RE: [Asterisk-Users] Forwarding calls
Any and all help is appreciated at this point. Thanks for the tip. This is the only thing I have not been able to get working and ironically it is the most important. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mitchel Constantin Sent: Wednesday, March 30, 2005 01:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Forwarding calls I think from what I remember you have to use agi to do this, so you can send the command once the call is bridged. I don't know how off the top of my head though but I do think this is the route to look at. mitchel On Wed, 30 Mar 2005 01:48:14 -0600, Paul [EMAIL PROTECTED] wrote: I have setup the menu system, it works fine, but I can't get it to forward the call to another outside number. The sites you gave me are on setting up the IVR. Any thoughts? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Wednesday, March 30, 2005 00:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Forwarding calls connected to one of them. Basically my goal is to have someone call into the incoming POTS line and be presented with a menu where they would select an exten = 1,2,Goto,cellphone|s|1 Nice try, but take a look here: http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu or here http://users.pandora.be/Asterisk-PBX/IVR.htm or here http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/d ocs-html/x720.html all of which were found using google interactive voice menu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Installation
Title: Asterisk Installation http://www.voip-info.org/tiki-index.php All you will need is a network card in each system. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Tuesday, March 29, 2005 11:54 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk Installation Dear User I am new to the ASTERISK not even tried to install it yet just want to know can I user ASTERISK as a VOIP without any Hardware DIGIUM Card I just want to install ASTERISK as a IAX between two office, And can you suggest me a document for NEW person like me. Best Regards Vipul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPSwitchBoard Version 0.71 Released
Version 0.71 - 30. march 2005. . Fixed a memory leak, and optimized performance drastically. Download from here: http://ipswitchBoard.thorben.dk IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET application which gives you: Unattended/attended transfers. Park calls and retrieve/forward them again. Organize all your SIP and IAX extensions (automatically retreived from Asterisk). Monitor all extensions. Monitor all queues. Monitor Agents. Monitor Parked Calls. Dynamically log extensions in and out of queues. Integration with CRM software on the web. Drop any active call. Import/Export extensions to/from Asterisk Server DB. Set Do Not Disturb on Extensions and give a reason. Speed Dialling. Share Speed Dial files among all users of IPSwitchBoard. User selectable ring tones for IPSwitchBoard. User selectable button colors. Regards Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Installation
Look at www.voip-info.org, you'll have a lot of answers there. Yves [EMAIL PROTECTED] wrote: Dear User I am new to the ASTERISK not even tried to install it yet just want to know can I user ASTERISK as a VOIP without any Hardware DIGIUM Card I just want to install ASTERISK as a IAX between two office, And can you suggest me a document for NEW person like me. Best Regards Vipul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
_ Emotikony a pozadi programu MSN Messenger ozivi vasi konverzaci. http://messenger.msn.cz/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comprehensive Asterisk Load Testing
Hello, I would like to test the capabilities of the various hardware that I run AstLinux on: - Soekris Net4801 (266mhz Geode) - 1ghz P3 - 1ghz Via C3 - 2.5ghz Celeron - 3 ghz x 2 Xeon What I would like to do is use * on the higher end machines to pound as many calls as possible (probably 10, 20 at a time) into * on the lesser machines. I will then try to keep track of system resources (CPU usage, memory usage, etc) on the client machines. I want to do this with various codecs, jitterbuffer yes/no, trunk yes/no, SIP, IAX, across all of this hardware to at least get an idea of what I can expect from these CPU's (as far as transcoding goes). show translations is just not cutting it anymore... :) Not to self-promote, but AstLinux looks like a perfect platform to do testing like this because of consistency and the fact that it can run from flash and RAM, so disk I/O should not ever be a problem... I am thinking some combination of app_milliwatt the outgoing call spool or manager interface would be a good way to go about this. The wiki page has no specifics for doing this, so I thought I would ask. How is this normally done, or is there a completely different, better way to do it? Thanks in advance! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comprehensive Asterisk Load Testing
Hey, Most of the time you dont need a big machine to test a small machine. Just make sure there is no transcoding on the sending end. I did all the tests you mentioned (Except for the jitter buffer) on a dual xeon and a via c3. That took me about 2 months fulltime (its a lot harder than it looks), you can find some of the results on www.astertest.com (there you will find also some imature version of a callgenerator for asterisk that would probably help you to do things faster). I could also help you off list if you want. Zoa. Kristian Kielhofner wrote: Hello, I would like to test the capabilities of the various hardware that I run AstLinux on: - Soekris Net4801 (266mhz Geode) - 1ghz P3 - 1ghz Via C3 - 2.5ghz Celeron - 3 ghz x 2 Xeon What I would like to do is use * on the higher end machines to pound as many calls as possible (probably 10, 20 at a time) into * on the lesser machines. I will then try to keep track of system resources (CPU usage, memory usage, etc) on the client machines. I want to do this with various codecs, jitterbuffer yes/no, trunk yes/no, SIP, IAX, across all of this hardware to at least get an idea of what I can expect from these CPU's (as far as transcoding goes). show translations is just not cutting it anymore... :) Not to self-promote, but AstLinux looks like a perfect platform to do testing like this because of consistency and the fact that it can run from flash and RAM, so disk I/O should not ever be a problem... I am thinking some combination of app_milliwatt the outgoing call spool or manager interface would be a good way to go about this. The wiki page has no specifics for doing this, so I thought I would ask. How is this normally done, or is there a completely different, better way to do it? Thanks in advance! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX realtime dynamic
Title: Message I am having a similar problem, at least trying to access the dynamic user on a second asterisk machine that pulls from mysql. Are you getting anything in your debug log? I'm using the same layout as the sample sip users table from the wiki, the only difference being I added "auth" and "peercontext" field. I'm stumped on this one, I feel it's a code bug as my logs indicate success with mysql. I've even packet sniffed to watch the action,and still no luck.. Matt -Original Message-From: Wojciech Tryc [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 29, 2005 2:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] IAX realtime dynamic Good Afternoon, I am just playing with realtime on one of my boxes (running obviously HEAD). The voicemail portion works just fine, howevere I am having difficulties getting iax portion to work. Sip and extensions left for later for now. Could anyone send me sample database dump of his/her config? Also, what about the iax.conf should i leave the [general] section? or remove the file completly. Basically, at this point it cannont create any iax channels unless the user name and password exists in extensions.conf. Thanks, Wojtek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 FXO with Asterisk
Ed Greenberg wrote: Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work? There's something about this on the wiki. Dig it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Troubles with VoIP providers
Hello all, I had tested about ten VoIP providers, but no one gave me the quality I was looking for. My calls, depending the hour of the day, have diferent quality. Sometimes I felt cuts in the conversation or lost the sound on one of the end point. All of the providers I tested had any kind of trouble. My internet gateway is an 1 Mb. ADSL conection y I make QOS by the router 70% of bandwidth for SIP and IAX2 protocols and 30% for others protocols. With 3 simultaneus calls. I thing that the problem is in the providers side, cause we make calls between our diferents offices via IAX2 without quality problems, but I am not sure. I said that because when in US the people wake up and start to work, about local time 13:00, our calls get more troubles, like cuts, but before that time our calls goes better than after. Have you got troubles with your provider, or the sound quality is always the same? Am I the only one who feel troubles with VoIP providers? Could you tell me the witch provider you use? Any clue will be welcomed. Thanks for your time Obihuan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source Billing Software
Take a look at http://ebills.sourceforge.net/ I uses latex to create nice pdfs. regards, klaus Christopher Snell wrote: On Tue, 29 Mar 2005 09:53:03 +1000, Rod Bacon [EMAIL PROTECTED] wrote: What I would like to know is has anyone found an open-source billing platform that performs basic billing functionality (pre/post) from RADIUS and/or Asterisk CDR and is written (well-written) in either PHP or PERL. What features and functionality is needed for such a system? I've been thinking about using Perl to write LaTeX source files, which can then be compiled into pretty PostScript and PDF paper bills or plain text that can be sent out by e-mail. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Test Line
Somewhere in the Wiki I read that the best way to adjust the rxgain and txgain is to dial a type 102 milliwatt test line. This line is usually found in xxx-958- or xxx-959- ranges. I'm in area code 323 in Los Angeles. Does anybody know the test number here?? The number assigned to the milliwatt generator is up to each telco; there is no standard other then some larger operating companies will sometimes develop an internal standard. But, each operating telco will oftentimes develop their so called standard with no input from other telcos. In the olden days of electo-mechanical central office switches, the xxx-xx98 and xxx-xx99 numbers were frequently reserved for testing, and in some cases the milliwatt generator was assigned to one of those numbers in the central office. But, a central office with 10,000 lines will have about 100 of those number combinations. Much easier to call your telco 'repair service' and simply ask them for the number of that office's milliwatt generator. If they don't know, ask them to forward you to a central office technician. If that doesn't work, the next time you see a telco truck in the area (or at the coffee shop), ask the driver. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: does Sipura SPA 3000 support UK caller id?
On Tue, 22 Mar 2005 10:45:42 -0800, Trevor Peirce [EMAIL PROTECTED] wrote: Mike Dent wrote: Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? Yes it does Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Egytpian call progress frequencies and cadences (second request)
Hi, Can anyone provide me with the call progress frequencies and cadences for the Egyptian PSTN. I need to make the TDM card zaptel driver to be able to detect busy, ring, dialtone and congestion tones coming from the PSTN. Also correct me if I'm wrong, once I get this information I code it into the zonedata.c file and recompile the zaptel module, right? Ezabi signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio codec MP108 please help
hi all, can any 1 pls tell me the context i shld add on sip.conf for Audiocodec MP108 8 fxs please. i want to add 2 phone on MP108 port assign extention and dial each other, can`t get a dialtone only busy signal. Thnx ppls Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Installation
Thanks Friends -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yves Sent: Wednesday, March 30, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Installation Look at www.voip-info.org, you'll have a lot of answers there. Yves [EMAIL PROTECTED] wrote: Dear User I am new to the ASTERISK not even tried to install it yet just want to know can I user ASTERISK as a VOIP without any Hardware DIGIUM Card I just want to install ASTERISK as a IAX between two office, And can you suggest me a document for NEW person like me. Best Regards Vipul -- -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk GLIB_2.0 Error
Title: Asterisk GLIB_2.0 Error Hello everybody, I'm trying to install spandsp_0.0.2pre11 on Debian with a 2.6.6 kernel. I followed every instruction I could find, and compilation did not produce any errors, but when I start Asterisk I get following message: WARNING[27090]: loader.c:301 __load_resource: /usr/lib/libmysqlclient.so.10: symbol errno, version GLIBC_2.0 not defined in file libc.so.6 with link time reference Mar 30 13:36:19 WARNING[27090]: loader.c:509 load_modules: Loading module res_config_mysql.so failed! And Asterisk stops running. Does anyone have any suggestions to solve this problem? Thanks in advance, Dennie __This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV. _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV. Register for AXS Mail at http://www.secuteam.com! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 nat
Is it possible to have 2 (working) iax2 phones behind port restriced nat? Interesting you ask, since I just had an incident concerning this. I have an IAXy and got an IAX hardphone which I tested at home behind the same NAT. Using IAX soft clients before in this situation, they would work, but the hardphone had a lot of trouble being reachable and was shown at port 1024 while all other peers were listed on the correct 4569. Putting that same hardphone in the DMZ (which I've never gotten to work before with say, SIP phones) made it work normally. It also seems that while the IAXy will work with qualify=300, none of the IAX phones I've tested so far will be reachable if qualify is used. I've wondered why this is, in case anyone has the answer? My guess is that qualify uses a message that these phones don't answer. Whether two or more iax phones work behind a nat is highly dependent on exactly how the nat box handles data flows that initiate on the same udp port. Not all nat boxes function the same. Example: two iax phones behind an inexpensive nat box. Both iax phones use a source and destination port of udp 4569, and each iax phone has its own internal IP address. When internal iax phone #1 contacts an external asterisk box, that udp session will oftentimes use udp 4569 for both the source and destination ports. The packet leaving the nat box will have a source IP address of the external nat interface. When internal iax phone #2 attempts the same thing, the nat box already knows (via its internal tables) that source and destination ports 4569 are in use (with the outside IP address), and will remap the source udp port to something else (eg, 1024 or higher). There are some cheep nat boxes that mess that map process up.) Assuming the nat box mapped these two correctly, both iax phones should be registered. However, both are using udp, and udp is a connectionless protocol. When the nat box maps those ports, it also starts a timer that will be used to time out those table entries. The timeout value can be as small as a minute or two, or as long as no timeout (drop the oldest entries when the table becomes full). If you think about how many times your pc goes to the internet to resolve dns entries (for all pc access, whether its a phone or web surfing), those dns entries (also using udp) will become a rather large number. If the nat box has limited internal memory resources, the manufacturer will likely have a rather small timeout value that could actually be in seconds. Now, what is going to happen to your iax phones when the nat box decides to drop the table map entries? (Ans: no more communications.) Some nat boxes will let you configure the udp map timeout values while others won't even publish their default values. And in some cases, the manufacturer will change their unpublished default value from one version of firmware to another. The 'qualify' statement was intended (as one purpose) to pulse the remote phone and keep the nat table entries from timing out. That usually works just fine if the iax phone uses the register method. If the iax phone does not use the register method (and you have the * iax definitions in terms of 'peer' and 'user'), you're likely to have a nat box problem. Why? Because asterisk will attempt to contact both iax phones by sending udp packets to the same nat address using udp port 4569. The nat box won't know what to do with that pkt. The work around to that is to statically map 4569 to one phone and map 4570 to the second phone (in the nat box). Then in the * config, ensure your dialplan uses the same port numbers to reach each phone. If you've followed along thus far, then what happens when the iax phone sends an arbitrary pkt (of any type) to asterisk? The nat box will likely get in the middle again and map that outgoing pkt to yet another port, and * may become rather confused. Bottom line: when having problems with two or more phones behind a nat box, you almost always have to use a packet sniffer on the inside and outside of the nat box to see what that box is actually doing to you. If the iax phones allow you to select a udp source port range that it will use, then set the range to different values for each phone. E.g., iax phone #1 uses source udp ports 10,000 - 10,100, and phone #2 uses ports 10,200 - 10,300, or something like that. Since I don't use any iax phones, I don't have a clue if any of the common ones provide such an option. You may also find that different iax phones will operate differently using the same nat box. Its not uncommon for programmers to force the use of udp port 4569 for _both_ the source and destination ports. Two instances of that kind of phone will likely cause the nat issues noted above. If a different iax phone allows the source port to be chosen by the system, there is a much smaller chance of having a nat problem. (The small chance results when both phones happen to chose the same source port and
Re: [Asterisk-Users] Help Debugging my code?
do you really have
[specialized]
[specialized]
it is twice try removing one entry
Jason
On Wed, 23 Mar 2005 02:37:42 +, Scheda [EMAIL PROTECTED] wrote:
Hey, I'm currently using the GotoIf application to set it so if
certain caller ID's call my number, it will transfer it to my cell
phone, here is the code I have so far. I get an error message that
states call rejected by 198.22.67.70: No such context/extention.
when I call the number from my house number. Anyway, here is the code
I have.
[inbound]
exten = 8667393960,1,Answer()
exten = 8667393960,2,GotoIf($[${CALLERIDNUM} =
${house}]?specialized,8667393960,1:2)
exten = 8667393960,3,GotoIf($[${CALLERIDNUM} =
${kendra}]?specialized,8667393960,1:2)
exten = 8667393960,4,GotoIf($[${CALLERIDNUM} =
${rob}]?specialized,8667393960,1:2)
exten = 8667393960,5,GotoIf($[${CALLERIDNUM} =
${jen}]?specialized,8667393960,1:2)
exten = 8667393960,6,GotoIf($[${CALLERIDNUM} =
${mom}]?specialized,8667393960,1:2)
exten = 8667393960,7,GotoIf($[${CALLERIDNUM} =
${dad}]?specialized,8667393960,1:2)
exten = 8667393960,8,Wait(3)
exten = 8667393960,9,Background(/root/asterisk-1.0.6/sounds/ast-intro)
exten = 8667393960,10,Wait(12)
exten = 8667393960,11,Hangup()
[specialized]
[specialized]
exten = 8667393960,1,SetCallerID(${cid})
exten = 8667393960,2,Wait(1)
exten = 8667393960,3,SetMusicOnHold(danecook)
exten = 8667393960,4,Dial(${TRUNK}/${scheda},35,t)
exten = 8667393960,5,Hangup()
I have all the global variables set up correctly, so I'm not sure what
it is exactly
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Re: [Asterisk-Users] Implant GIPS's codec to Asterisk
El 30/03/2005, a las 7:40, Dominic Lu escribió: Hello, If purchase the codec from GIPS, how difficult it is to implant it in Asterisk? What the cost will be? Our company has two Asterisk, one in headquarter and the other in branch office. We only need the communication between them. We are not satisfied with current codec either in bandwidth usage or voice quality. Since Skype really impress us in voice quality, so this kind of idea is generated. BR, Dominic You are talkina about GIPS ILBC ? http://www.globalipsound.com/products/iLBCfreeware.phpThere is a free ILBC codec http://www.ilbcfreeware.org/ ILBC is compiled by default by asterisk. My friends usually say sound quality in asterisk is better than Skype one, i've heard that Skype use a modified version of ILBC ·· Adrià Vidal ... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Cannot answer
MDS wrote: I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6 months, no problems with my grandstreams. I'm fairly familiar with the ins and outs of asterisk... If you are going to use CVS HEAD, you _must_ stay up to date. There have been a large number of SIP-related fixes in CVS HEAD since the 19th. Please update your system and try again before reporting problems. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with 401 Unauthorized
Mike Miller wrote: They're both running on 192.168.1.100 Sorry -- I probably should've clarified that. Yeah. that would have helped! For some reason, they were not only running on the same machine, but sharing the same port number, which shouldn't really be possible... But in any case, if you want to run Linphone (or any softphone) on the same box as Asterisk, you need to configure the softphone to use a different port for SIP than 5060, and different port for RTP than 1-2 (the Asterisk default). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i transfer/forward a call out?
[cellphone] exten = s,1,Flash exten = s,2,Dial,Zap/2/9729796243 exten = s,4,Congestion I never done this, but I believe you are missing a final part. If you do the same thing on a regular phone, the scenario would be this : 1- you are connected with the remote person 2- you hit Flash and get a dialtone 3- you dial the number you want to reach and get connected 4- you hangup the line and both remote are connected to eachother so you need to change your dialplan like this [cellphone] exten = s,1,Flash exten = s,2,Dial,Zap/2/9729796243 exten = s,3,Hangup I never used this so maybe I miss a step, maybe you have to flash again before you do the hangup. Can't test it as I don't have 3-way calling on my phoneline b.t.w.: You had a wrong sequence in there, 1-2-4... missing the 3 hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implant GIPS's codec to Asterisk
Skype uses the wideband version of iLBC. I beleive that this wod be very interesting in *, but I've also read that the wideband version is not freely available for use. Michael On Wed, 30 Mar 2005 13:49:48 +0200, adria vidal wrote: El 30/03/2005, a las 7:40, Dominic Lu escribió: Hello, If purchase the codec from GIPS, how difficult it is to implant it in Asterisk? What the cost will be? Our company has two Asterisk, one in headquarter and the other in branch office. We only need the communication between them. We are not satisfied with current codec either in bandwidth usage or voice quality. Since Skype really impress us in voice quality, so this kind of idea is generated. BR, Dominic You are talkina about GIPS ILBC ? http://www.globalipsound.com/products/iLBCfreeware.phpThere is a free ILBC codec http://www.ilbcfreeware.org/ ILBC is compiled by default by asterisk. My friends usually say sound quality in asterisk is better than Skype one, i've heard that Skype use a modified version of ILBC ·· Adrià Vidal ... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Troubles with VoIP providers
On March 30, 2005 05:24 am, Obihuan wrote: My calls, depending the hour of the day, have diferent quality. Sometimes I felt cuts in the conversation or lost the sound on one of the end point. All of the providers I tested had any kind of trouble. Sounds like the trouble is on your end then. I use nufone almost exclusively and put about 5000 minutes a month through them, with multiple simultaneous calls (mid-size business) and while I occassionally have some audio problems, I have never had issue with nufone's network. I have been able to (in my mind anyway) prove that the connectivity issue was on my end, as when the problem occurs it occurs with any provider I happen to be using, and they all take wildly different paths once it leaves my (decently connected) internet provider. My internet gateway is an 1 Mb. ADSL conection y I make QOS by the router 70% of bandwidth for SIP and IAX2 protocols and 30% for others protocols. With 3 simultaneus calls. I thing that the problem is in the providers side, cause we make calls between our diferents offices via IAX2 without quality problems, but I am not sure. I said that because when in US the people wake up and start to work, about local time 13:00, our calls get more troubles, like cuts, but before that time our calls goes better than after. Is there any heavy downloading or uploading going on around that time? The unix program 'rate' or even tcpdump or ethereal should be able ot help you determine this. Remember that you can only rate-limit your OUTGOING traffic. Traffic headed for you can be dropped in an attempt for tcp's automatic backoff to slow down the connection, but as the name implies it only works for TCP. Feel free to try my traffic control script: http://www.mixdown.ca/~andrew/dump/rc.tc -- it runs on our upstream router and with it I am able to keep our connection loaded but still have voice traffic pass through as top priority. Again, it tries to limit the incoming traffic but that's more based on luck than anything else. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD queue question
That's what I thought would happen, but after about an hour and 100 or so incoming calls, it was still ringing the agents in the order that they were listed in the agents.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Tuesday, March 29, 2005 10:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ACD queue question The first call for each agent probably goes that way, but then after a few calls have rolled through the queue, the strategy you specify (like LeastRecent) should come into play. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Tuesday, March 29, 2005 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ACD queue question I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Solaris install from HEAD
Hi, I am trying to compile Asterisk on Solaris. I have tried on a number of different platfroms, Solaris 8 on sparc and Solaris 10 on X86 and have run into a number of problems. The voip-info wiki talks about working installs, but I am not having much luck. Environment: gcc 3.4 gmake ginstall Any help would be appreciated. Regards, -Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem on outgoing calls (quadbri card and bristuffed Asterisk latest) ?
Hi, I have strange behavior on outgoing calls (I can receive calls and I can make outgoing calls to ISDN lines ok (035778421 and 5778421 for instance - 03 is area code). I use latest bristuffed Ast. under Suse 9.2. My zapata.conf and zaptel.conf are at the end of mail. Any help, advice - I guess there is something wrong with settings... But when I call my cellular on 041 461 620 - exactly as I type on phone, I get this : -- Executing Dial(IAX2/[EMAIL PROTECTED]/6, ZAP/g1/041461620|60) in new stack -- Called g1/041461620 -- Zap/1-1 is making progress passing it to IAX2/[EMAIL PROTECTED]/6 -- Channel 0/1, span 1 got hangup Mar 30 14:40:10 WARNING[9744]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Hangup(IAX2/[EMAIL PROTECTED]/6, ) in new stack == Spawn extension (from-internal, 041461620, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]/6' -- Hungup 'IAX2/[EMAIL PROTECTED]/6' and under debug span 1: [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 133/0x85) (Terminator) Message type: CALL PROCEEDING (2) -- Zap/1-1 is making progress passing it to IAX2/[EMAIL PROTECTED]/4 Protocol Discriminator: Q.931 (8) len=23 Call Ref: len= 1 (reference 133/0x85) (Terminator) Message type: DISCONNECT (69) [08 02 82 83] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: No route to destination (3), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] [28 09 4e 4f 20 52 4f 55 54 45 20] Display (len= 9) [ NO ROUTE ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 40 (cs0, Display) -- Channel 0/1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 5/0x5) (Originator) Message type: RELEASE (77) [08 02 81 83] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: No route to destination (3), class = Normal Event (0) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Hangup(IAX2/[EMAIL PROTECTED]/4, ) in new stack == Spawn extension (from-internal, 041461620, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]/4' -- Hungup 'IAX2/[EMAIL PROTECTED]/4' Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 133/0x85) (Terminator) Message type: RELEASE COMPLETE (90) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null **/etc/zaptel.conf loadzone=nl defaultzone=nl # qozap span definitions # most of the values should be bogus because we are not really zaptel #span=1,1,3,ccs,hdb3 #span=2,0,3,ccs,hdb3 #span=3,0,3,ccs,hdb3 #span=4,0,3,ccs,hdb3 span=1,1,3,ccs,ami,crc4 span=2,0,3,ccs,ami,crc4 span=3,0,3,ccs,ami,crc4 span=4,0,3,ccs,ami,crc4 bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 **/etc/asterisk/zapata.conf [channels] switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 ;callerid=asreceived overlapdial=yes ;--- ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp context=isdn-incoming group = 1 ; S/T port 1-3 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 ;channel = 4-5 ;channel = 7-8 ;--- ; p2p NT mode (for connecting an ISDN PBX in point-to-point mode) signalling = bri_net context=pbx-incoming group = 2
Please do not use 'reply' for new threads? (was: Re: [Asterisk-Users] Egytpian call progress frequencies and cadences (second request))
On Wed, March 30, 2005 13:04, Ezabi said: Hi, SNIP Ezabi Ezabi, (a.o.) I am assuming that you aren't using a threaded email reader, as you would be aware of what replying to a message in order to start a new thread - which I am assuming you did, judging from the results - would do to the threading if you were... (I am not holding that against you G, just drawing conclusions!) Please (to all) start a *new* message when you want to start a new thread... Replying to an existing message and changing the subject will *not* start a new thread. Threading (in proper clients at least) is based on special information in the message headers, which does not get altered by changing the subject. The result is that the 'new' thread gets weaved in to the existing one, with unwanted results for both threads... TIA! PS: Please, no replies on whether this should be done on or off list. I happen to think it belongs on-list for the education of all. A 'discussion' on this subject will only server to pollute both threads even further, but will only end, as previously, in agreement to disagree... ;-) BRgds -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Comprehensive Asterisk Load Testing
[EMAIL PROTECTED] wrote: Hey, Most of the time you dont need a big machine to test a small machine. Just make sure there is no transcoding on the sending end. I did all the tests you mentioned (Except for the jitter buffer) on a dual xeon and a via c3. That took me about 2 months fulltime (its a lot harder than it looks), you can find some of the results on www.astertest.com (there you will find also some imature version of a callgenerator for asterisk that would probably help you to do things faster). I could also help you off list if you want. Zoa.. Have you done the test using call generator on test or production boxes? Ta Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD queue question
Using which strategy? Remember, if you change strategies and reload, it'll forget where it was and start over. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Wednesday, March 30, 2005 6:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ACD queue question That's what I thought would happen, but after about an hour and 100 or so incoming calls, it was still ringing the agents in the order that they were listed in the agents.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Tuesday, March 29, 2005 10:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ACD queue question The first call for each agent probably goes that way, but then after a few calls have rolled through the queue, the strategy you specify (like LeastRecent) should come into play. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Tuesday, March 29, 2005 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ACD queue question I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk::AGI script won't work?
I installed the AGI perl library then put the
following script in a file called
/var/lib/asterisk/agi-bin/send_clid.agi,
updated my [incoming] context with exten =
s,1,AGI(send_clid.agi) and did a restart now.
use Asterisk::AGI;
my $agi = Asterisk::AGI-new();
my %input = $agi-ReadParse();
my $clid = $input{callerid};
my $dnid = $input{dnid};
open(CS, call_id_test);
print CS INCOMING CALL FROM . $clid . \n;
print CS $dnid . \n;
close(CS) || die can't close\n;
system(wall $clid);
The cli seems to indicate it worked:
Launched agi script
/var/lib/asterisk/agi-bin/send_clid.agi
AGI script send_clid.agi completed, returning 0
however I see no output from wall and if I do a cat
call_id_test it's empty. call_id_test has permission
set to 777.
Any idea what I'm doing wrong?
Thanks, again for all the help thus far.
Richard
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[Asterisk-Users] Bristuff and startup scripts
Hi, I'm not the kind of Linux guru and I was wondering how I could start automatically the Zaphfc script. What I mean is that before starting asterisk, I have to type : make load from the zaphfc directory in order to load the zaptel driver. How can I do that automatically. This can be very useful in case of unattended reboot, Thanks Best regards David Masure ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ext matching problems
On Mon, 21 Mar 2005 15:03:14 -0400, Francisco Moreno [EMAIL PROTECTED] wrote: Now, when I dial from any of the ext. to '0' It actually matches the '0', plays the goodbye message, but doesn't hangup but gets directly to the 'pasvalide' context. Same thing happens when I dial to the ext. 1002 (the one that doesn't have voicemail), either it rings further than 10secs or it's busy, it does not hangup but gets straight to the 'pasvalide' context. As far as I understood, it should not happen, it should go through the dialplan leaving those context included at the end and in the orther they are included. your pavalide context is the problem [pasvalide] exten = _.,1,Answer() exten = _.,2,Playback(invalid) exten = _.,3,Playback(goodbye) exten = _.,4,Hangup() _. matches all numbers including h which means hangup, change pasvalide to this [pasvalide] exten = _X.,1,Answer() exten = _X.,2,Playback(invalid) exten = _X.,3,Playback(goodbye) exten = _X.,4,Hangup() And all should be good Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk @ home
Hi, What happened to asterisk @ home 0.7 that the dialout-default macro no longer works? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is ZAP ? newbie question sorry
Hi Pros, Please advice whats the purpose of ZAP, if i have softphones and ATA 186 with PSTN trunk, wht ZAP will do ? do i zap to route calls internal softphone to softphones ? thnx a lot Ronny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No D-channels available!
I checked and checked and. When there was no hope left. I found out that my PSTN provider had removed the crc4 without telling. Everything works just fine... Thanx for the help. Rikard -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard Sent: den 29 mars 2005 16:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No D-channels available! On Tuesday 29 March 2005 14:40, Rikard Westlund wrote: Nope! that I have checked. 1. Double check 2. Change the D channel to be 24 and retry 3. Cycle all channels through all possibilities. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard Sent: den 29 mars 2005 15:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No D-channels available! On Tuesday 29 March 2005 14:08, Rikard Westlund wrote: [...] When I start Asterisk(asterisk -vc) I get this: Mar 29 15:02:15 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down [...] I'll hazzard a guess and say you have the card jumpered for T1 instead of E1. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implant GIPS's codec to Asterisk
Dominic Lu wrote: Hello, If purchase the codec from GIPS, how difficult it is to implant it in Asterisk? What the cost will be? Our company has two Asterisk, one in headquarter and the other in branch office. We only need the communication between them. We are not satisfied with current codec either in bandwidth usage or voice quality. Since Skype really impress us in voice quality, so this kind of idea is generated. BR, Dominic The narrow band codec from GIPS (narrowband iLBC) is free to use, and already included in Asterisk. Therefore, I assume you are talking about the wideband GIPS codec (wideband iLBC) which must be purchased, and which Skype uses. The key issue is that right now Asterisk is rather narrowband oriented. So far, I don't think any work has been done to make it work with wideband codecs. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Cannot answer - SOLVED
SOLVED! By updating my CVS head just now, my Polycom IP 600 works great! Thank you! Mark MDS wrote: I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6 months, no problems with my grandstreams. I'm fairly familiar with the ins and outs of asterisk... From: Kevin P. Fleming If you are going to use CVS HEAD, you _must_ stay up to date. There have been a large number of SIP-related fixes in CVS HEAD since the 19th. Please update your system and try again before reporting problems. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Cannot answer
I don't think you want both dynamic and defaultip set But that should not cause what you describe. I hvae seen other issues with head. Perhaps checkout the latest? On Mar 30, 2005, at 12:29 AM, MDS wrote: I googled and googled but could not find anything regarding this problem. I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6 months, no problems with my grandstreams. I'm fairly familiar with the ins and outs of asterisk... IP600 with latest sip 1.4.1 and bootrom from my FTP server. Standard config files from http://www.freedomphones.net/polycom/files/ No changes other than typical ip address of phone and server. Grandstream (192.168.2.20) is exten 2000, Polycom (192.168.2.22) is 2006. I can make calls out to my Grandstreams from the Polycom all day. No problem. When I try to call the Polycom I get this stuff: -- Executing Dial(SIP/2000-972f, SIP/2006|10|r) in new stack -- Called 2006 -- SIP/2006-f8ea is ringing -- SIP/2006-f8ea answered SIP/2000-972f -- Attempting native bridge of SIP/2000-972f and SIP/2006-f8ea -- Got SIP response 481 No Such Call back from 192.168.2.20 == Spawn extension (from-sip, 2006, 1) exited non-zero on 'SIP/2000-972f' -- Got SIP response 500 Internal Server Error back from 192.168.2.22 -- Got SIP response 500 Internal Server Error back from 192.168.2.22 When I answer the polycom it just hangs up and hangs the grandstream online. I have to manually hang up the grandstream. It doesn't get a SIP notifcation of call failure or hangup. When I tcpdump the asterisk box, I can see RTP streams from the Grandstream toward the server. But nothing coming from or toward the Polycom. When I call the Grandstream from the Polycom, the call connects and I see both RTP streams to and from the Asterisk box for both phones and everything is happy. anyone have any ideas as to why inbound calls fail? I've tried several combinations of friend/peer/progressinband/canreinvite etc... No change at all. Here's my sip.conf for the Polycom [2006] type=friend username=2006 secret=2006 host=dynamic dtmfmode=rfc2833 defaultip=192.168.2.22 progressinband=no context=from-sip [EMAIL PROTECTED] callgroup=1 pickupgroup=1 thank you for any insight! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is ZAP ? newbie question sorry
ZAP are channels which connect through hardware boards installed within the * server. If only using softphones feel free to not use;-) On Mar 30, 2005, at 7:32 AM, iMRAN wrote: Hi Pros, Please advice whats the purpose of ZAP, if i have softphones and ATA 186 with PSTN trunk, wht ZAP will do ? do i zap to route calls internal softphone to softphones ? thnx a lot Ronny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Giving sip users multiple contexts
How would I go about giving sip users multiple contexts? For instance right now I have them all in: from-sip-internal Is there a way I can (for sip users) also include say my [dial-911] [dial-local] and [dial-longdistance].. bearing in mind that I want to have different sips allowed to do different things so I can't just do includes for those in my from-sip-internal. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk @ home
On Wed, 30 Mar 2005 08:29:39 -0500 Matt [EMAIL PROTECTED] wrote: Hi, What happened to asterisk @ home 0.7 that the dialout-default macro no longer works? ___ EVERYONE This is NOT the [EMAIL PROTECTED] list group. Please go to: http://sourceforge.net/forum/?group_id=123387 To get help for [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Combatting echo in VOIP
Hey Chris, What type of phone are you using for testing? I found a big difference when I switched from a cheap testset to a better phone. The only problems I get with voipjet is when people talk over each other - but I'm not sure how to fix that but everything else has been very good. J ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk @ home
Sure Robert, but you are going to get a lot of cross posted questions for certain topics (though I agree this one was totally about [EMAIL PROTECTED] so should have gone to the sourceforge forum). Post here if it is a straight asterisk question but on sourceforge http://sourceforge.net/forum/?group_id=123387 if it is an [EMAIL PROTECTED] question or on https://sourceforge.net/forum/?group_id=121515 if it is a AMP related question. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Wednesday, March 30, 2005 9:34 AM To: Matt; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk @ home On Wed, 30 Mar 2005 08:29:39 -0500 Matt [EMAIL PROTECTED] wrote: Hi, What happened to asterisk @ home 0.7 that the dialout-default macro no longer works? ___ EVERYONE This is NOT the [EMAIL PROTECTED] list group. Please go to: http://sourceforge.net/forum/?group_id=123387 To get help for [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Implant GIPS's codec to Asterisk
Hi everybody, GIPS have different products, not only codecs: * Voice enhancements: packet loss concealment algorithms, noise concealment, jitter buffer, agc, aec (can be used with any codec) * Codecs: iLbc (free), ISAC, G711 Wideband... You can include in asterisk voice enhancements and use them iLBC for example, for increasing the quality mainly in face of the packet loss, without using wideband codecs. I'm not a GIPS employee :-), you can view more information in the GIPS website. G. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Steve Underwood Enviado el: miércoles, 30 de marzo de 2005 15:35 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Implant GIPS's codec to Asterisk Dominic Lu wrote: Hello, If purchase the codec from GIPS, how difficult it is to implant it in Asterisk? What the cost will be? Our company has two Asterisk, one in headquarter and the other in branch office. We only need the communication between them. We are not satisfied with current codec either in bandwidth usage or voice quality. Since Skype really impress us in voice quality, so this kind of idea is generated. BR, Dominic The narrow band codec from GIPS (narrowband iLBC) is free to use, and already included in Asterisk. Therefore, I assume you are talking about the wideband GIPS codec (wideband iLBC) which must be purchased, and which Skype uses. The key issue is that right now Asterisk is rather narrowband oriented. So far, I don't think any work has been done to make it work with wideband codecs. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comprehensive Asterisk Load Testing
Its a very very bad idea to do this on production boxes. Especially if you are trying to see how far you can go, and then you cross that tiny border :) Your production calls will not like an idle cpu% of 0% and a load of 500. zoa, Bicom Systems wrote: [EMAIL PROTECTED] wrote: Hey, Most of the time you dont need a big machine to test a small machine. Just make sure there is no transcoding on the sending end. I did all the tests you mentioned (Except for the jitter buffer) on a dual xeon and a via c3. That took me about 2 months fulltime (its a lot harder than it looks), you can find some of the results on www.astertest.com (there you will find also some imature version of a callgenerator for asterisk that would probably help you to do things faster). I could also help you off list if you want. Zoa.. Have you done the test using call generator on test or production boxes? Ta Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk::AGI script won't work?
Richard Reina wrote:
I installed the AGI perl library then put the
following script in a file called
/var/lib/asterisk/agi-bin/send_clid.agi,
updated my [incoming] context with exten =
s,1,AGI(send_clid.agi) and did a restart now.
use Asterisk::AGI;
my $agi = Asterisk::AGI-new();
my %input = $agi-ReadParse();
my $clid = $input{callerid};
my $dnid = $input{dnid};
1st rule with Perl scripts: use strict;
2nd rule: use warnings;
Then.
Are you sure about the capitalization?
I.e. if the variable is ${CALLERID} in asterisk,
You should use $agi-get_variable ('CALLERID') I think.
open(CS, call_id_test);
You should use an absolute path, i.e.
/tmp/call_id_test
print CS INCOMING CALL FROM . $clid . \n;
print CS $dnid . \n;
close(CS) || die can't close\n;
system(wall $clid);
On my system, wall takes input from STDIN. So
open FP, |wall;
print FP CLID: $clid;
close FP;
Might work better.
Regards,
Jean-Michel.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---
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Re: [Asterisk-Users] Fun with CAPI
On Thu, 24 Mar 2005 14:19:20 +, Gavin Hamill [EMAIL PROTECTED] wrote: Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension numbers.. get a dialtone and then dial onward from there... use show application disa in the cli and send them there rather than playing dial tone, this should do what you want Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] job offer - in german only
Produktentwickler(-in) in Java Ihre Aufgabe: Produktentwicklung in Java Anbindung an eine Datenbank (SQL) Asterisk und VoIP Affinität Ihr Anforderungsprofil: Mehrjährige Berufserfahrung in der (Java-) Softwareentwicklung Erfahrungen in der Umsetzung browserbasierter Anwendungen Erfahrung in der Java-Entwicklung mit Applikationsservern Erfahrung mit SQL-Datenbanken (Postgres, Oracle, SQL-Server) Ausgeprägte Teamfähigkeit Ihre Zukunft: Sie sind massgeblich an der Einführung und Erweiterung eines Softwareprodukts beteiligt Sie arbeiten in einem jungen dynamischen Umfeld Sie finden ein flexibles Arbeitszeitmodell vor In unserem Unternehmen erwartet Sie gute und kreative Teamarbeit und Teamgeist Interesse? Ihre aussagefähigen Bewerbungsunterlagen senden Sie bitte per E-mail an [EMAIL PROTECTED]: Easywe GmbH Ralf Ziegler Ettlingerstr 5a 76137 Karlsruhe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving sip users multiple contexts
Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc... Then in that context, include the features you'd like for each group, and give each sip user the correct context. Julian J. M. On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote: How would I go about giving sip users multiple contexts? For instance right now I have them all in: from-sip-internal Is there a way I can (for sip users) also include say my [dial-911] [dial-local] and [dial-longdistance].. bearing in mind that I want to have different sips allowed to do different things so I can't just do includes for those in my from-sip-internal. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving sip users multiple contexts
Well, I thought about that but wanted to check to see if there was another way... at the moment I have: 911 calling local calling international calling long distance. That's only 4.. but there are various combinations 911 and local... 911 and local and long distance... 911 and international (no long distnace)... I guess I can make up seperate contexts.. but it would be helpful if I could just delve the sips directly into a context. On Wed, 30 Mar 2005 15:49:16 +0100, Julian J. M. [EMAIL PROTECTED] wrote: Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc... Then in that context, include the features you'd like for each group, and give each sip user the correct context. Julian J. M. On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote: How would I go about giving sip users multiple contexts? For instance right now I have them all in: from-sip-internal Is there a way I can (for sip users) also include say my [dial-911] [dial-local] and [dial-longdistance].. bearing in mind that I want to have different sips allowed to do different things so I can't just do includes for those in my from-sip-internal. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: How to configure h323 channel driver ?
Hi, ALL:
I has installed my chan_h323 channel driver in my *.
my scenario is:
SIP UA = SER(mediaproxy) = Asterisk = chan_h323 = GNUGK = H323 EP
And my UA and EP all support codecs such as alaw ulaw G.729 at least.
I dial from UA behind NAT to H323 EP, and I answer from H323 EP too.
But I can not hear any voice from each side. Can anybody point out why it is?
h323.conf
--
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
accountcode = myaccountname
gatekeeper = IP of GNUGK
AllowGKRouted = yes
amaflags=default
type=h323
prefix=888248
e164=8881238
context=voip323
disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw
allow=g723.1
extensions.conf
--
[general]
static=yes
writeprotect=no
[globals]
[default]
exten = _.,1,Dial(H323/${EXTEN})
--
Best Regards
Charles
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Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote: How do you toggle the realtime cache? Check in the configs/iax.conf.sample file of a recent CVS download and it should be in there. If there were too many fields in the table, could you foresee this being a problem? No, because I have lots of extra company specific fields in my sip_users table that asterisk doesn't use at all and I've had no problems. ie iax users have peercontext and auth. Just for curiosity sake, have you tried any SIP RealTime stuff? Perhaps this is an IAX problem? I remember helping a guy a few weeks ago get his SIP RealTime working. This is the first IAX I've dealt with. And I have no IAX stuff to test with. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving sip users multiple contexts
Is there a way I can (for sip users) also include say my [dial-911] [dial-local] and [dial-longdistance].. bearing in mind that I want to have different sips allowed to do different things so I can't just do includes for those in my from-sip-internal. Just make different context for different privileged like from-sip-internal-privileged and from-sip-internal-nopstn, etc in each context you include only the context you want them to have hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implant GIPS's codec to Asterisk
Gustavo García wrote: Hi everybody, GIPS have different products, not only codecs: * Voice enhancements: packet loss concealment algorithms, noise concealment, jitter buffer, agc, aec (can be used with any codec) * Codecs: iLbc (free), ISAC, G711 Wideband... You can include in asterisk voice enhancements and use them iLBC for example, for increasing the quality mainly in face of the packet loss, without using wideband codecs. I'm not a GIPS employee :-), you can view more information in the GIPS website. G. From what I have seen it appears those GIPS products are not particularly sophisticated. For example, have you any reason to believe they can achieve better jitter and packet loss handling than * with the new jitter buffer and PLC? That is not the world's most sophisticated, but as far as I get tell it is about on par with the GIPS offering. Does anyone have any evidence to the contrary? Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source Billing Software
Looks interesting, From the FAQ it looks like a 'metered' plugin for CDRs is coming but not available yet. Is this out of date or am I missing something? Of course you could just do the translation yourself from what I read... On Wed, 30 Mar 2005 12:24:23 +0200, Klaus Darilion [EMAIL PROTECTED] wrote: Take a look at http://ebills.sourceforge.net/ I uses latex to create nice pdfs. regards, klaus Christopher Snell wrote: On Tue, 29 Mar 2005 09:53:03 +1000, Rod Bacon [EMAIL PROTECTED] wrote: What I would like to know is has anyone found an open-source billing platform that performs basic billing functionality (pre/post) from RADIUS and/or Asterisk CDR and is written (well-written) in either PHP or PERL. What features and functionality is needed for such a system? I've been thinking about using Perl to write LaTeX source files, which can then be compiled into pretty PostScript and PDF paper bills or plain text that can be sent out by e-mail. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for SS7 design input
Wow, that did not take long. As with the current case before the US Supreme Court about file sharing and music copying, I am just writing software. What people do with the software is not under my control. My SS7 channel, app, stack or what ever, will be written from scratch in C++. If it just happens to work with Asterisk so much the better. Thank God there were no lawyers available when the when the wheel was invented. Between the Royalties and the Law suits we would all still be walking or riding horses. Race The 'I object your Honor' Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Tuesday, March 29, 2005 10:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for SS7 design input Race Vanderdecken wrote: I am looking for input on what an SS7 interface to Asterisk should look like and what it will need to be of any use. I was under the assumption that the licencing of SS7 prohibited it from being added to a GPL'd version of Asterisk... Is this not the case? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] APP CBMYSQL
I compiled and installed cbmysql.From the command line if I do a show applications should I see cbmysql in that list? I guess what I am trying to see is if cbmysql is connected to my mwqsql. IS there anyway. I was hoping to be able to do it from * CLI. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving sip users multiple contexts
just think it the other way round, group your users in different groups acording to what you want to let them do (ie: Managers, Marketing Employees, Salesman, etc) then create a context for each group, and include into each of those contexts what you want to let them do. hope this helps. bye, M. - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 11:53 AM Subject: Re: [Asterisk-Users] Giving sip users multiple contexts Well, I thought about that but wanted to check to see if there was another way... at the moment I have: 911 calling local calling international calling long distance. That's only 4.. but there are various combinations 911 and local... 911 and local and long distance... 911 and international (no long distnace)... I guess I can make up seperate contexts.. but it would be helpful if I could just delve the sips directly into a context. On Wed, 30 Mar 2005 15:49:16 +0100, Julian J. M. [EMAIL PROTECTED] wrote: Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc... Then in that context, include the features you'd like for each group, and give each sip user the correct context. Julian J. M. On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote: How would I go about giving sip users multiple contexts? For instance right now I have them all in: from-sip-internal Is there a way I can (for sip users) also include say my [dial-911] [dial-local] and [dial-longdistance].. bearing in mind that I want to have different sips allowed to do different things so I can't just do includes for those in my from-sip-internal. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving sip users multiple contexts
I know this is slightly round about and probably not recommended... but could I do an #include for each user... include their sip config in there as well as: context=sip-usersphonenumber [sip-usersphonenumber] include = theirsettings include = localstuff include = 911 ? On Wed, 30 Mar 2005 10:21:50 -0500, Matt [EMAIL PROTECTED] wrote: Right, I understand the logic behind this, and normally this is what I'd do.. but in this particular instance.. some users are going to have configs that are different then what others have... I guess the answer is NO.. you can not have multiple contexts on a sip without creating a context and includes... was hoping I could do includes on the sip user. On Wed, 30 Mar 2005 12:18:12 -0300, Matias G. [EMAIL PROTECTED] wrote: just think it the other way round, group your users in different groups acording to what you want to let them do (ie: Managers, Marketing Employees, Salesman, etc) then create a context for each group, and include into each of those contexts what you want to let them do. hope this helps. bye, M. - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 11:53 AM Subject: Re: [Asterisk-Users] Giving sip users multiple contexts Well, I thought about that but wanted to check to see if there was another way... at the moment I have: 911 calling local calling international calling long distance. That's only 4.. but there are various combinations 911 and local... 911 and local and long distance... 911 and international (no long distnace)... I guess I can make up seperate contexts.. but it would be helpful if I could just delve the sips directly into a context. On Wed, 30 Mar 2005 15:49:16 +0100, Julian J. M. [EMAIL PROTECTED] wrote: Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc... Then in that context, include the features you'd like for each group, and give each sip user the correct context. Julian J. M. On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote: How would I go about giving sip users multiple contexts? For instance right now I have them all in: from-sip-internal Is there a way I can (for sip users) also include say my [dial-911] [dial-local] and [dial-longdistance].. bearing in mind that I want to have different sips allowed to do different things so I can't just do includes for those in my from-sip-internal. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving sip users multiple contexts
Right, I understand the logic behind this, and normally this is what I'd do.. but in this particular instance.. some users are going to have configs that are different then what others have... I guess the answer is NO.. you can not have multiple contexts on a sip without creating a context and includes... was hoping I could do includes on the sip user. On Wed, 30 Mar 2005 12:18:12 -0300, Matias G. [EMAIL PROTECTED] wrote: just think it the other way round, group your users in different groups acording to what you want to let them do (ie: Managers, Marketing Employees, Salesman, etc) then create a context for each group, and include into each of those contexts what you want to let them do. hope this helps. bye, M. - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 11:53 AM Subject: Re: [Asterisk-Users] Giving sip users multiple contexts Well, I thought about that but wanted to check to see if there was another way... at the moment I have: 911 calling local calling international calling long distance. That's only 4.. but there are various combinations 911 and local... 911 and local and long distance... 911 and international (no long distnace)... I guess I can make up seperate contexts.. but it would be helpful if I could just delve the sips directly into a context. On Wed, 30 Mar 2005 15:49:16 +0100, Julian J. M. [EMAIL PROTECTED] wrote: Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc... Then in that context, include the features you'd like for each group, and give each sip user the correct context. Julian J. M. On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote: How would I go about giving sip users multiple contexts? For instance right now I have them all in: from-sip-internal Is there a way I can (for sip users) also include say my [dial-911] [dial-local] and [dial-longdistance].. bearing in mind that I want to have different sips allowed to do different things so I can't just do includes for those in my from-sip-internal. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] username/password for PolyCom IP500 web interface?
Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. -Garrett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hi! If I want to use ISDN card for connecting phones to it, that card must be HFC-S, because of NT mode. How about if I am connecting ISDN card to the external ISDN phone line (to local telephone companys s-bus) when card must be in TE mode, do I still have to have HFC-s card that I could forward incoming calls from pbx to phone(s) or could that be any ISDN card? Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA 2000 - Miltiple Ring Tones
Matias G. wrote: yes, ring back tone in Regional (Admin - advanced options in the web config utility) (this info is regarding Linksys PAP2 NA but they're almost identical) Are you suggesting to disable the ring indicator all together on the ATA? I don't think that would solve our problem. When calling SIP to SIP or SIP to IAX it's fine. Just SIP to Zap has the transposed ring effect. Unfortunately I haven't a clue how to debug this, so suggestions are welcomed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] username/password for PolyCom IP500 web interface?
Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. Don't Polycom often use the serial number of the device as the password? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username/password for PolyCom IP500 web interface?
Garrett Nelson wrote: Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. Those are in the ADMIN GUIDE http://www.polycom.com/common/pw_item_show_doc/0,,3641,00.pdf Username: Polycom Password: 456 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Comprehensive Asterisk Load Testing
[EMAIL PROTECTED] wrote: Its a very very bad idea to do this on production boxes. Especially if you are trying to see how far you can go, and then you cross that tiny border :) Your production calls will not like an idle cpu% of 0% and a load of 500. I could not agree more with you hence my question :) However, the tests results produced on test boxes: How realistic it is? Does it really presents real life scenarios and results? Does it take in consideration different type of services (calls, IVR, queues) ? I am not trying to put down anyone or anything here, I am just curious. Ta Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username/password for PolyCom IP500 web interface?
Garrett Nelson wrote: Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. -Garrett Polycom/456 Caps are important. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username/password for PolyCom IP500 web interface?
Polycom and 456 - Original Message - From: Garrett Nelson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 10:24 AM Subject: [Asterisk-Users] username/password for PolyCom IP500 web interface? Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. -Garrett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home 0.8 released
[EMAIL PROTECTED] 0.7 was a little buggy so we decided to release 0.8 It even has a few new features. AMP 1-10-007a SpanDSP 0.0.2pre11 vsftpd server If you have question about installing or configuring [EMAIL PROTECTED] please read the [EMAIL PROTECTED] Handbook. http://asteriskathome.sourceforge.net/handbook/ If you cant find what you need try posting to our discussion forum. http://sourceforge.net/forum/?group_id=123387 __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] username/password for PolyCom IP500 web interface?
http://www.voip-info.org/wiki-Polycom+Phones It's in the admin guide. User: Polycom; password: 456 Good luck. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garrett Nelson Sent: Wednesday, March 30, 2005 7:24 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] username/password for PolyCom IP500 web interface? Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. -Garrett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface?
I did find that in the admin guide, and it does not work. I have tried Polycom both capitalized and not capitalized. -Garrett Polycom/456 Caps are important. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for SS7 design input
Hi, You can write a GPL'ed SS7. There is nothing protected in the SS7 design. I don't think there ever were any patents. However, if there were they ran out long ago. Our non-GPL SS7 (because it is commercial) stack is written as a library in C. A modified chan_zap links it into Asterisk at the moment. This will change in the near future. Regards, Steve Race Vanderdecken wrote: Wow, that did not take long. As with the current case before the US Supreme Court about file sharing and music copying, I am just writing software. What people do with the software is not under my control. My SS7 channel, app, stack or what ever, will be written from scratch in C++. If it just happens to work with Asterisk so much the better. Thank God there were no lawyers available when the when the wheel was invented. Between the Royalties and the Law suits we would all still be walking or riding horses. Race The 'I object your Honor' Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Tuesday, March 29, 2005 10:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for SS7 design input Race Vanderdecken wrote: I am looking for input on what an SS7 interface to Asterisk should look like and what it will need to be of any use. I was under the assumption that the licencing of SS7 prohibited it from being added to a GPL'd version of Asterisk... Is this not the case? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over
On 23:34, Tue 29 Mar 05, Mitchel Constantin wrote: Matt, This isn't meant as a flame, rather I'm curious about what other people think about the following situation...maybe it's just the philosopher in me, what happens when the load balancer fails? Good point. Was thinking the same thing. Why load balance with one machine ? This is where CARP would be great. But besides that, what happens when connectivity to this specific location goes down ? Only way to provide real HA is to use 2 seperate locations, like 2 different countries :) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CISCO 7970 COLOR FROZEN
Title: CISCO 7970 COLOR FROZEN Hey Everyone, I bought a Cisco 7970 Color IP phone. I wanted to reset it back to factory defaults. I went through the sequence of holding down the pound key when the unit is powering on and then when the sequence changes to press 123456789*0#. The phone seemed to do something different after that. Now it is stuck in the constant cycle of going down the line buttons in a row of green lights. Can anyone help me with this? Thanks a million Dan - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using HFC-S card
Hi! If I want to use ISDN card for connecting phones to it, that card must be HFC-S, because of NT mode. How about if I am connecting ISDN card to the external ISDN phone line (to local telephone companys s-bus) when card must be in TE mode, do I still have to have HFC-s card that I could forward incoming calls from pbx to phone(s) or could that be any ISDN card? Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Upgrade *@home to CVS-HEAD
Hi Dean I haven't found any limitations as such. It just seems a lot of people have this impression of [EMAIL PROTECTED] as being a beginners tool. It was fabulous for the first couple of weeks, (as I have been running it for a couple of weeks), but I want to see how I go about migrating up to the big brother as it were. I gather the build of asterisk in [EMAIL PROTECTED] is a little behind the main distribution so since I think I have got what I want working I want to make sure I am up to date. It seems upgrading to the CVS build is a more effective option than trying to upgrade to [EMAIL PROTECTED] 0.7. Re: AMP I gave up on that after about half an hours playing and went straight into editing the .conf files manually, (SSH/NANO). I tried AMP but things didn't work so I chose to do it manually, I learn better that way :) Plus a lot of the discussion in here revolves around the conf files, and so to understand the discussion you really need to be familiar with the .conf files themselves. I have no complaints about [EMAIL PROTECTED] It did what I wanted, got me set up no problem. Once I'm upgraded and stable I'm going to try getting my PSTN linked in, and then probably a cheap SIP / IAX hardphone. Cheers Mark Charlton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: 30 March 2005 00:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Upgrade [EMAIL PROTECTED] to CVS-HEAD Mark, what exactly are the limitations you are finding? You do know that you can make modifications to the AMP dial plan don't you? Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Charlton Sent: Tuesday, March 29, 2005 6:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Upgrade [EMAIL PROTECTED] to CVS-HEAD Hi I installed [EMAIL PROTECTED] 0.6 to play with the system, and learn. I tried AMP but didn't like it and so set forth into the conf files manually. I have it all set up how I want, all my extensions work etc. Reading this list and playing in the wiki and google, I get the impression [EMAIL PROTECTED] is great for learning, but has a few limitiations. I would like to upgrade my box to the latest stable cvs build, but can't find any info on the process. How to save my .conf and recorded prompts and upgrade. If someone could point me in the right direction for resources on how to upgrade my * I would be most greatful. I assume the * build in [EMAIL PROTECTED] is a standard distribution, which I can just upgrade somehow. Thanks again for any help. Mark Charlton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call-ID and Unique-ID
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, March 29, 2005 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call-ID and Unique-ID The Call-ID is internal to the SIP protocol, and not exposed inside Asterisk (or via manager/AGI). The UniqueID is assigned by Asterisk to the call itself and should be used for tracking the call via the Asterisk interfaces. Thank you very much! Alex ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 1750 TDM400P - Power
-Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] I thought the TDM was broke on 1750's...?? I could never get passed that NMI issue. I don't know about the 1750s. On my 800, loading the TDM modules the first time causes an NMI, but it seems to be harmless. Wish I could make that front panel light stop blinking, though. ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface?
try this sir, Polycom SpIp- Original Message - From: Garrett Nelson To: Sent: Wed, 30 Mar 2005 10:01:05 -0600 Subject: RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface? I did find that in the admin guide, and it does not work. I have triedPolycom both capitalized and not capitalized. -Garrett Polycom/456Caps are important.Sean___ Asterisk-Users mailing listAsterisk-Users@ lists.digium.comhttp://l ists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman /listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris products available in the US?
Weird. For some reason when I googled I only got a page in the the UK. Anyone know if I can buy the 4801 and case and add a X100P card to it? I notice the bundle it with a Sangoma T1 card, but at the moment I need to test a single analog line. Thanks, Philip On Mar 29, 2005, at 8:58 PM, Josh McAllister wrote: Soekris is headquartered in Santa Cruz, CA. Buy direct from their website: http://www.soekris.com Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Philip Trauring Sent: Tuesday, March 29, 2005 6:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Soekris products available in the US? Anyone know if Soekris products are available in the US? Thanks, Philip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What the best Asterisk architecture for 900+ users?
Hi good people, A local Kenyan company wishes to improve its communication system by embracing VoIP technology. They currently have a legacy PBX with 17 analogue trunk lines and about 900 extensions. Going by the tender document, the main features they are looking for include: 01) Converged voice/data infrastructure fully compatible with ISDN. i.e. single connection point for both data (PC) and voice (telephone) 02) Cost control, i.e. who can call where and when, class of service, account codes, LCR and ARS. 03) Plug and play 04) Inter-branch connectivity (WAN) 05) Call detail reporting (call logging) software 06) Built-in voicemail 07) Built-in automated telephone operators [auto-attendant] 08) Built-in out of office notification 09) Built-in call conferencing 10) Built-in Direct Inward Dialing (DID) and Caller ID 11) Secretarial features i.e. ability for secretary to support several individuals using single handset. 12) Built-in hunt calling groups 13) Multi-line telephone handsets 14) Software integration (Built-in Computer Telephony Integration - CTI) 15) Built-in support for external/internal music on hold 16) Scalability to over 1400 extensions 17) Reliabililty i.e. real time operating system, hard drive mirroring and redundant power supply. 18) PSTN (Telkom) line interfaces i.e. digital and analogue lines 19) PABX connectivity i.e. ability to connect to traditional PABX using standards based protocol 20) Wireless handset/client. The LAN network consists of 800 access points at 100Mbps on a 1 Gigabit Ethernet backbone. The WAN connection (VPN) to a remote office is via a VSAT link at 64Kbps but is being upgraded to 128Kbps. I'm interested in giving them a proposal with Asterisk at the core, but I'm not sure of the architecture that best fits their needs. The architecture I have in mind would consist of at least 1 or 2 E1/PRI connections with DID to the Central Office then using a couple of rack mountable Asterisk servers with fully redundant hardware, doing the call processing bit in a distributed fashion. There are 2 branch offices with each having less than 10 users. Regarding IP phones, cost is not really the driving factor but rather the ease of central management with respect to configuration, troubleshooting and periodic firmware upgrades. The Polycom range look very attractive. So, what are your views on proposing Asterisk for this tender? How many high-end Asterisk servers do you think will be required to serve 900-1400 users? Is it possible to manage several Asterisk servers as a single virtual server? Your comments and remarks are welcome. Thanks regards, Alphonse Ogulla Nairobi, Kenya ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor command full static
I have a T1 going into *, SIP phones Grandstream Polycom IP500. Everything works great, but when I use the monitor command, or use IP Switchboard to record a call, the call has really loud static, and you can only make out maybe 1 or 2 words spoken. I have tried the IN-OUT, and combined wav files, and they are all bad. Running on Redhat 9 with SOX ,MPG123 0.59r, TE405p. These are calls going in, or out the T1. Any thoughts? Thanks much, Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for SS7 design input
I am looking for input on what an SS7 interface to Asterisk should look like and what it will need to be of any use. If you don't want to help then don't whine and complain about how you don't need SS7. All comments made in jest are welcome; points will be awarded for cheekiness and good puns. The code won't be written for a while because the design must predate the coding. But please let me know if you would like it done a certain way or need a certain feature. CLASS 5 or 4 SCP, SSP, SCT Local Exchange MU2A, MU3A SG Maybe you could throw some effort over here http://ss7box.com/asterisk.html This design to me looks well thought out, scaleble, GPL :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comprehensive Asterisk Load Testing
Bicom Systems wrote: [EMAIL PROTECTED] wrote: Its a very very bad idea to do this on production boxes. Especially if you are trying to see how far you can go, and then you cross that tiny border :) Your production calls will not like an idle cpu% of 0% and a load of 500. I could not agree more with you hence my question :) However, the tests results produced on test boxes: How realistic it is? Does it really presents real life scenarios and results? Does it take in consideration different type of services (calls, IVR, queues) ? I am not trying to put down anyone or anything here, I am just curious. Ta Senad Senad, I have yet to take a real hard look or contact Zoa, but if all you are doing is calling an extension (very rapidly and many, many times) it really would not be very hard to test queues, music on hold, meetme, etc. I am downloading the callgenerator from astertest.com right now... The most realistic test is to (obviously) register as many phones as possible and hire hundreds of people to talk on them... :) -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newline in an sms
How do I embed a newline into a sms message using the sms originate in * ? Julian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface?
try this sir,Polycom SpIp - Tried that, didn't work. Is my phone just messed up? Is there way I can change that password through the phone itself? Is there a way to reset the phone to factory settings? I know how to reboot it but didn't see a way to reset everything. -Garrett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Provider problems
We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. ADSL has slower upload speeds than download speeds (your 2Mbps is download). so you may have problems with your outgoing packets of sound. g.711 codec (the default codec for most voip providers because there is virtually no sound quality loss) uses about 84Kbps per channel or simultaneous connection. For example if you have an Upload speed of 128Kbps. and you try to have 2 phone conversations you would need 168kbps transfer speed. That is 40kbps more than your upload speed. This is a major problem with ADSL the upload and download speeds are not equal. Another potential problem is that your provider is over subscribed for the available bandwidth. What this means is that when allot of people are using their connection to your provider. The provider may not be able to handle all those users at once and packets get dropped or delayed. Dropping or delaying packets is very bad for VoIP especially if they do not do QoS or ToS routing which most providers do not. What is your upload speed? Some other possibilities are to use some compression codecs which will cause some sound quality loss like gsm or iLibc and g.729 to pack more calls in the limited bandwidth limitations. Another option is to use SDSL where the speeds of both the upload and download are the same. We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. This time is when businesses in the us are opening and starting to do business In the united states. Both for phones and Data. We do not know if this is it correct or all the people using VoIp provider feel the same quality? This may mostly be in relation to you Internet provider and how many hops you have to take to get to the VoIP provider and if they oversubscribe their bandwidth capacity. One provider may be good for one person with one person in a different ISP than an ISP you have. And you are even right next door to each other. This is as a result of how the internet is connected and may not nessessarly be geographic. For example you may be connecting to a server in your own city lets say Chicago but you are actually routed to San Francisco then back to Chicago. But it will not always take the same path the next time you may be routed through New York. This is a simplification of how it works. The closer you are to a Tier 1 provider(they own the major trunks interconnects) the less time it will take to get to your target. Anyone knows any provider without this kind of problems? I have seen many Providers have both Good and bad connection links. It is best to have a provider that routes with QoS and/or ToS within their routers and have only one or two hops between your provider and a tear 1 provider. Witch provider do you use to get the best sounds quality? It is not that simple. But you can begin by doing a traceroute to the many providers at different times of the day. This will see the route changes and time delays between hops to get to VoIP Providers gateways. Hope this helps in understanding the problems involved with choosing a provider. Thanks, Max ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris products available in the US?
Philip Trauring wrote: Weird. For some reason when I googled I only got a page in the the UK. Anyone know if I can buy the 4801 and case and add a X100P card to it? I notice the bundle it with a Sangoma T1 card, but at the moment I need to test a single analog line. Thanks, Philip Philip, I have a few people doing it with AstLinux (that I know of). You can find them on the astlinux-users mailing list at http://lists.kriscompanies.com to ask them about specifics. P.S. - If you want to use Asterisk on a Net4801, AstLinux is definitely the way to go (but it is me talking, after all)... -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using HFC-S card
On Wed, 2005-03-30 at 19:10 +0300, [EMAIL PROTECTED] wrote: How about if I am connecting ISDN card to the external ISDN phone line (to local telephone companys s-bus) when card must be in TE mode, do I still have to have HFC-s card that I could forward incoming calls from pbx to phone(s) or could that be any ISDN card? You don't need a HFC card in that case. Have a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+CAPI+Channels =Stefan signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using HFC-S card
On Wed, 2005-03-30 at 19:10 +0300, [EMAIL PROTECTED] wrote: If I want to use ISDN card for connecting phones to it, that card must be HFC-S, because of NT mode. Correct. How about if I am connecting ISDN card to the external ISDN phone line (to local telephone companys s-bus) when card must be in TE mode, do I still have to have HFC-s card that I could forward incoming calls from pbx to phone(s) or could that be any ISDN card? That card can be any ISDN card. Only the card to which you connect the phones needs to be an HFC-S card. -- dwmw2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for SS7 design input
Dunno if this matters at all but before embarking on a new project, you might want to have a look at this: http://www.openss7.org/ Maybe the license isn't open enough. I am but a poor peasant boy :) Race Vanderdecken wrote: Greetings All, I am looking for input on what an SS7 interface to Asterisk should look like and what it will need to be of any use. If you don't want to help then don't whine and complain about how you don't need SS7. All comments made in jest are welcome; points will be awarded for cheekiness and good puns. The code won't be written for a while because the design must predate the coding. But please let me know if you would like it done a certain way or need a certain feature. CLASS 5 or 4 SCP, SSP, SCT Local Exchange MU2A, MU3A SG Race The Tyrant Vanderdecken Somewhere near Timbuktu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soekris products available in the US?
There has been some discussion about this. Apparently true Digium X100p cards will work 3.3 volts, but some clones or other variety of X100p run at 5 volts and do not work. check out ASTLinux if you are interested in the Soekris. -Nate -Original Message- From: Philip Trauring [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Soekris products available in the US? Weird. For some reason when I googled I only got a page in the the UK. Anyone know if I can buy the 4801 and case and add a X100P card to it? I notice the bundle it with a Sangoma T1 card, but at the moment I need to test a single analog line. Thanks, Philip On Mar 29, 2005, at 8:58 PM, Josh McAllister wrote: Soekris is headquartered in Santa Cruz, CA. Buy direct from their website: http://www.soekris.com Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Philip Trauring Sent: Tuesday, March 29, 2005 6:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Soekris products available in the US? Anyone know if Soekris products are available in the US? Thanks, Philip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implant GIPS's codec to Asterisk
Steve Underwood wrote: Gustavo García wrote: Hi everybody, GIPS have different products, not only codecs: * Voice enhancements: packet loss concealment algorithms, noise concealment, jitter buffer, agc, aec (can be used with any codec) * Codecs: iLbc (free), ISAC, G711 Wideband... You can include in asterisk voice enhancements and use them iLBC for example, for increasing the quality mainly in face of the packet loss, without using wideband codecs. I'm not a GIPS employee :-), you can view more information in the GIPS website. G. From what I have seen it appears those GIPS products are not particularly sophisticated. For example, have you any reason to believe they can achieve better jitter and packet loss handling than * with the new jitter buffer and PLC? That is not the world's most sophisticated, but as far as I get tell it is about on par with the GIPS offering. Does anyone have any evidence to the contrary? I've read about GIPS' jitterbuffer stuff, and I think that our jitterbuffer implementation offers basically the same featureset. I would imagine that at this point, GIPS' implementation is probably better tested, but would be much more difficult to integrate into *. As far as the other DSP functions you mention, libspeex provides all of these, in varying degrees of progress (i.e. AGC, VAD, Denoise work pretty well, AEC does not yet work very well). Also, as far as wideband codec support, Speex supports both wideband (16khz) and ultra-wideband (32khz) modes, and these both work really well, as I use them in other applications. The work to include these (free, as in speech and beer) codecs would probably be roughly the same as for the wideband iLBC (not free, as in speech _or_ beer), and would benefit everyone out-of-the box, as opposed to just those who want to go through the trouble (and expense) of licensing a commercial codec. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SMS configuration
Wilson Pickett wrote: Quoting the wiki at http://www.voip-info.org/wiki-Asterisk+cmd+Sms appended to the end. The telco can define a default sub address (9 in the UK) which is used when the extra digit is not appended to the end. It says there's a default anyway. Note smsq doesn't send one (I guess this is bug in smsq). I rewrote the number to 17940099 anyway and it didn't make a difference. However it doesn't explain the issue. I send: smsq 0 register And it's supposed to text me back with a 'successful' message. The outgoing works. This isn't about extra digits or anything like that. The message centre then calls me back, but asterisk can't receive the message.. it hangs up after the first response. I listened to what is coming back and it is *not* trying to send voice.. it's actually a silent line that drops after about 5 seconds if it doesn't get the correct response (I'm guessing that asterisk isn't sending the correct response). Has anyone got this working in the UK? Do I have to set a country specific setting? Tony btw. I asked T mobile and they do not support direct SMS to landlines - they in fact couldn't understand why I would ever want it. The explains the mobile not working. They need adding to the list of mobile companies that do not work. (this renders the whole exercise academic of course since I can't actually use it.. it still annoys me it doesn't work). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface?
It is... Polycom 456 The setup for using new confs and app files is done through the phone anyway. Just setup the FTP server and your files. Then at least you should be able to get the latest app file son the phone to ensure it works right, even if not configured correctly. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garrett Nelson Sent: Wednesday, March 30, 2005 9:38 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface? try this sir,Polycom SpIp - Tried that, didn't work. Is my phone just messed up? Is there way I can change that password through the phone itself? Is there a way to reset the phone to factory settings? I know how to reboot it but didn't see a way to reset everything. -Garrett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bristuff and startup scripts
David, The Makefile in your zaphfc directory contains zaptel and zaphfc modprobe's for different systems (2.4 or 2.6 kernel, etc). Add the lines for your system to the asterisk startup script. eg: #! /bin/sh /sbin/modprobe zaptel /sbin/insmod /usr/src/bri-stuff.0.1.0-RC4a/zaphfc/zaphfc.o /sbin/ztcfg -v /usr/sbin/asterisk Eric. -Original Message- From: David Masure [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 3:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Bristuff and startup scripts Hi, I'm not the kind of Linux guru and I was wondering how I could start automatically the Zaphfc script. What I mean is that before starting asterisk, I have to type : make load from the zaphfc directory in order to load the zaptel driver. How can I do that automatically. This can be very useful in case of unattended reboot, Thanks Best regards David Masure ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can Asterisk do this ?
Hi 1. I wonder Asterisk can do this (refer to the following diagram) or not ? (Can I make a call from the SIP phone to the normal phone ) 2. Is the Asterisk server 2 called the PSTN Gateway ? 3. What are the hardware that I need to do that ? Hope that anyone can help me in this newbie question , thanks in advance for all . Rgds, Koa E-mail Disclaimer: This e-mail and any attachment(s) contain confidential information and are privileged. If you are not the intended recipient, dissemination or copying of this communication is prohibited and may be in breach of the applicable law. Please notify the sender and delete this email from your system. Thank you. From the Likom Management.attachment: question1.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for SS7 design input
On Wednesday 30 March 2005 11:16, TC wrote: I am looking for input on what an SS7 interface to Asterisk should look like and what it will need to be of any use. If you don't want to help then don't whine and complain about how you don't need SS7. All comments made in jest are welcome; points will be awarded for cheekiness and good puns. The code won't be written for a while because the design must predate the coding. But please let me know if you would like it done a certain way or need a certain feature. CLASS 5 or 4 SCP, SSP, SCT Local Exchange MU2A, MU3A SG Maybe you could throw some effort over here http://ss7box.com/asterisk.html This design to me looks well thought out, scaleble, GPL :) Hmm, my understanding is that Mike is developing a commercial SS7. -- NVC List Manager (Not Asterisk's) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
