Re: [asterisk-users] Caller ID Spoofing to be banned in the USA
On Jul 2, 2007, at 11:56 PM, Ron Stephan wrote: Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate Sure - it goes like this - The less scrupulous among us might use a spoofed cid to get people to do something they normally wouldn't. Imagine a spoofed CID of your corporate headquarters and somebody calling your employees saying they were HR and needed to confirm SSN numbers... I will let you fill out the rest of the disaster. Trouble is you don't think as evil as some people do. It annoys the hell out of me too - I would love to spoof my cell CID. I would love to have three or four cells with the same CID (all pointing back to my astericks box). It seems damn near impossible hear in Kalifornia. Ron Elvis Stephan Not to not pick, but I think you went beyond what Andrew was saying... Misleading or inaccurate, I would read this to imply that I'm not miss leading you, I'm not providing inaccurate information I am providing you with a means to contact me back. I'm opening stating who I am and where you can reach me with no malitious attempt. The Misleading or inaccurate part would encompass the scenario you describe above. Much the same the intention isn't to target corporate offices where employee's have DID's but their caller ID shows up as the trunk line which feeds to the building / company operator. Now, if I goto a provider and tell them my caller ID is the corporate number for Maytag and start calling people at 3am with is your refrigerator running that would count as Misleading :) -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk config files and #include
I don't have the source for 1.2.18 handy and didn't bother digging through my 1.4.4 tree looking but a quick grep for the exact error you see didn't reveal anything... although i greped the typo in maaximum However, correct that and that leads you to config.c #define MAX_INCLUDE_LEVEL 10 I suspect if your nesting a lot of includes you would probably need to up this level. I don't see a way to change this in asterisk.conf so I would suggest if you really need to go that deep in includes edit this option re-compile and be happy. NOTE *** This was in 1.4.4 maybe different in your version. I'm also not qualified to say from a quick glance if upping this limit has any negative impact but I would imagine it wouldn't and is more to help keep from causing loops. On Jun 21, 2007, at 6:37 AM, Deepak Bhat wrote: Im sure its not a circular include. Like you said its mostly realted to the number of nested includes but the exact meaning is not clear to me. Anyways to get it working I have consolidated most of my queue config files and am not including anything from files that are included. Thanks! Tzafrir Cohen wrote: On Thu, Jun 21, 2007 at 12:35:30PM +0530, Deepak Bhat wrote: Hi all, I am using asterisk version 1.2.18. I recently tried to change my asterisk configuration by using #include statements to include other config files in my extensions.conf and queues.conf files. My queues.conf is in /etc/asterisk. It includes several files which are in /etc/asterisk/queues. Each of these files contains the config of individual queues. Again each of the individual queue config files in /etc/asterisk/ queues includes files which are in /etc/asterisk/queues/queue_members. The problem is that when I reload this config I get the following error: - *WARNING: Maaximum include level exceeded : 10* Has anyone encoutered this before and does anyone know what it means ?? Any help will be deeply appreciated as I have been unable to find any documentation on this. Sounds like a circular include: in extensions.conf: #include extensions.conf The circle may include more than one file. To trac this, enable debugging and debug logging. There is a debug comment for each included file. Unless you really have such a complex nesting structure of include files and want that constant changed. That it easy to do by a code change. I don't really see a reason to make this configurable, until someone shows me a case where this does not indicate a circular include. Hmmm... so should the error message be changed to: *WARNING: Maaximum include level exceeded : 10. Check for circular includes.* ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
I would first ask are do you have mysql client libraries installed Do you have them installed in the standard locations... I tend to never install anything in normal places for me it makes easier version control to put everything in specific places. did you try just running ./configure and watch for the part about mysql libraries did it find them? try just running make and see if the error gives you a bit more information about missing files. if you get around just that you can simply copy the .so file to your asterisk directory but ofcourse it's got to compile first. On Jun 21, 2007, at 5:52 AM, Khaled Chehab wrote: No one faced a problem like this !! From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql …. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
On Jun 21, 2007, at 10:33 AM, Khaled Chehab wrote: I am using asterisk 1.4.5 with asterisk-addons-1.4.2 On /var/log/asterisk/cdr-csv/Master.csvthe unique id But in mysql database ,the unique id is not shown ,how can I fix it .. did you see http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? about changing the compile time option -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and reinvites...
I'm talking out my rear so someone please apply an attitude adjustment if I'm way off base. But, if you are using Dundi as a lookup engine it should know the contact information both endpoints and how to reach them perhaps not ONLY knowing how to comunicate via another asterisk box. Much like simply initializing a base dns infrastructure for the CPE devices. If the CPE devices are configured to accept SIP transactions from $domain or both asterisk servers server A should be able to send a invite directly to client B and bring up the inbound call. As far as the client knows it's still talking and placing outbound calls with server B. IE: Client A calls Client B Client A hits Serv A. Serv A does lookup finds it knows about Client B Serv A sends the call direct to Client B's IP. I'm assuming that both servers are acting as mirrors of eachother, in that voicemail and all that is a //shared// resource.. so if Client B rings unavail/busy that your serv A knows what to do with the call. In general as long as a client device knows to understand and accept sip messages from $host an inbound call does not have to come from the server they registered to. If you look at a linksys adapter this is one of the reasons they have that domain parameter which controls the list of hosts that are allowed to send SIP transactions to the unit. Am I wrong on this? The only other artifact I can think of is the fact of NAT traversal, where if client B that's to recieve the call is behind a NAT firewall and you are not doing port forwarding of the SIP signaling then ofcourse it won't get the call because server A has not established the NAT association. But assuming you are using a common 'sbc' or gatekeeper (ser) that box would know the association and things would be happy. On Jun 7, 2007, at 7:11 PM, Jared Smith wrote: On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! While I haven't taken the time to actually try this, I might suggest that you could set up separate user and peer sections in sip.conf, so that you can handle inbound calls differently that outbound calls. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write to multiple databases as redundancy scheme
Why would you do this why put the overhead inside asterisk when mysql has perfectly good replication mechanisms built in? On Jun 8, 2007, at 12:44 PM, [EMAIL PROTECTED] wrote: Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Right now I can only speak to the WIP300 but I've been evaluating it for about a week now and really I have to say I'm fairly pleased. It works it works //well// but that's not to say it's perfect. - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. - Call quality, so far so good however, I do believe the unit has a bit of an over touchy MIC.. the quality is clear but but it seems to pick up background noise and white noise pretty good. That's not to say that it will drown out your voice but you will hear the background if your in a server or climate controlled room. - The CPU on the phone does appear to be a bit underpowered. Two devices right next to each-other one a PC soft-phone and one the WIP 300 using the 'qualify' feature in asterisk you can clearly see a different in latency and how long it takes for the wip300 to process some sip transactions. This doesn't effect call quality but it is something worth noting if I'm taking the time to write this out. - Firmware: when you get the phone if it's running anything 1.0.9 upgrade to the latest from linksys, there are a slew of bugs that existing the factory shipping version that will likely make you think you really got cheated if you don't upgrade the firmware. Although make a note of the earlier thread upgrading has some bugs too, and don't try it form a mac. - I haven't tried the email function, as lack of intelligent keying (adaptive text for word completion) to me makes this a worthless feature. - Wireless, actually after changing to the latest version I've been fairly happy with the range coverage and life of the unit. You can load in multiple profiles for which AP you are talking to and the phone will register with that profile. You can associate different AP's with different SIP accounts which could be handy for traveling offices. The documentation doesn't mention it but you can create a profile that's a wild-card which will cause the phone to register to any open AP it finds. This I've found works fairly well as well as I can go from the east side to the west side of the building and the unit will switch AP's without much trouble, but do expect a dropped call in the process. If you have a mesh setup then the drop shouldn't happen but that's another story all together. I haven't played with the various encryption options so far as I've only been evaluating based on open access points. My thought is that likely the encryption may show more with the CPU load depending on how the unit manages this with it's chip set but even at that I don't think it will cause any red alarms. I could be wrong though. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote: On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. Considering I too have the sausage finger problem... the buttons are incredibly similar to what you find on the Nokia candy bar style phones. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linksys wip300 firmware
If your getting an error about invalid Octet key or something make sure you are doing the load from a Windows Computer it's been noted that a lot of people have had problems loading that version from various OS's. I went through a few computers before it took the load (for the record it worked from winxp / ie7) On Jun 2, 2007, at 7:48 PM, Ilan Rabinovitch wrote: Hello, Does anyone have access to version 1.00.07 of the Linksys WIP 300 firmware? The only version on their site at the moment is 1.00.09, which the phone refuses to load. Regards, Ilan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
for inbound connections how does asterisk manage host=host-name returning multiple A records... will it allow authentication for any of the IP's returned? I would assume that in the case of 'inbound' if you specify a host- name that you have PTR records for you could do it in one entry again I'm making a blind assumption. IE say you 10.23.23.3, .4, .5 as his IP's if you created entries either in your own dns or etc hosts (depending on os) you should be able to create entries for each of his IP's all resolving to the same name... and then one entry ... for his transactions from him - you. now the reverse of you - him you would in theory loose control over which host you send the call to but if he doesn't care then it wold work... and while this assumes you have no moral / security objection to using host-names. someone would have to keep my honest here though as I haven't looked at where asterisk does the NS lookup and how those transactions work. if it only read the conf file and did a translation at startup via a single lookup for host name then this wouldn't work. On May 30, 2007, at 6:11 AM, Yusuf wrote: Thing is, he does not REGISTER to me, he just uses me as proxy for his calls. I authenticate his calls in his IP. Alexandre VERNIOL wrote: Not supported jsut use host=dynamic with username and secret. Alex Yusuf a écrit : Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Does it loose it's IP address consistently and at a designated interval. I've seen something similar in a number of various cases where it was always an issue of the 'client' device blocking the DHCP renew traffic. But when it went into rebinding it would drop $filter and allow the dhcp traffic back through. I would say check to see if it occurs at a regular interval and compare that with the lease times. Also consider it's sometimes a good idea if the phone supports it (I don't know if your does) but setting up a cheap syslog host that can catch the syslog messages from the unit. Some units will log why they dropped their network (ie dhcp) or something. This was just a wild stab. On May 18, 2007, at 8:51 AM, Zeeshan Zakaria wrote: Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is effecting his business. Apparently this is a router related issue. Its a D-Link DI-624 router. This happens even when there is no Internet activity at all. How can this issue be resolved. What are the reasons for losing contact with the router. Is there some interference at port 5060, bad wiring, or something else? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR changes in 1.4.3?
snipped above and bellow I did a quick search through the online forums for reference to CALEA and didn't see much, What is the stance with asterisk and CALEA compliance. My assumption is and correct me if I'm off base it's a pbx box not an ss7 so any did's / npa-nxx are being delivered form a LEC. That mean the LEC is required to be complaint however asterisk in it's own guts is not. That being said if you were trying to be compliant the solution would be a external device that one of the clearing houses can talk to and setup the monitoring. Am I entirely off base here or am I looking at this the correct way? From a billing standpoint no whats the point? For statistical purposes I think its useful. For VoIP serviceprovider also very useful customer probably wants full call logs. I don't think your idea is too much CALEA-compliant either. Viva la CALEA! OK. Forget the filtering. The Gov shall have everything. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk manager interface stability
why not do it via an snmp interface? If you spend the time building an solid snmp base you would open up for an easier world of custom gui's as well as possibly some cleaner ties into an nms infrastructure. On May 16, 2007, at 10:14 AM, Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100 + clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. http://astguiclient.sourceforge.net/acqs.html MATT--- On 5/16/07, Dean Collins [EMAIL PROTECTED] wrote: If it's not stable what needs to be done to improve this? What are the issues? What are the alternatives (eg is Adhearsion an alternative here) I am about to start looking into a project that requires every user to have AMI access so looking to fund development in this space. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] ] On Behalf Of Damon Estep Sent: Wednesday, 16 May 2007 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk manager interface stability There are many past posts stating that AMI is not stable when multiple client computers are allowed to connect, particularly when connections are dropped. Has much progress been made on this? Is it more stable now than in the past? As of what versions were these issues improved? Is it feasible to connect a large number of windows computers directly to AMI for the purpose of initiating calls from software? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rapid DTMF missing digits
Yea that was my first guess until I saw the packet dump prove out that the ATA was transmitting it. Eh, let me go searching through the bug lists see if I can find something in older versions. On May 11, 2007, at 3:04 PM, Matt wrote: I have actually seen this behaviour on 1.2.x. I always assumed it was just me dialing too fast for the ATA. On 5/11/07, Bryan Laird [EMAIL PROTECTED] wrote: Version 1.4.2 but to be honest I have no reason at all to suspect that this is a problem with the asterisk software. I've able to replicate this from a few different client net connections and a across a few different linksys ata's. Where when you call into the host and enter the extension to connect to you miss the last digit of the extension. Almost every time you miss the last digit of the extension (in a 4 digit extension). My suspicion is simply because of the network we are currently using to host the asterisk box, as a packet dump on the lan segment clearly showed that the ATA transmitted all digits (rfc2833) but the asterisk host only recieved 3 of the 4. The second you dial slower everything works fine; also the lines for voice are clear with no noticeable impairments. I'm more curious if anyone else has ever run into a similar problem and what the resolution was if they found one (IE a sturdier net connection for the asterisk host), or Tweaking the timers on the ata's to slow down how fast and how long they transmit digits. I've done a few different tests and if I use a 'softphone' dialing directly into the server things work perfectly. I can dial as fast as I want, however when I come in through the pstn trunks through the upstream provider I find this problem. has anyone else ever seen this? Or seen a case where mis-matched dtmf modes across multiple providers causes this problem? minor detail on what I referred to as the 'pstn trunks' I have no analog or digital circuts all handoffs are sip. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird Saving Lost Packets since 1994 Have you seen this packet? 101010010 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rapid DTMF missing digits
Version 1.4.2 but to be honest I have no reason at all to suspect that this is a problem with the asterisk software. I've able to replicate this from a few different client net connections and a across a few different linksys ata's. Where when you call into the host and enter the extension to connect to you miss the last digit of the extension. Almost every time you miss the last digit of the extension (in a 4 digit extension). My suspicion is simply because of the network we are currently using to host the asterisk box, as a packet dump on the lan segment clearly showed that the ATA transmitted all digits (rfc2833) but the asterisk host only recieved 3 of the 4. The second you dial slower everything works fine; also the lines for voice are clear with no noticeable impairments. I'm more curious if anyone else has ever run into a similar problem and what the resolution was if they found one (IE a sturdier net connection for the asterisk host), or Tweaking the timers on the ata's to slow down how fast and how long they transmit digits. I've done a few different tests and if I use a 'softphone' dialing directly into the server things work perfectly. I can dial as fast as I want, however when I come in through the pstn trunks through the upstream provider I find this problem. has anyone else ever seen this? Or seen a case where mis-matched dtmf modes across multiple providers causes this problem? minor detail on what I referred to as the 'pstn trunks' I have no analog or digital circuts all handoffs are sip. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird Saving Lost Packets since 1994 Have you seen this packet? 101010010 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
