 trunking configuration? URGENT HELP much appreciated
Dear Please send us, your iax configurations. best On Mon, Jul 5, 2010 at 7:10 AM, bruce bruce bruceb...@gmail.com wrote: Hi guys, I have two Asterisk servers (with FreePBX) connected together with IAX2 trunking. When I call from server A-B call connects but hangs up after 30 seconds. What could be cause? Can anyone please share working configuration between two asterisk server in IAX2 trunking for FreePBX? Thanks a lot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in establish call from a2billing users.
add the a2billing configurations to the sip.conf best On Thu, Jul 1, 2010 at 7:34 PM, bruce bruce bruceb...@gmail.com wrote: Yes, you are missing a whole bunch of configurations from creating SIP users to making sure they show as peers on Asterisk to making sure you use dnid, etc.You probably might want to search google for some configuration help On Wed, Jun 30, 2010 at 11:24 AM, gokulakrishnan alagudr...@gmail.comwrote: Hi All, I installed a2billing with asterisk FreePBX . I can able to login and make a call with FreePBX but when i am using the users which is created in a2billing the call was not established . I know somewhere i missed the configuration please any one help me to resolve this issue . Thanks in advance. regards, gokul., -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with extensions in IVR and queues
please send your extension.conf 2010/6/30 Anahi Ludueña a_ludu...@hotmail.com Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, * * * -- * *Anahi Ludueña* -- ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! http://www.ayudartepodria.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP response 482 Loop Detected
Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial our clients based on their email address. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
Has anyone mentioned Teraterm in this thread? I know it's very old but I also know it worked well with XP. I preferred it over Putty, but I haven't used Putty in years either. Nowadays, I use mostly Mac with occasional virtual XP - and the OS X terminal is great. It's a little surprising that no one has written a more modern version of something like Teraterm, but maybe the majority of Windows users don't do SSH? The fact is that when I mention SFTP to them (we don't do ftp at all usually) I can hear the crickets over the phone. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Trunk configuration problem - fromdomain
Hello, I'm trying to register to my provider sip trunk, I got from him an host IP (a.b.c.d) to connect to and my provider recognize me based on the fixed IP (x.y.z.w) he gave me (no need for username and password) In the sip.conf I add: [mytrunk] type=friend insecure=no host=a.b.c.d fromdomain=x.y.z.w qualify=3600 nat=no ; change to yes if you are behind NAT bindport=5060 bindaddr=0.0.0.0 context=default disallow=all allow=ulaw allow=alaw Now, my asterisk resides in my internal network (10.100.101.107) and in the SIP requests that sent to the provider I can see (via a sniffer) that the From and Contact fields have - sip:aster...@10.100.101.107 and not the x.y.z.w I expected to see as a result of the fromdomain=x.y.z.w. Any idea? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [NAT] * + private IP + locked-down firewalls?
Hello In case Asterisk is used in a private LAN behind a firewall while allowing remote SIP clients to connect from the Net, we must open UDP5060 for SIP and a range of UDP ports (as set in rtp.conf) so let incoming voice packets. Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP... 1. Can Asterisk use eg. STUN to open those ports and keep them open through keep-alive packets? If not, is there another solution to solve this issue in this context? 2. In case a dynamic solution is available, does Asterisk provide a tool to monitor that the ports are correctly open, so as to ease problem-solving in case a customer has a problem and needs to be helped over the phone? Knowing that ports on the firewalls are correctly open is one less area to worry about in case Asterisk doesn't work as planned. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 482 Loop Detected
- Original Message - Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial our clients based on their email address. Grabbing a SIP debug I see: --- Transmitting (no NAT) to 10.172.120.5:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060 From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u To: sip:us...@seconddomain.com Call-ID: 66b3314cc6d1-jxu0nhluv4zt CSeq: 2 INVITE Server: secret Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: sip:us...@172.30.14.8 Content-Length: 0 And am guessing that as the source from IP matches the Contact: address Asterisk sees that as a loop ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On Mon, Jul 05, 2010 at 12:09:30PM +0200, Randy R wrote: Has anyone mentioned Teraterm in this thread? I know it's very old but I also know it worked well with XP. Teraterm only supports the old, insecure and much less capable ssh1 protocol, IIRC. Many recent SSHDs disable ssh1 support nowadays. Don't use it. I preferred it over Putty, but I haven't used Putty in years either. Nowadays, I use mostly Mac with occasional virtual XP - and the OS X terminal is great. It's a little surprising that no one has written a more modern version of something like Teraterm, but maybe the majority of Windows users don't do SSH? What's wrong with putty? Oh, and: see my previous link for msysgit. The fact is that when I mention SFTP to them (we don't do ftp at all usually) I can hear the crickets over the phone. (SFTP is one of the new features of the SSH2 protocol) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reg. EMT-22 IP Phone
Hi I have a EMT-22 IP Phone. I need user name and password to access it from ip address. Thanks Gsphull -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
PS: http://www.ayera.com/teraterm/ I'm pretty sure there was a last update or patch or something because -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On Mon, Jul 5, 2010 at 1:43 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Teraterm only supports the old, insecure and much less capable ssh1 protocol, IIRC. Many recent SSHDs disable ssh1 support nowadays. Don't use it. I'm pretty sure there was a last update or patch or something because we only use ssh2 on our servers. It's been at least 2 years since I used Windows for SSH though. What's wrong with putty? Nothing's wrong with it, I just didn't like it as well. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On Mon, 5 Jul 2010 14:05:09 +0200, Randy R wrote: PS: http://www.ayera.com/teraterm/ I'm pretty sure there was a last update or patch or something because For as long as I have used Asterisk I have used either the freeware PuTTY or a commercial SSH/SFTP client called Private Shell. http://www.privateshell.com/ Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in establish call from a2billing users.
hello you must to do a configuration of yor sip.conf like that [the login of sip] type=friend context=default secret=(the password of sip ) host=dynamic dtmfmode=auto disallow=all allow=alaw allow=ulaw qualify=yes Regads 2010/7/5 Pezhman Lali l...@lopl.net add the a2billing configurations to the sip.conf best On Thu, Jul 1, 2010 at 7:34 PM, bruce bruce bruceb...@gmail.com wrote: Yes, you are missing a whole bunch of configurations from creating SIP users to making sure they show as peers on Asterisk to making sure you use dnid, etc.You probably might want to search google for some configuration help On Wed, Jun 30, 2010 at 11:24 AM, gokulakrishnan alagudr...@gmail.comwrote: Hi All, I installed a2billing with asterisk FreePBX . I can able to login and make a call with FreePBX but when i am using the users which is created in a2billing the call was not established . I know somewhere i missed the configuration please any one help me to resolve this issue . Thanks in advance. regards, gokul., -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_fax_digium and T.38 error correction
Hello, i just had some fax abortions because of some packet loss. so i startet to examine in the pcap recording from the res_fax_digium, if the T.38 EC mode redundancy was really used. So i watched into it, and compared it with a t.38 pcap from spandsp (same asterisk setup, but with app_fax) and i see differences in t38.error_recovery (error-recovery: secondary-ifp-packets) With spandsp here are three items, and with res_fax_digium zero items. (t38.secondary_ifp_packets) I this the the t.38 error correction? I ask this questions, because the fax for asterisk admin manual, there are no information about the T.38 error correction, and if i better use Redundancy or FEC. Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with extensions in IVR and queues
hello, i had the same issue when using x-lite when i verify i found that the issue is related to configuration of x-lite i change the value in x-lite option and now there is no issue all function good Hope it can help you 2010/6/30 Anahi Ludueña a_ludu...@hotmail.com Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, * -- * *Anahi Ludueña* -- ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! http://www.ayudartepodria.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with ulaw/g729 translation
Dear Folks, I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is simple: connection to the PSTN directly via SIP, using g729 codec, and connection to the softphones (X-lite 3.0 build 56125) trought local network, using ulaw codec. Sometimes, I got messages like: [Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported SDP media type in offer: image 65344 udptl t38 And then a lot of messages like: [Jul 1 15:27:00] WARNING[26549]: translate.c:274 ast_translator_build_path: No translator path from alaw to unknown That's stopping the phone system. When I got the messages, I can't make or receive calls. Then, a few minutes later (or when I stop and start asterisk), the phone system back to work again. Some confs and system status: sip.conf: [1050] ; THAT'S A SOFTPHONE type=friend host=dynamic callerid=Softphone 1050 secret= context=call-center disallow=all allow=alaw allow=ulaw dtmfmode=rfc2833 canreinvite=yes nat=no qualify=yes call-limit=1 allowtransfer=yes insecure=no promiscredir=no useclientcode=no videosupport=no [7600] ; THAT'S PSTN CONNECTION username=7600 type=friend secret= qualify=no port=5060 nat=yes mailbox=7...@default host=dynamic dtmfmode=rfc2833 context=out canreinvite=no callerid=7600 disallow=all allow=g729 [sipgvt] ; THAT'S PSTN CONNECTION username=1121317600 type=peer secret= port=5060 insecure=very host=gvt.com.br fromuser=1121317600 fromdomain=gvt.com.br dtmfmode=rfc2833 context=in disallow=all allow=g729 neuwald01*CLI g729 show licenses 3/8 encoders/decoders of 30 licensed channels are currently in use Licenses Found: File: G729-x.lic -- Key: G729- -- Host-ID: x -- Channels: 30 (Expires: 2030-06-07) (OK) neuwald01*CLI core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) 65536 (1 16) (0x1) image jpeg (JPEG image) 131072 (1 17) (0x2) imagepng (PNG image) 262144 (1 18) (0x4) video h261 (H.261 Video) 524288 (1 19) (0x8) video h263 (H.263 Video) 1048576 (1 20) (0x10) video h263p (H.263+ Video) 2097152 (1 21) (0x20) video h264 (H.264 Video) neuwald01*CLI core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- --- - gsm- -222 21 37 --2 - ulaw- 2-12 21 37 --2 - alaw- 21-2 21 37 --2 - g726aal2- 222- 21 37 --2 - adpcm- 2222 -1 37 --2 - slin- 1111 1- 26 --1 - lpc10- 2222 21 -7 --2 - g729- 2222 21 3- --2 - speex- ---- -- -- --- - ilbc- ---- -- -- --- - g726- 2222 21 37 --- - g722- ---- -- -- --- - Any idea? Thanks, Felipe Neuwald. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] How to change the IP in the SIP contact header
Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: Polycom sip:100@ x.y.z.w;tag=as7435100b To: sip:144@ a.b.c.d Contact: sip:1...@10.100.101.107 Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change the IP in the SIP contact header
Have you tried setting externip= In the [general] of your sip.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman Sent: Monday, July 05, 2010 1:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to change the IP in the SIP contact header Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: Polycom sip:100@ x.y.z.w;tag=as7435100b To: sip:144@ a.b.c.d Contact: sip:1...@10.100.101.107 Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 482 Loop Detected
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: - Original Message - Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial our clients based on their email address. Grabbing a SIP debug I see: --- Transmitting (no NAT) to 10.172.120.5:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060 From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u To: sip:us...@seconddomain.com Call-ID: 66b3314cc6d1-jxu0nhluv4zt CSeq: 2 INVITE Server: secret Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: sip:us...@172.30.14.8 Content-Length: 0 And am guessing that as the source from IP matches the Contact: address Asterisk sees that as a loop ? I don't know these things, but you should probably post more of a SIP trace. Maybe turn on full sip debug to a file for long enough to see what the SIP conversation looks like that asterisk 1.6.2.9 is having with itself. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On Mon, Jul 5, 2010 at 5:05 AM, Randy R randulo2...@gmail.com wrote: PS: http://www.ayera.com/teraterm/ I'm pretty sure there was a last update or patch or something because Whats different about teraterm compared to putty? I know back in the day I used to send files to my linux box with xmodem over ssh. Does this newer version do that? :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 482 Loop Detected
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients. We run a multi-tenant server
and have set sip.conf to forward calls to a public context based on
incoming domain name. This was all working before but not it is
complaining of a loop back as the source and target server are the
same.
Any ideas on how to overcome this problem as we dial our clients
based
on their email address.
Grabbing a SIP debug I see:
--- Transmitting (no NAT) to 10.172.120.5:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u
To: sip:us...@seconddomain.com
Call-ID: 66b3314cc6d1-jxu0nhluv4zt
CSeq: 2 INVITE
Server: secret
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:us...@172.30.14.8
Content-Length: 0
And am guessing that as the source from IP matches the Contact:
address Asterisk sees that as a loop ?
I don't know these things, but you should probably post more of a SIP
trace. Maybe turn on full sip debug to a file for long enough to see
what the SIP conversation looks like that asterisk 1.6.2.9 is having
with itself.
From what I have read hairpin calls are not supported by asterisk; so am
guessing something has been fixed in the 1.6.2.X branch that should have not
worked in 1.6.1.X anyway :) While I continue the research have implemented
using a workaround via the AstDB and the following changes to the uri-dial
plan:
exten =
_[a-zA-Z0-9].,n,GotoIf(${DB_EXISTS(URI/${ext...@${sipdomain})}?inturi:exturi)
exten = _[a-zA-Z0-9].,n(inturi),Goto(${DB(URI/${ext...@${sipdomain})})
exten = _[a-zA-Z0-9].,n(exturi),Macro(uridial,${ext...@${sipdomain})
This is a bit of pain as we have to make sure we update the DB when a new
inbound URI is added; though it works and means we can stick with the 1.6.2.X
branch.
Would be interested to hear from a dev though as to whether they think it
should work as we originally had it configured ?
--
Thanks, Phil
--
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Re: [asterisk-users] How to change the IP in the SIP contact header
Yes, I tried and it did not solve the problem, Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A. Stapleton Sent: Monday, July 05, 2010 9:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to change the IP in the SIP contact header Have you tried setting externip= In the [general] of your sip.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman Sent: Monday, July 05, 2010 1:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to change the IP in the SIP contact header Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: Polycom sip:100@ x.y.z.w;tag=as7435100b To: sip:144@ a.b.c.d Contact: sip:1...@10.100.101.107 Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/05/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change the IP in the SIP contact header
By definition, correct values for localnet, externip and nat=yes for this trunk should solve this problem. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-05 3:40 PM, Eyal Goltzman egoltz...@gmail.com wrote: Yes, I tried and it did not solve the problem, Thanks *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jamie A. Stapleton *Sent:* Monday, July 05, 2010 9:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] How to change the IP in the SIP contact header Have you tried setting externip= In the [general] of your sip.conf? From: asterisk-... No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/05/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reinvite to alaw after T.38 reception
I'm having issues with T.38 on Asterisk 1.6.2.8. A few lines are received OK
and then I get only garbage. I'm using ReceiveFAX() provided by app_fax to
receive the faxes.
After talking to the engineers on the telco, they said Asterisk is sending a
REINVITE to alaw after the T.38 reception is complete, and that could be the
cause of the problems.
I personally am not totally convinced of that, but they asked me if it's
possible to make Asterisk not reinvite to alaw after a T.38 fax reception. Is
that possible at all?
Here's the relevant sip.conf and extensions.conf portions:
[voxip]
username=5421047000
nat=yes
type=peer
secret=supersecret
port=5060
canreinvite=no
insecure=port,invite
host=10.150.65.16
fromuser=5421047000
fromdomain=10.153.66.146
dtmfmode=rfc2833
context=entrada-e1
disallow=all
allow=alaw
qualify=no
t38pt_udptl=yes
[macro-recebefax]
exten = s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1])
exten = s,n,Set(FAXCOUNT=${DB(fax/count)})
exten = s,n,Set(FAXFILE=fax-${DB(fax/count)}-rx)
exten = s,n,Set(LOCALSTATIONID=5421047008)
exten = s,n,ReceiveFAX(/var/spool/asterisk/fax/${FAXFILE}.tif)
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000
- Kyle Kienapfel doctor.w...@gmail.com escreveu:
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients. We run a multi-tenant
server
and have set sip.conf to forward calls to a public context based
on
incoming domain name. This was all working before but not it is
complaining of a loop back as the source and target server are the
same.
Any ideas on how to overcome this problem as we dial our clients
based
on their email address.
Grabbing a SIP debug I see:
--- Transmitting (no NAT) to 10.172.120.5:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
From: User A sip:us...@172.30.14.8;tag=c3zqlidz1u
To: sip:us...@seconddomain.com
Call-ID: 66b3314cc6d1-jxu0nhluv4zt
CSeq: 2 INVITE
Server: secret
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:us...@172.30.14.8
Content-Length: 0
And am guessing that as the source from IP matches the Contact:
address Asterisk sees that as a loop ?
I don't know these things, but you should probably post more of a SIP
trace. Maybe turn on full sip debug to a file for long enough to see
what the SIP conversation looks like that asterisk 1.6.2.9 is having
with itself.
--
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[asterisk-users] Anybody with experience with Aculab Groomer II
Hi, Does anybody have experience working with Aculab groomer II, to convert between ISDN E1 and non-ISDN T1, or anything similar. I am looking for sample config files. We have asterisk as ISDN E1, but for testing we set it up as regular T1 if we get sample config files. Zeeshan A Zakaria -- www.ilovetovoip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi on solaris
Hello all, Does anybody know if is it possible to install dahdi on solaris 10? I've only found a zaptel modified code for solaris at solarisvoip site. I'd appreciate any comment or experience about asterisk + dahdi/zaptel on solaris.. Best regards, Caio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi on solaris
Hi Claudio, As far as I am aware, dahdi is not able to compile on Solaris, although I've not attempted to compile it. There may be others out there that may have better experience than I with dahdi on Solaris. Thanks Bruce -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Claudio Furrer Sent: 05 July 2010 22:11 To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi on solaris Hello all, Does anybody know if is it possible to install dahdi on solaris 10? I've only found a zaptel modified code for solaris at solarisvoip site. I'd appreciate any comment or experience about asterisk + dahdi/zaptel on solaris.. Best regards, Caio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html There's not enough info on Dialogic and Asterisk based forums/mailing-list. and Dialogic boards documentation is based on Windows. Thanks: Amardeep -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with voicemail contexts
Hello all, I am putting together an installation for our organization. My dialplan has most users in context [inside], and a separate [users] context includes the inside context. My voicemail config file has these users in a [users] context. I did this so I could get the name directory to work and vector calls to the right extensions. Now, however, I don't get message waiting lamp to show up on the phones and when the recipient of a voicemail tries to retrieve the message Alyson says you have no messages. This is true. The message doesn't get moved into the INBOX directory for the mailbox. I am flummoxed. Any ideas welcome! Cassius Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote: Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Your not going to find much; there is no channel driver for Dialogic. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 97 issues marked 'Ready for Testing'
List, Its been 2 weeks since my previous email and this time I am linking all 97 issues marked 'Ready for Testing' [1]. Simply follow the link, view the available patches, download, compile and install. Report your result into the actual issue, we can them continue to triage the issue. The more testers the better. If you have any problems or questions, jump on #asterisk-testing on Freenode. [1] https://issues.asterisk.org/search.php?project_id=7status_id=55sticky_issues=onsortby=last_updateddir=DESChide_status_id=90 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Externnotify on pollmailboxes=yes
Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks for any help,-Eric _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
On Mon, 2010-07-05 at 19:59 -0400, Paul Belanger wrote: On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote: Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Your not going to find much; there is no channel driver for Dialogic. Dialogic drivers were only supported in Asterisk Business Edition (ABE) and never in the free version because of proprietary drivers. Now that ABE is discontinued there is no support for Dialogic cards. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with voicemail contexts
OK, feeling very stupid right now. The test mailbox had delete=yes option set. All cleared up; sorry for cluttering up the list. Cassius snip Now, however, I don't get message waiting lamp to show up on the phones and when the recipient of a voicemail tries to retrieve the message Alyson says you have no messages. This is true. The message doesn't get moved into the INBOX directory for the mailbox. snip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
What do you mean now that ABE is discontinued? My company payed thousands of dollars this year for the product and the support it provides! -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 5, 2010, at 5:42 PM, Carlos Chavez wrote: On Mon, 2010-07-05 at 19:59 -0400, Paul Belanger wrote: On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote: Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Your not going to find much; there is no channel driver for Dialogic. Dialogic drivers were only supported in Asterisk Business Edition (ABE) and never in the free version because of proprietary drivers. Now that ABE is discontinued there is no support for Dialogic cards. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
On Mon, 5 Jul 2010 18:17:48 -0700, Jim Dickenson wrote What do you mean now that ABE is discontinued? My company payed thousands of dollars this year for the product and the support it provides! Well, last year when I took my dCAP that is what my instructor commented. Since Digium now has Switchvox they were to discontinue ABE but I guess they have not done that yet. I suppose the installed base is big enough to sustain the product. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
On Monday 05 July 2010 20:17:48 Jim Dickenson wrote: What do you mean now that ABE is discontinued? My company payed thousands of dollars this year for the product and the support it provides! Those who paid for ABE support will continue to get it, and those who really want ABE can still purchase a license, but we found that a good many people who purchased ABE simply wanted better support for their open source installations and only moved to ABE for that reason. Digium now provides paid support for the open source edition of Asterisk and is de-emphasizing ABE as the path for people who want paid support. That's really all there is to it. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Externnotify on pollmailboxes=yes
On Monday 05 July 2010 19:17:00 Eric Hiller wrote: Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks for any help, The externnotify script is only run when voicemail is left through the Voicemail application. I'm not sure if you're leaving voicemail messages through an external app or if you're expecting the script to be run when the count changes, but it's only run in that single case. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
I am writing to you privately because I am an asterisk consultant and if you need any help I can help you for a fee. I have worked with dialogic cards for several years, until I kicked them out my life when Intel bought Dialogic J Having said that however, these are my thoughts: You have to switch your thought frame when you go from dialogic to asterisk. Although asterisk supports dialogic drivers, the entire frame is moot, and the support is in reality just marketing plot, since IVR programming options from asterisk are just different. Beginning with the fact that dialogic's call handling is hardware based and asterisk's is software based. In short, the dialogic card will only end up being an interface card, and all the programming logic will have to be rewritten. In any case, anything I can do for you guys, just ask C. Savinovich From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of AMARDEEP SINGH Sent: Monday, July 05, 2010 7:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk? Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html There's not enough info on Dialogic and Asterisk based forums/mailing-list. and Dialogic boards documentation is based on Windows. Thanks: Amardeep -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On Mon, Jul 5, 2010 at 9:03 PM, Kyle Kienapfel doctor.w...@gmail.com wrote: Whats different about teraterm compared to putty? I know back in the day I used to send files to my linux box with xmodem over ssh. Does this newer version do that? :) THe next time I turn on the XP box, I'll try to remember to look. It's been a long time since I've used it for daily work. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
