[asterisk-users] ekiga register problems
returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ekiga register problems
204/20466.176.193.46D 5063 Unmonitored It just came up after a reboot on its own??? Go figure, windows problem! Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, May 29, 2007 12:17 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ekiga register problems On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote: returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall What exactly is the problem you get? What is the line for 204 in 'sip show peers'? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto/forced call
Can anyone guide me to a how to on automating a call? I know a little piece of code (normally python) has to be place some where and then a file has to be mv into the spooler. Where do I get the run down? I have a button on another application that sends an email and I want it to also send a text message through asterisk! Brad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk GUI issue, minor
I installed the asterisk GUI, Asterisk web manager, it loads fine, but if I go to the AGI section, I get a permission denied Obviously apache cannot access the /etc/asterisk directory. I added apache as group, but still the same problem. Suggestion any one? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom Provisioning Problems
Access the phones through the web interface, Compare version numbers with the phones that work Compare only with other 501 phones Make sure all settings are identical, most polycom web interfaces will loose there setting adjustments if you click on another tab, so do one page at a time, click save, then let it reboot, then go to next section. Also, is you asterisk server local? If remote, and the above does not work, look at routing. Are you behind NAT? Try and access the phone via telnet from a remote server to the auth ports of the phone and vice verse. (I know you can telnet from a Cisco phone, I would imagine polycom has a similar features) Let me know if this helps!!! Polycom is very picky! Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: Wednesday, April 25, 2007 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Provisioning Problems Hello I am having some difficulties provisioning a set of polycom 501 phones, while another set of phones are working just fine. My Asterisk box is dual homed. On one network, where the asterisk box runs dhcpd and there are only phones, provisioning works as expected. However, for phones that are connected thru the other interface (and receive their IP address from a separate router), they are not provisioning. To add to the confusion, it seems that they fail in inconsistent ways. Even after specifying the FTP server address, name and password, these phones will complain that they cannot connect to the server, and begin loading the stored configuration. In addition, when they come up, their dates are set to Jan 1, 2001. (I think I can fix this by specifying the snmtp address, but the other phones seem to be able to find the snmtp on their own.) In inspecting the MAC-boot.log files, the phones that fail have CDP enabled, while the phones that succeed have CDP disabled. I think this is Continual Data Protection, but don't see where to disable it on the phone interface. Is this a cause of the failure? Any insight will be greatly appreciated. Thanks -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Pix firewalls
Pix usually uses NAT, A quick fix is to simply forward the ports in your NAT statements. If the pix is new, call Cisco and cheat like I do so often! Brad _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Wednesday, April 25, 2007 9:31 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk Pix firewalls Don This may not be a solution to your question, but I would like to share that Ive been having one way audio issues when connecting point to sight to a PIX 515E using SIP. I changed to IAX and this is working perfectly now. It was paynless to configure IAX2, so you might want to consider it. Ed From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don E. Wisdom Sent: Tuesday, April 24, 2007 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Pix firewalls Hi, I asked this last week but i didn't get any answer So i will elaborate on my question. I need to setup a pix 515 firewall (running 7.2.2 OS) to allow sip traffic thru it from a sip phone wherever i may be. The pix is where all my servers are colocated and i will need to connect thru it from softphones / hardphones wherever i happen to be traveling. I need help setting up the pix for inbound and outbound sip/iax traffic. Any help would be greatly appreciated. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SLES?
This question should really be asked at Linux. Basically FC, Red hat, Centos and SUSE are all the same. Some minor security defaults and a few directory changes. Last time I check (it has been some time now); All of the above on their enterprise level basically only supported the install, updates (which are free on Yum anyways) and some minor other stuff. More advanced was a few thousand and $15,000 for priority for a year. Digium rates have gone up for support, but WELL WORTH IT when it deals with Asterisk and Linux, minor to advanced! I pay the piper from time to time and always get the job done quickly! Outside of that, this mailing list is a great place for support, we all work together! Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet Sent: Wednesday, April 25, 2007 5:24 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SLES? On Mon, 2007-04-23 at 04:46 +0300, Tzafrir Cohen wrote: On Mon, Apr 23, 2007 at 01:49:12AM +0200, Hans Witvliet wrote: Hi all, Just curious, Quite a while a go, i was checking for supported SW-platform. AFAIR, it was RHES and SLES Now it's only RHES-4 and FC-3 or FC-4. Not a single syllable about CentOS or SLES-9 or SLES-10 It probably just runs fine, but any chance of getting support for their *-enterprise version? (just in case of, if one needs it) Asterisk is an official package of SLES. Consider asking them as well regarding support (including newer versions of Asterisk). I knew that it included in the retail version (prof-10.x) and in open-suse (no support). And it was surely NOT included in SLES-9. At that time i suggested to get it included with sles-10, but marcus/andreas replied that they considered asterisk not stable enough to be able to have SLA-contracts connected to it, hence they would not include it. I'm pretty sure that one way or another, asterisk will just work fine on SLES-10. Point is however, that management would like to see a possible backup for support, in case the shit hits the fan. Official, with SLA-contracts and so on It took years to get SLES into the organisation, so open-suse, fedora or Centos are out-of-the-question, and RHEL will be another long struggle. hw -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] call dispatching - legacy application
Then you take the number you get from your database and put it into the asterisk spooler. Remember, the temp file you create has to be moved to the spooler using the mv command. Nothing else works. There might be one other step, I am not sure with Asterisk 1.4. I had a friend help me do it before and he said he had to write a little piece of python code to make it work properly (we were making asterisk call phone automatically). I am not sure if you will need this or not. I know the process because I had it done for me before. I am at the beginning trying to do the same thing, though my php is rusty. Maybe you can hook a brother up with the proper code to grab caller id and query mysql? To answer your question, Yes, you are on the right track! Brad [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of adriano ghezzi Sent: Wednesday, April 25, 2007 6:13 PM To: asterisk users Subject: [asterisk-users] call dispatching - legacy application Hy all need to preprocess 1) incoming call get caller id lookup some info in my db, 2) based on the result dispatch the call to the right operator step 1 is ok I developped a small .php script that connect manager and parse events, now I have to tell AAH do dispatch call to the right operator questions 1) is this the right practice ? 2) where to find a complete manager api reference, (to buy too) note that there is a legacy application that query the db actually php script send the request to this app and wait for response I'm a programmer at very first installation of AAH , just testing capabilities thanks in advance for any help and suggestion. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] No Audio with SIP to only one provider whenswitching servers
I would not rule your firewall out as the problem! Port 5060 is only the authentication port, the rtp stream is normally 10,000 thru 20,000. Some of your phone may have STUN modules on them. Open 10,000 thru 20,000 and 5060 on the firewall. Stick some holes in it for testing purposes. Verify ports are open with telnet:port number both ways, telnet is your friend. If it works, close the holes up and consult your firewall docs Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hadar Pedhazur Sent: Wednesday, April 25, 2007 6:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Audio with SIP to only one provider whenswitching servers I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is configured identically to the old one as well. All of my IAX connections just worked. All but one of my SIP connections just worked as well (which is why I can't believe it's a firewall issue). StanaPhone, which I use for 2 incoming DIDs, registers correctly, and rings my phones correctly when a call comes in. However, once answered, there is dead silence in both directions, on 100% of the calls. There isn't any problem on StanaPhone's side (which has provided a _fantastic_ service ever since I signed up!), because I can connect to them with X-Lite and receive calls with audio. More importantly, if I fire up Asterisk on the old server, it still works!!! I can connect with X-Lite to the new server, so the new server definitely accepts SIP connections, and audio works. It's _not_ a codec problem. I verified that on both the working and non-working servers the connection is established with ulaw on both sides. I have dumped the peer and the channel on both, while the call was active, and they look identical to me, except for the random bits associated with a particular connection. Here are the ones from the machine that fails: *CLI sip show peer XX * Name : XX Secret : Set MD5Secret: Not set Context : default Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : sip.stanaphone.com Addr-IP : 204.147.183.18 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: 12345678 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : OK (20 ms) Useragent: Reg. Contact : new*CLI sip show channel [EMAIL PROTECTED] * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: AAA.BBB.CCC.DDD (local) Our Tag:as360c7ca5 Their Tag: 0bd46ffd48e4fbffb3a68f13f8ad2599 SIP User agent: Username: 87654321 Peername: 12345678 Original uri: sip:204.147.183.55:1024 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=as360c7ca5;lr=on DTMF Mode: rfc2833 SIP Options:(none) Finally, I built 1.2.18 from source today, and everything is working perfectly _except_ for StanaPhone, which continued to connect with no problems, but deliver no audio in either direction. I have no idea what else to try, and would appreciate _any_ guidance. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Marketing 101
Personally, I look for specialty applications. Work smart not hard! I myself am looking for outstanding marketers for a fire hot industry / telecom application. I have all of the correct duckies in a row, just need to send it to the market the correct way. [EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Wednesday, April 25, 2007 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Marketing 101 Agreed. Highly-considered purchases like telco infrastructure are not as much a push as a pull sale. It's about being in the right place at the right time with all the right answers. Almost like buying a home. Since the turnover is SO long with core business process equipment, it's almost a beauty contest when the time comes around. A better analogy would probably be in luxury car buying. You need to look good, have a good feature set, be luxurious to drive, have all the right bells and whistles above and beyond basic requirements, and then of course have a track record of reliability and great service. Just my $.02 -- --- Robert Goodyear Managing Partner Brand Up LLC Knight West 949.542.7001 DIRECT 949.542.7010 FAX 888.272.6387 x501 [EMAIL PROTECTED] [EMAIL PROTECTED] --- On Apr 25, 2007, at 10:52 AM, SIP wrote: Businesses RARELY are in a position to choose new Telco systems providers. Oftentimes, that sort of decision is made by whomever leases them the office space, or was made once back in the beginning, and they've had no real reason to re-evaluate their service/provider. There are, however, plenty of Telco events where the providers hawk their wares and the installers tout their expertise. Cold Call/Networking/Word of Mouth are decent methods of getting your name out there as an alternative, but be prepared to run into a great many situations in which the system or provider they have 'works well enough' so they're not interested in changing. shadowym wrote: Thanks for the advice. Maybe I should clarify what I was asking. It's not so much the how but the what. What are people doing to get PBX Sales/Support business. I know how to get IT business but potential customers still see the Telco business as quite different and are used to using separate companies for that. What I was asking is how the traditional telco guys get new sales/support/consulting business. With IT it's usually a combination of cold call/networking/word of mouth. I'm hoping that Telco is the same but I never see any telco guys at networking events so I am thinking they cold call and advertise targeted at business owners. I'm not sure though. -Original Message- From: dave cantera [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 24, 2007 9:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Marketing 101 shadowym, best thing to do is talk to a lot of consultants, coaches, and marketing people... take the approach you do with learning open source only reverse it... instead of reading source (internal) ask people (external)... it is a big undertaking and the most important task you have... marketing is a bigger task than the technical (for a tech anyway) don't go it alone nothing happens without marketing (and sales)... marketing is *not* sales... daveC shadowym wrote: I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
RE: [asterisk-users] Polycom SP 601 Reboot Issue- Help!
Hard reset the phone first! Provision and see if it is fixed. No? Upgrade software (watch out for provisioning changes). Still rebooting? Downgrade software. Still rebooting? You now have a new door stop! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Garstang Sent: Wednesday, April 25, 2007 7:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help! That used to happen to us _ALL_ the time. Sometimes you'd just have to press the 'Directory' key and the phone would instantly reboot. It was very easy to reproduce and Polycom where useless at admitting it might be a problem. It occurred on several phones. Funnily enough, the phone it was most reproducable on was a 601 being used as a Receptionist phone with 3 sidecars... and about 35 buddies being watched. Hmmm! Russ Beaupre wrote: We had a situation where the 601 base went missing and the electrical connection between the side cars and the 601 was broke. Might be worth a look to see if the phone got damaged. -Original Message- From: Jerry Jones [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 24 Apr 2007 12:27:46 -0500 Subject: Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help! The only reboot issue I have with 1 sidecar is the side car deciding to randonly rebbot, not the phone itself Perhaps upgrading to 2.1 will help? On Apr 24, 2007, at 10:51 AM, J French wrote: I have a Polycom 601 with 3 expansion modules running 2.0.3. We have Buddywatch set up on around 42 users on the expansion modules. We are experiencing reboots on the 601. Today it happened twice after users paged through the phones. The page groups have about 23 phones each. There is a third page group comprising all 46 phones. I'm thinking it may be an issue with changing buddywatch state on so many buddies so quickly. Also, the cpu usage is pegged at 100% for around 3 minutes after it reboots, FWIW. Anyone else experiencing rebbots on the 601? Advice is really needed! Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Random Asterisk deaths
test -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Jensen Sent: Wednesday, April 25, 2007 7:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Random Asterisk deaths Asterisk 1.2.13 (newest available in Debian Etch) No VMs, nothing strange whatsoever about the setup. In the box: TE405P X100P clone AMD Athlon XP 2400+ 512 MB RAM In the logs I get a lot of zt hook failed: Device or resource busy and Avoided initial deadlock for '0xXX', 10 retries! but these happen all the time and don't increase or decrease in frequency around the time that Asterisk dies. On 4/24/07, Bryan M. Johns [EMAIL PROTECTED] wrote: What version are you running? Anything creative like VMs or other unique configurations in use? Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Wayne Jensen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 24, 2007 7:24:26 PM (GMT-0500) America/New_York Subject: [asterisk-users] Random Asterisk deaths Every once in a while for no apparent reason, Asterisk has been dying on me, dropping all calls in progress. There's nothing in the log file or on the Asterisk console that indicates the reason. Some days it doesn't happen at all. Other days it happens two or three times. The problem began on Friday, but the last time anything was changed on that box was at least a week before that. Any suggestions on what to do/where to look to find out what's going on and fix the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)
I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load available. If you log into cisco.com, they have it under software. Sometimes people post it on the internet. Asterisk is supposed to be more skinny friendly these days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Howk Sent: Wednesday, April 25, 2007 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP) From reading the SLA docs, SIP hints are use to get the lights on the phone to show the correct state. I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should behave. It seems apparent that the phones should not register with asterisk, otherwise all the phones will try to register to be THE phone for a given extension. should these lines be built like a trunk/peer? if I could be an example of how lines for SLA should look in sip.conf, that would be helpful. Also I'm somewhat annoyed that I have to compile zaptel drivers that I don't use in order to compile the app_meetme.so module so I can have the SLA functions available to the dialplan... Any feedback is greatly appreciated! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help making a voice record server $$$
Hey there folks, Looking to my favorite mailing list for assistance and have a few bucks to pay you for your time. Me: Played with asterisk for a while in the early days and getting stuck on silly stuff on a time sensitive project for a friend. Project: PSTN incoming call to asterisk and then back to PSTN again, asterisk will hold and record the RTP stream. Upon disconnect, asterisk will name the record file by CID and Date. That's it! E mail me with how much you want for your time and this will surely grow into other project that are later going to be implemented on this server. Sincerely, Brad [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Question about DSP in Digium card
Whether it is IAX, SIP, H323 or ? These are authentication handshakes to establish an rtp stream. SIP = user name and password in a standardized IP packet IAX = same H.323 = same Is also has to do with what codec are supported as well. As far as NAT is concerned! Yep, tell your ISP to forward the authentication port or just junk their gear and get something like a low end Cisco. Or Get IP Phones with STUN (a little pricey) Or Trick Use some type of tunneling gear to an outside IP (outside your NAT) and then bounce your authentication from this new gateway!!! i.e. establish a VPN connection to an outside router from an internal router and drive the call through there. Brad _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A. Levy Sent: Tuesday, March 27, 2007 6:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Question about DSP in Digium card well, ...,we did not choose SIP because our customers are located behind NAT router (using private IP's) and those routers are not managed by them but by the ISP so it is very difficult to establish full duplex phone calls because you can not initiate voice over ip session from the internet (outside) to LAN side (inside) with private IP's. We could not establish 2-way phone calls, I mean, the conversation is listened in 1-way only. As I mentioned before, we can not configure PAT into the NAT router neither because is handled by the ISP and the passwords are unknown That's why we decided to use IAX instead of SIP, I mean, IAX is more robust than SIP when the NAT router is 3th-party managed and the PAT feature is not enable. On the other and we tested IAX over dialup links and it worked fine Those are the reasons we choose IAX as acess protocol to our SIP/H323 Network. You know, the access networks of the customers are different completely: Private IP Address over DSL lines (NAT Router), Public IP Address over DSL lines, Corporate Networks over dedicated Links (Public and IP Addresses), Dialup links, .. Any comment would be welcomed, thanks a lot Levy.- 2007/3/24, A. Levy [EMAIL PROTECTED]: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Refresher course needed!
Hello everyone
My name is Brad, I am an old Asterisk Vet of the very early days just coming
back to join the group.
Ok, for starters, I feel like the monkey with the light bulb looking at
extensions.conf and sip.conf.
It has been some time.
A friend ask me to set up a asterisk server that records phone calls.
FC4
Asterisk 1.4
And all the latest and greatest
Problem number 1
Some good get back into the grove literature.
I work CLI only, never much for graphics and gui's
Problem number 2
We have asterisk logged into teliax but cannot see the inbound call come up
on the CLI
Tethereal says this;
1660 3.829799 207.174.202.4 - 66.109.17.92 SIP Status: 100 Trying(1
bindings)
1661 3.831357 207.174.202.4 - 66.109.17.92 SIP Status: 200 OK(1
bindings)
Asterisk says this;
*CLI
Nothing, notta!
My extensions.conf
(yes, I loaded the samples)
[general]
static=yes
writeprotect=no
clearglobalvars=no
;#include filename.conf
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;From here is brads stuff
exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten = YOURNUMBER,1,Answer()
exten = YOURNUMBER,1,DIAL(SIP/user,20)
Thanks to all!
Brad
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[asterisk-users] Zaptel silly issue
I am geet this error, I assume because I have zero digium hardware installed. This is to be an entirely web based PBX. Can anyone point me to an easy 123 for installing zaptel in dummy form? I need music on hold for a VPS server. Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, March 19, 2007 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Configuring Faxs any help :) younss azzayani wrote: Hi everybody, after installing hylafax iaxmodem i get this email == The HylaFAX software thinks that there is a problem with the modem on device /dev/ttyIAX that needs attention; repeated attempts to initialize the modem have failed. This would be better off on the HylaFAX+ mailing list. Please, when posting there, include your configuration files. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
