[asterisk-users] VIA EPIA DeadLock Issues
Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code) Asterisk Versions 1.2.12 - 1.2.14 - with and without metermaid patch Problem seems to happen more on systems that use parking lots. The system will run for around 24 hours or so fine, and then mysteriously, without any errors leading up to it, will stop being able to send calls to the chan_sip. System from that point on reports the following in the logs. Dec 13 12:07:04 DEBUG[16415] chan_zap.c: Took Zap/1-1 off hook Dec 13 12:07:04 VERBOSE[16415] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! attempting to stop asterisk from the CLI causes the CLI to become unresponsive and a trace shows chan_sip goes into a mutex_wait state. Anybody seen this? Have a fix? Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226___ --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] Server Hardware
Dell PE 1800s are our standard build. They are tower or Rack capable, have 3 open slots for expansion (2 if you get the remote access card). They are big though (5U) which is both a good and a bad thing. Good in that they have GREAT air flow inside the system so there is rarely any concern of overheating Digium cards. Bad in that they are friggin huge. Realistically though, if comparing in size to your average Avaya PBX for up to 100 users, than really the size is about the same. Regards, Raymond McKayPresidentRAYNET Technologies LLChttp://www.raynettech.com(860) 693-2226 x 31Toll Free (877) 693-2226 - Original Message - From: David Sampson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, August 16, 2006 10:57 AM Subject: [asterisk-users] Server Hardware Hello I am curious as to what hardware folks are using successfully from HP or DELL. I will likely be running just a quad span T1 card with the system. I appreciate your input. Thanks, Dave ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP asterisk over Linksys VPN
I usually run the RV series of router for this. Much better thoughourput on the VPN. Remember these low end devices can usually only handle about 1Mbps - 3Mbps of encryption max depending on the unit. Other than that, I have had up to 8 behind a VPN such as this. I do generally recommend though that a small appliance style asterisk box sit on any side of a remote connection with a 1 port FXO card installed for timing, emergency 911 capability, and trucking and jitterbuffer support over IAX2. This, IMHO, tends to provide for better reliability. I generally recommend some kind of HDD less Compact Flash based system. Less mechanicals to break and you can pick one up generally for $600-$800 with the digium card depending on speed and number of phones to support. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 1:23 PM Subject: [asterisk-users] SIP asterisk over Linksys VPN Has anybody tried using a VPN and around 10 phones behind the tunnel to connect to an asterisk server using Linksys VPN routers? Like this one: http://www.linksys.com/servlet/Satellite?c=L_Product_C2childpagename=US%2FLayoutcid=1115416832495pagename=Linksys%2FCommon%2FVisitorWrapper ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP asterisk over Linksys VPN
phones. Now I can call him be he can't hear anything. Everything else works fine through the vpn. Most importantly, I can't trouble shoot correctly. I finally gave up and got him callvantage. Now all I have to worry about is forwarding a DID number. I seem to remember some issues like that with older firmware on the routers. It was passing through some traffic but loosing others. Since I've upgraded every linksys I deal with to the latest firmware, all similar issues have disappeared. You might want to give that a try Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] Typical Asterisk Company
I'm flying all over the world installing systems for companies ranging from 5 employees - 100s of employees. I don't think there really is a typical company per say. Types of business span the financial services sector, to manufacturiring, to software development. The same general need exists across all of them. They all wish to. 1) Get a full featured phone system but not at the bloated prices charged by Avaya and Cisco 2) Take advantage of the substational savings through VoIP providers. 3) Integration of their communication technologies. I think after I install a 100 systems this year, I'll make some graphs on company type and sectors and see if there are any trends, but as of now, its all over the place. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 - Original Message - From: Martin Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 20, 2006 11:39 PM Subject: Re: [asterisk-users] Typical Asterisk Company On Jul 20, 2006, at 8:35 PM, Douglas Garstang wrote: I'm wondering what types of companies are most likely to be implementing Asterisk? These occur to me: CLEC's, and small rural telephone companies. ISP's wishing to provide voice services. Any others? Schools and fortune 500 companies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Encrypting the Conversation
Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN. Agreed. I have seen and heard of a lot of attempts to bring SRTP support into Asterisk but the idea of SRTP just doesn't make sense to me. Asterisk, and VoIP servers in general, are meant to be communications services not security services. In my mind at least, it would seem to make sense to let security hardware such as a router or firewall handle such tasks as encryption and let the phone server handle what it does, signaling and transcoding. Otherwise, you end up with a device that is not ever going to be optimized for security, handling your security. On top of that, you also are reducing the level of scalability you can achieve on the phone server by adding yet another chore to its duty roster. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] Encrypting the Conversation
That is at the server iself - you could then argue that the transit RTP could be tapped by a corrupt tech working for your ISP or provider, which could happen also with physical lines, the difference being that the RTP tap is so virtual it can be made to leave no trace. A physical tap can be found by a routine inspection on the lines, an RTP tap cannot. If we want Asterisk to be a step forward in the right direction, security concerns *must* be addressed at some stage. First let me say, I'm not totally against the idea of SRTP. There are certainly times when a VPN solution is unfeasible that it could be useful. In cases where you are trying to encrypt back to the PSTN to protect your call to that point, could also be another useful implementation for the paranoid or well justified security conscious. The idea though, that anyone anywhere can monitor your call just isn't true. Or at least isn't true without a lot of work consider what would be involved here. 1) First off, you would need to know the endpoints of the call. Capturing all the random RTP streams out there just isn't practical. The nefarious individuals of the world generally aren't going to work that hard so unless they have this information in particular, they will likely move on 2) You would need to be able to spoof, or otherwise compromise high end networking equipment within the ISP network. Generally, most people are using providers that are peered no more than 15 hops away. Most of those hops are on-net. Most large providers have pretty sophisticated IDS running. Heck I've even set off Comcast's a few times with my security analysis for other companies. This is not to say such a measure would not be possible, but you would be taking a lot of work to go this far. 3) If you have gone this far already, you either have balls of steel or you are an industrial espionage spy. These guys, and gals, are not going to be stopped by ANY security measure if they want to get in. Of course that is no reason to leave the door unlocked and hence SRTP on that leg of the VoIP journey might be useful. The problem is, no measure is going to stop an on network attack. If a disgruntled ISP tech has access to the SIP gateway on either side, any amount of encryption isn't going to do anything. If you can control the endpoints you can pretty much do anything you want. As always though, the weak point of any security is the people that run it. And managing that kind of security issue is a whole different topic all together. With all that said, I stand by my best practice concept of security happening on your network level devices. Such a design offers a scalable, centrally managed security model of which your trusted personnel will have access. It allows your communication hardware to focus on communication, a function it is optimized for, while your security hardware focuses on security. Additionally it leverages the infrastructure that most business users have already at this point while minimizing costs, and offering a reasonably secure platform. So to summarize and clarify my stance. 1) SRTP good in small doses where applicable 2) General Asterisk security ALWAYS a good idea. 3) VPN and other specialized security technologies are generally he most appropriate for scalability and overall security Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] menu system - configurator
Is there something out there that does something similar? Or does anyone know how to make such a script? If possible, we prefer mysql-driven menu's... as all other stuff is in mysql already... Try looking at the FreePBX front end or the Trixbox Distrobution. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] tdm04b strange noise when answering calls
Starting simple switch on 'Zap/2-1' HERE I START TO HEAR THE HIGH PITCH TONE Jul 4 22:50:12 WARNING[6658]: chan_zap.c:6135 ss_thread: CallerID returned with error on channel 'Zap/2-1' -- Executing Answer(Zap/2-1, ) in new stack -- Executing Playback(Zap/2-1, hello-world) in new stack -- Playing 'hello-world' (language 'en') -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (mefdialinc, s, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' Looks to me you are catching the end of the Caller ID signalling. Perhaps your telco is not behaving. Try putting a wait in your dialplan before answering. This should workaround the problem, or at least it has worked for me in the past. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] sip codec convertion on the fly
sip phone (ulaw) - asterisk - internet - asterisk - sip phone (ulaw) it is possible to force the two asterisk to convert the codec from ulaw to, say, gsm ? i mean, without touching the two sip phones Of course. On the trunk between the two Asterisk servers, just add disallow=all allow=gsm to the trunk config. From that point on, only GSM will transverse the trunk and the two asterisk boxes will transcode. Remember though, transcoding takes processor power so if you have more than one phone on each end, you are going to eat up processor power quick. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] Possible Bug?
I believe I've found a possible bug in the Zaptel channel drivers. I've been able to recreate this on a couple of servers. One with Asterisk 1.2.7.1 and 1.2.9.1,and also Zaptel 1.2.5 and Zaptel 1.2.6. I was testing a server configuration with some T1's we have with ATT. When I disconnected the ATT T1's we were no longer able to check voicemail. I noticed then that no messages from the server could be heard. Playback, voicemail or any other message the server would play to the caller on the phone. It would display on the CLI that it was playing the message but nothing could be heard and it would hang the call until it timed out. It took a lot of trial and error before I figured out that if I unloaded all related files to the Zaptel drivers the messages could then be heard. Reactivated the Zaptel with no T1's attached and again it killed the messages. One server has a Wct4xxp card and the other using dynamic with Redfone's from Fonbridge. Both reacted the same way. Sounds more like you have set the T1s as your primary timing source with no secondary. Are the phones connected to this analog off a channel bank by chance? If so if you don't specify something to take over timing, the channel banks won't sync audio properly to the Asterisk server and what you describe is likely to occur. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call waiting using free PBX
hi list, i have tried to set the call waiting function using freePBX but it dosent work. i think there is something wrong with the coding. Has anyone experienced this sort of problems? Can you expand a bit more on your problem? What versions of software are you running? What have you tried so far? Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Hardware Reliability
Also as Bruno suggests I'll pick a new UPS that has the phone line protection as well, though are phone lines are underground to the local station even though we are in a rural location. Cheaper than hanging it on poles I guess. A little tidbit of trivia here I've found the underground lines in some rural areas were a somewhat expensive experiment tried by some telcos. In some rural places in SC it was tried because the strong thunderstorms in the area tended to frequent damage above ground lines. The thought was putting them underground, while a bit more costly, might save some money in the long run. So in certain sections they tried running underground. As a result, those areas of the state usually now can't get things like DSL because it costs them too much to repull the grade of line to support it. That is until they suffer water damage such as in places like Mississippi after the last hurricanes. But I digress... Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Hardware Reliability
So now it appears to be working again, don't know what failed, don't know what made it work. and afraid of the next power outage at this rural SOHO. Might I recommend a large UPS connected to the asterisk Box. Power goes out and system then shuts down gracefully... This should equal no worries of card damage. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] h263 Video Support Questions
Hi, What asterisk release (stable or dev) has support for a softphone like Xlite (free) that uses h263 for video codec? (audio works fine) I have successfully used video support with all v1.2 code and current dev code Also, what (proven/tested) hardphones with video support can be used with asterisk? There are a few out there. I use the wooksung video phones. They are relatively expensive and seem to be highly stable and usable.. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] Digium Hardware Reliability
How reliable is Digium hardware in general.? My new TDM400P just died. I haven't had a Digium card go bad yet that I did not expect to go bad. By this I mean in the 100+ cards that I have installed, the only ones that have gone bad have been in servers that didn't have enough ventilation to handle them Digium cards tend to get very hot. If they stay in an overly hot environment, they tend to malfunction due to that heat and eventually die. If you had the card in a server where there isn't enough airflow, the power spike could have been enough to kill it in this instance. Remember the air flow need, and you should have a relatively reliable Digium experience. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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
Based upon your experience on the field what wifi sip phone would you reccomend ? I have used just about all the WiFi phones out there at this point. To be honest, its hard for me to recommend any of them. The biggest problem with most is that they just have no battery life. Take the Linksys WIP-300 for example. A great functioning phone. Works on just about every kind of wireless security out there. The problem is, you are luckly to get a few hours of use out of it. The UTStarcom phones have better battery life, but I've seen some wierdness when it comes to support of anything besides WEP encryption. Many other offerings only support WEP which can be a problem if your customer has moved on from them. So far out of what I tried, try the WIP-300 if you need support for WPA or WPA-RADIUS and the UTStarcom if you need better battery life. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] STUN?
Hi all, Could someone point at resources for running Asterisk behind a firewall. STUN keeps coming up but, alas, Im easily confused. J Ray ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk to use an outbound proxy
Dear all, Do anyone know to setup asterisk's SIP channel to use an outbound proxy outside of asterisk's network to proxy the SIP message? Thanks Ray -- ___ --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] Group funcations not functioning
Dear all,
we have try to limit the outgoing channel by using GROUP() and
GROUP_COUNT() to limit number of calls to a channel/trunk. but lately
we upgraded to 1.2.5, 1.2.6 or SVN 1.2 , both functions not work at
all. Is this a bug or just a misconfiguration on our part?
exten = s,1,Set(GROUP()=${count})
exten = s,n,GotoIf($[${GROUP_COUNT(${count})} 1]?IncCount)
Regards
Ray
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Re: [Asterisk-Users] Re: What causes deadlock?
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Does this happen with ooh323 channel driver? -- Tomislav Parcina tparcina#lama.hr ___ --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 sip to sip channels as well ___ --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] sip hang channels
Dear all, we have problem with hang sip channels when lot of incoming calls to registered users. we are using SVN 1.2 16771 version. xxx.xxx.xxx.xxx 639489919160 14aea9430d0 00102/0 unkn No Tx: CANCEL xxx.xxx.xxx.xxx 639485949581 1b3dd036127 00102/0 unkn No Tx: CANCEL xxx.xxx.xxx.xxx 639484589314 12f8b0fb4d6 00102/0 unkn No Tx: CANCEL xxx.xxx.xxx.xxx 639484131558 386850c668e 00102/0 unkn No Tx: CANCEL xxx.xxx.xxx.xxx 639482001240 466700fe087 00102/0 unkn No Tx: CANCEL xxx.xxx.xxx.xxx 639487637376 3aeedb022ef 00102/0 unkn No Tx: CANCEL xxx.xxx.xxx.xxx 639482001248 02f859b8511 00102/0 unkn No Tx: CANCEL xxx.xxx.xxx.xxx 639483957822 307a0bd50a5 00102/0 unkn No Tx: CANCEL thanks Ray ___ --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] Too many open files
Dear all, we have encounter problem when starting asterisk in the foreground, asterisk -gc with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! Apr 5 00:48:36 WARNING[14887]: chan_local.c:523 local_new: Unable to allocate channel structure(s) Apr 5 08:48:36 NOTICE[14887]: app_dial.c:1042 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown) Apr 5 08:48:36 WARNING[14893]: res_agi.c:246 launch_script: unable to create fromast pipe: Too many open files Apr 5 08:48:37 WARNING[14894]: res_agi.c:246 launch_script: unable to create fromast pipe: Too many open files Apr 5 08:48:38 WARNING[14897]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! Apr 5 00:48:38 WARNING[14897]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! Apr 5 00:48:38 WARNING[14897]: chan_local.c:523 local_new: Unable to allocate channel structure(s) Apr 5 00:48:38 NOTICE[14897]: app_dial.c:1042 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown) Apr 5 00:48:38 ERROR[14899]: rtp.c:933 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Apr 5 08:48:38 WARNING[14899]: chan_sip.c:3079 sip_alloc: Unable to create RTP audio session: Too many open files ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 65535 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16383 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited thanks Ray ___ --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] G729 and Meetme
I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the case. Can anyone explain to me exactly why this is. I don't really mind buying more licenses if I need to but I can't seem to wrap my head around where the Codec translation that is requiring the license is taking place. Regards, Raymond McKayPresidentRAYNET Technologies LLChttp://www.raynettech.com(860) 693-2226 x 31Toll Free (877) 693-2226 ___ --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] T38 fax pass thru to Cisco as53xx
Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? Weve tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw h323 sip Thanks Ray ___ --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] What business IP phone to use
there today, but have ended up sticking with my favorite three. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] 4 TE411P in one server installation
Dear all, Does anyone try to install 2 or multiple TE411 card into one server? Can it be done? What about stability? Thanks Ray ___ --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] agi/cagi call limit using group_count
Dear all,
Anyone has experience using group and group_count to limit
outgoing calls in AGI/CAGI?
SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP
EXEC Gotoif $[${GROUP_COUNT([EMAIL PROTECTED])}
1]?BLOCK
SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP
But it doesnt work as it should. Tried in extensions.conf
and it works.
Any idea.
Thanks
Ray
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[Asterisk-Users] CVS HEAD chanisavail not working for sip channel?
Dear all,
I have encountered problem with app_chanisavail for sip
channels. I have setup call-limit=1 in sip.conf as instructed, but when
making call to app_chanisavail, the channels did not increment correctly. I
end up dialing multiple times to the first channel only. I think the
ast_device_state(trychan) did not returned correctly. Any idea?
Extensions.conf :
exten = _1234.,1,ChanIsAvail(SIP/1SIP/2SIP/3SIP/4SIP/5|s)
exten = _1234.,n,Dial(${AVAILORIGCHAN}/0${EXTEN:4}|45)
exten = _1234.,n+101,busy
sip.conf :
[1]
type=friend
context=default
host=xxx.xxx.xxx.xxx
username=abcd
secret=abcd
port=5060
call-limit=1
fromuser=abcd
fromdomain=xxx.xxx.xxx.xxx
nat=yes
canreinvite=no
insecure=yes
insecure=very
disallow=all
allow=g723
allow=g729
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[Asterisk-Users] CVS HEAD chanisavail not working for sip channel?
Dear all,
I have encountered problem with app_chanisavail for sip
channels. I have setup call-limit=1 in sip.conf as instructed, but when
making call to app_chanisavail, the channels did not increment correctly.
I end up dialing multiple times to the first channel only.
I think the ast_device_state(trychan) did not returned correctly.
Any idea?
Extensions.conf :
exten =
_1234.,1,ChanIsAvail(SIP/1SIP/2SIP/3SIP/4SIP/5|s)
exten = _1234.,n,Dial(${AVAILORIGCHAN}/0${EXTEN:4}|45)
exten = _1234.,n+101,busy
sip.conf :
[1]
type=friend
context=default
host=xxx.xxx.xxx.xxx
username=abcd
secret=abcd
port=5060
call-limit=1
fromuser=abcd
fromdomain=xxx.xxx.xxx.xxx
nat=yes
canreinvite=no
insecure=yes
insecure=very
disallow=all
allow=g723
allow=g729
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[Asterisk-Users] does anyone know how to use 1.2 CVS setgroup in CAGI script
Dear all, Any one has experience in CAGI script setgroup? Please let me know a bit of command detail. Thanks, Ray ___ --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] how to edit or delete calleridname in From URI
Dear all, I would like to delete the calleridname in the FROM URI so it will not forward to the gateway in SIP. Ive tried everything available in SIP.conf but not able to do it. Please help. [test] type=friend context=sip-in setvar(CALLERIDNAME = ) callerid=123123123 123123123 username=123123123 fromuser=123123123 fromdomain=xxx.xxx.xxx.xxx secret=xx host=xxx.xxx.xx.xxx port=5060 nat=yes canreinvite=no Ray ___ --Bandwidth and Colocation sponsored by Easynews.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] lastest spandsp-0.03pre1 don't compile
Dear all, Anyone get the lastest spandsp with udptl.c and tpkt.c compile in Fedora 3? tpkt.c: In function `accept_thread': tpkt.c:140: error: `TCP_NODELAY' undeclared (first use in this function) tpkt.c:140: error: (Each undeclared identifier is reported only once tpkt.c:140: error: for each function it appears in.) tpkt.c:144: error: invalid application of `sizeof' to incomplete type `mansession' tpkt.c:148: error: invalid application of `sizeof' to incomplete type `mansession' tpkt.c:149: error: dereferencing pointer to incomplete type tpkt.c:151: error: `block_sockets' undeclared (first use in this function) tpkt.c:156: error: dereferencing pointer to incomplete type tpkt.c:157: error: dereferencing pointer to incomplete type tpkt.c:158: error: dereferencing pointer to incomplete type tpkt.c:159: error: `sessionlock' undeclared (first use in this function) tpkt.c:160: error: dereferencing pointer to incomplete type tpkt.c:160: error: `sessions' undeclared (first use in this function) tpkt.c:163: error: `t' undeclared (first use in this function) tpkt.c:163: error: `session_do' undeclared (first use in this function) tpkt.c:164: warning: implicit declaration of function `destroy_session' tpkt.c: At top level: tpkt.c:171: warning: no previous prototype for 'init_tpkt' tpkt.c: In function `init_tpkt': tpkt.c:173: warning: passing arg 1 of `pthread_kill' makes integer from pointer without a cast tpkt.c:177: error: `addr' undeclared (first use in this function) tpkt.c:178: error: `portno' undeclared (first use in this function) tpkt.c:179: error: `ba' undeclared (first use in this function) tpkt.c:180: error: `val' undeclared (first use in this function) tpkt.c:180: error: `cfg' undeclared (first use in this function) tpkt.c:188: warning: `return' with a value, in function returning void tpkt.c:190: error: `x' undeclared (first use in this function) tpkt.c:195: warning: `return' with a value, in function returning void tpkt.c:201: warning: `return' with a value, in function returning void tpkt.c:205: error: `t' undeclared (first use in this function) tpkt.c: In function `tpkt_rx_packet': tpkt.c:241: warning: implicit declaration of function `decode_open_type' tpkt.c:211: warning: unused variable `stat2' tpkt.c:212: warning: unused variable `i' tpkt.c:213: warning: unused variable `j' udptl.c: In function `udptl_process_packet': udptl.c:150: warning: no return statement in function returning non-void udptl.c: In function `udptl_build_packet': udptl.c:541: warning: implicit declaration of function `udptl_debug_test_addr' udptl.c:541: error: `udptl' undeclared (first use in this function) udptl.c:541: error: (Each undeclared identifier is reported only once udptl.c:541: error: for each function it appears in.) udptl.c:543: error: `iabuf' undeclared (first use in this function) udptl.c:544: error: `payload' undeclared (first use in this function) udptl.c:544: error: `res' undeclared (first use in this function) udptl.c:544: error: `hdrlen' undeclared (first use in this function) udptl.c: At top level: udptl.c:571: warning: static declaration of 'udptl_debug_test_addr' follows non-static declaration udptl.c:541: warning: 'udptl_debug_test_addr' declared inline after being called udptl.c:541: warning: previous implicit declaration of 'udptl_debug_test_addr' was here udptl.c: In function `ast_udptl_read': udptl.c:643: error: `payloadtype' undeclared (first use in this function) udptl.c:643: error: `timestamp' undeclared (first use in this function) udptl.c:643: error: `hdrlen' undeclared (first use in this function) udptl.c:649: error: `AST_FORMAT_T38' undeclared (first use in this function) udptl.c: In function `ast_udptl_new_with_bindaddr': udptl.c:706: error: `s' undeclared (first use in this function) udptl.c: In function `ast_udptl_write': udptl.c:820: error: `codec' undeclared (first use in this function) udptl.c:820: error: structure has no member named `lastts' udptl.c:820: error: `hdrlen' undeclared (first use in this function) udptl.c: In function `ast_udptl_reload': udptl.c:1084: warning: implicit declaration of function `ast_load' udptl.c:1084: warning: assignment makes pointer from integer without a cast udptl.c:1138: warning: implicit declaration of function `ast_destroy' ray ___ --Bandwidth and Colocation sponsored by Easynews.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] dtmfmode problem
All, I have the following config problem with dtmfmode I use ANTEK gw which only support dtmfmode=info but it is not supported in Asterisk voicemail. I wonder if it is possilbe to setup config that is runtime determined. I mean say, if I dial to voicemail then the asterisk can choose dtmfmode=inband or rfc2833 while switch to dtmfmode=info when I outdail to pstn. Cheers. Raymond ___ 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 ringback tone for VOIP call to another SIP server
All,
I found that there is no ringback to the caller (a-party) for
VoIP call but when I make call to registered user, I can hear the
ringback tone.
Beloware the debug log for the two cases:
I wonder if anyone who can tell me why?
Thanks.
Raymond
Case 1: no ringback to the caller (a-party) for outbond
VoIP call to another SIP server
Apr 26 07:04:09
VERBOSE[2607]: -- Executing Dial("SIP/30511694-abfa",
"SIP/[EMAIL PROTECTED]") in new stackApr 26 07:04:09 DEBUG[2607]:
Outgoing Call for 99740185293137656Apr 26 07:04:09 DEBUG[2607]:
99740185293137656 is not a local userApr 26 07:04:09
VERBOSE[2607]: -- Called [EMAIL PROTECTED]Apr 26 07:04:09 DEBUG[2607]: (Provisional) Stopping retransmission (but
retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:04:13 DEBUG[2607]:
(Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:04:13
VERBOSE[2607]: -- SIP/192.168.11.194-8dc7 is making
progress passing it to SIP/30511694-abfaApr 26 07:04:13 DEBUG[2607]: Ooh,
format changed from unknown to ulawApr 26 07:04:13 DEBUG[2607]: Auto
destroying call '[EMAIL PROTECTED]'Apr 26 07:04:13 DEBUG[2607]: RTP NAT: Using address
192.168.19.241:64868Apr 26 07:04:13 DEBUG[2607]: Oooh, format changed to
8Apr 26 07:04:13 DEBUG[2607]: Ooh, format changed from unknown to
ulawApr 26 07:04:13 DEBUG[2607]: Ooh, format changed from ulaw to
alawApr 26 07:04:15 NOTICE[2607]: RFC3389 support incomplete. Turn off
on client if possibleApr 26 07:04:32 DEBUG[2607]:
update_user_counter(99740185293137656) - decrement outUse counterApr 26
07:04:32 DEBUG[2607]: 99740185293137656 is not a local userApr 26 07:04:32
DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.Apr 26 07:04:32
VERBOSE[2607]: == Spawn extension (siptest02, 85293137656, 1) exited
non-zero on 'SIP/30511694-abfa'Apr 26 07:04:32
VERBOSE[2607]: -- Executing Hangup("SIP/30511694-abfa",
"") in new stackApr 26 07:04:32 VERBOSE[2607]: == Spawn
extension (siptest02, h, 1) exited non-zero on 'SIP/30511694-abfa'Apr 26
07:04:32 DEBUG[2607]: cdr_mysql: inserting a CDR record.Apr 26 07:04:32
DEBUG[2607]: cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2005-04-26 07:04:09','\"cisco 7960\"
30511694','30511694','85293137656','siptest02',
'SIP/30511694-abfa','SIP/192.168.11.194-8dc7','Hangup','',23,0,'NO
ANSWER',3,'')Apr 26 07:04:32 DEBUG[2607]: update_user_counter(30511694) -
decrement inUse counterApr 26 07:04:32 DEBUG[2607]: Acked pending invite
102Apr 26 07:04:32 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: FoundApr 26 07:04:32 DEBUG[2607]:
Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: FoundApr 26 07:04:32 DEBUG[2607]:
99740185293137656 is not a local userApr 26 07:04:32 DEBUG[2607]: Stopping
retransmission on '[EMAIL PROTECTED]' of Response 102: Found
Case 2: When I make call to registered user, I
can hear the ringback tone:
Apr 26 07:05:49 DEBUG[2607]: Auto
destroying call '[EMAIL PROTECTED]'Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to
4Apr 26 07:05:50 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: FoundApr 26 07:05:50
DEBUG[2607]: Setting NAT on RTP to 4Apr 26 07:05:50 DEBUG[2607]: Check for
res for 30511694Apr 26 07:05:50 DEBUG[2607]: Call from user '30511694' is 1
out of 0Apr 26 07:05:50 DEBUG[2607]: build_route: Contact hop:
sip:[EMAIL PROTECTED]:5060Apr 26 07:05:50
VERBOSE[2607]: -- Executing Dial("SIP/30511694-581e",
"SIP/30511690|20|tr") in new stackApr 26 07:05:50 DEBUG[2607]: SIMPLE DIAL
(NO URL)Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4Apr 26
07:05:50 DEBUG[2607]: Outgoing Call for 30511690Apr 26 07:05:50 DEBUG[2607]:
Call from user '30511690' is 1 out of 0Apr 26 07:05:50
VERBOSE[2607]: -- Called 30511690Apr 26 07:05:50
DEBUG[2607]: (Provisional) Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: FoundApr 26 07:05:50 DEBUG[2607]:
(Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:05:50
VERBOSE[2607]: -- SIP/30511690-adb1 is ringingApr 26
07:06:00 DEBUG[2607]: update_user_counter(30511690) - decrement outUse
counterApr 26 07:06:00 DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.Apr
26 07:06:00 VERBOSE[2607]: == Spawn extension (siptest02, 1690, 1)
exited non-zero on 'SIP/30511694-581e'Apr 26 07:06:00
VERBOSE[2607]: -- Executing Hangup("SIP/30511694-581e",
"") in new stackApr 26 07:06:00 VERBOSE[2607]: == Spawn
extension (siptest02, h, 1) exited non-zero on 'SIP/30511694-581e'Apr 26
07:06:00 DEBUG[2607]: cdr_mysql: inserting a CDR record.Apr 26 07:06:00
DEBUG[2607]: cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,chan
[Asterisk-Users] Re: NO ringback tone for VOIP call to another SIP server
Hi all, To my surprise, I change the Dial statement in extensions.conf from: exten = _852.,1,Dial,SIP/[EMAIL PROTECTED],r to: exten = _852.,1,Dial(SIP/[EMAIL PROTECTED],20,r) I can hear ringback tone now. I don't know why but it just works. Cheers. Raymond - Original Message - From: raymond To: asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 3:22 PM Subject: NO ringback tone for VOIP call to another SIP server All, I found that there is no ringback to the caller (a-party) for VoIP call but when I make call to registered user, I can hear the ringback tone. Beloware the debug log for the two cases: I wonder if anyone who can tell me why? Thanks. Raymond ___ 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] codec negotiation with CISCO 7960 and Firefly softphone
Hi all, When I am making call with with my Cisco 7960 SIP phone, I found only codec g711 works. The network call is like this: CISCO AS5300 ---sip Asterisk sip--- CISCO7960 Peer User/ANR Call ID Seq (Tx/Rx) Formatcisco7960 30511694 152828207b0 00102/0 ulawciscoAS5300 34169980 31A6EE0B-AF 00101/00101 gsm Codec in CISCO AS5300 g711alaw G.711 A Law 64000 bps g711ulaw G.711 u Law 64000 bps g723ar53 G.723.1 ANNEX-A 5300 bps g723ar63 G.723.1 ANNEX-A 6300 bps g723r53 G.723.1 5300 bps g723r63 G.723.1 6300 bps g729br8 G.729 ANNEX-B 8000 bps g729r8 G.729 8000 bps gsmefr GSMEFR 12200 bps gsmfr GSMFR 13200 bps I already did the following config in sip.conf allow=g723allow=g729allow=gsmallow=ulawallow=alaw However, it seems that only codec gsm work fines. However, it still occupy more bandwidth. Just would like to know if asterisk can do codec g723 or g729? Anyone has encountered this problem or am I missing any config in asterisk? Thanks. Raymond ___ 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] Fw: Registration Problem with Firefly Softphone
Hi all,
I found that my Firefly Softphone is not able to register to
Asterisk.
However, if I define the following lines on
extensions.conf
[from-sip-external]
;appended by raymond 24 marexten =
_997402.,1,Dial,SIP/[EMAIL PROTECTED],trexten =
_997412.,1,Dial,SIP/[EMAIL PROTECTED],trexten =
_997492.,1,Dial,SIP/[EMAIL PROTECTED],tr;end appended by
raymond
I will be able to make call.
-- Executing Dial("SIP/192.168.0.244-09fe4940",
"SIP/[EMAIL PROTECTED]")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/192.168.1.194-ff84 is making progress passing it to
SIP/192.168.2.244-09fe4940 -- SIP/192.168.1.194-ff84
answered SIP/192.168.2.244-09fe4940 -- Attempting native
bridge of SIP/192.168.2.244-09fe4940 and SIP/192.168.1.194-ff84 ==
Spawn extension (from-sip-external, 99749285234169800, 1) exited non-zero on
'SIP/192.168.2.244-09fe4940'
It appears that the call is default to the context
[from-sip-external].
I did entered my config in sip.conf
[34169788]type=friendusername=34169788secret=password88host=dynamiccanreinvite=nocontext=sipdisallow=alldtmfmode=rfc2833qualify=4permit=0.0.0.0/0.0.0.0
However it is not going to works.
Can anyone have setup on firefly with * and send me some
sample config?
Many thanks.
Raymond
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Re: [Asterisk-Users] Fw: Registration Problem with Firefly Softphone
Hi, I also define: The same thing with context [sip] in extensions.conf but it doesn't works so that why I cut-and-paste those lines: exten = _997402.,1,Dial,SIP/[EMAIL PROTECTED],tr exten = _997412.,1,Dial,SIP/[EMAIL PROTECTED],tr exten = _997492.,1,Dial,SIP/[EMAIL PROTECTED],tr from context [sip] to context [from-sip-external] Thanks for your advice on IAX2. However, my purpose is to the SIP conectivity. Raymond - Original Message - From: Stefan Gofferje [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 5:24 PM Subject: Re: [Asterisk-Users] Fw: Registration Problem with Firefly Softphone Hi, raymond schrieb: [...] [from-sip-external] [...] I did entered my config in sip.conf [...] context=sip What about using the same context for the firefly phone in extensions.conf and sip.conf? Besides, why don't you use IAX2? Firefly speaks IAX2 and for external clients, I think, IAX2 is better because it's nat-transparent. The remote client can be behind a nat without any problems. Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ 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] SIP Softphone for testing with Asterisk
Hello Rod, I try firefly but got problem of "Sip registratons failed for network (503)" Is it possible for you to advise me your config? Below is what I put on sip.conf [34169788]type=friendusername=34169788secret=password88host=dynamiccanreinvite=nocontext=internaldisallow=alldtmfmode=rfc2833qualify=4permit=0.0.0.0/0.0.0.0 asterisk1*CLI sip show peersName/username Host Dyn Nat ACL Mask Port Status 301/301 (Unspecified) D 255.255.255.255 0 Unmonitoredphone2/kissops (Unspecified) D 255.255.255.255 0 Unmonitored34169788/341697 (Unspecified) D A 255.255.255.255 0 UNKNOWN Thanks. Raymond - Original Message - From: "Rod Bacon" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Thursday, April 07, 2005 12:02 PM Subject: Re: [Asterisk-Users] SIP Softphone for testing with Asterisk I've tested about a dozen of them, and find firefly one of the best (others have more features, but I find firefly is a good mix of quality/features/performance). Make sure you get the third-party firefly though, not the one that's limited to virbiage. Try here... http://www.virbiage.com/firefly/download/firefly-thirdparty.exe - Original Message - From: "raymond" [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, April 07, 2005 1:41 PM Subject: [Asterisk-Users] SIP Softphone for testing with AsteriskHi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ 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] SIP Softphone for testing with Asterisk
Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ 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] codec for asterisk
Hi, I had try to set up the call routing for asterisk to interwork with cisco AS5300 and found thatAsterisk only support codec g711alaw and g711ulaw. For the other codecs (g723, g729, gsmfr), the callswere disconnected with cause value 63 (service option not available) or 127 (interworking error). Can anyone advise whether this is a restriction on asterisk? (or if I need to change anything on the standard config). Thanks. Raymond ___ 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] H323 for Asterisk
Hi all, I'm new to asterisk and had just install it on my linux server. Can anybody told me how to setup it up for interworking with cisco h323 voip gateway? I check throught the manual on http://www.digium.com/downloads/marketing/asterisk.pdf but cannot find any information for configuring h323. Cheers. Raymond Lau ___ 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] FC3 + udev + Asterisk v1.0.3 - Temporary Fix
I haven't seen anybody so far post a complex fix for the udev problems on
FC3 with the latest kernel. On that note, I have a temporary fix to allow
zaptel to load somewhat normally. I found that modifying the zaptel script
to 1) load, unload, then load the driver modules and 2) insert a pause
between modules seems to allow things to work. This assumes you have
followed the instructions and modified the udev rules and permissions as
documented on the wiki. Also, you may need to modify the length of the
sleep statements depending on the speed of your system. Modified zaptel
init script as follows
#!/bin/sh
#
# zaptelThis shell script takes care of loading and unloading \
# Zapata Telephony interfaces
# chkconfig: 2345 9 92
# description: The zapata telephony drivers allow you to use your linux \
# computer to accept incoming data and voice interfaces
#
# config: /etc/sysconfig/zaptel
# Source function library.
. /etc/rc.d/init.d/functions
[ -f /etc/sysconfig/zaptel ] || exit 0
# Source zaptel configuration.
. /etc/sysconfig/zaptel
# Check that telephony is up.
if [ ${TELEPHONY} = no ]; then
exit 0
fi
[ -f /sbin/ztcfg ] || exit 0
[ -f /etc/zaptel.conf ] || exit 0
RETVAL=0
MODULES=wcfxs wcfxo
RMODULES=wcfxs wcfxo
if [ ${DEBUG} = yes ]; then
ARGS=debug=1
fi
# See how we were called.
case $1 in
start)
# Load drivers
rmmod wcusb /dev/null
rmmod wcfxsusb /dev/null
rmmod audio /dev/null
action Loading zaptel framework: modprobe zaptel
echo -n Loading zaptel hardware modules:
for x in $MODULES; do
if modprobe ${x} ${ARGS} /dev/null; then
echo -n $x
sleep 1
fi
done
echo
# Unload Driver Modules.
echo -n Unloading zaptel hardware drivers:
for x in $RMODULES; do
if rmmod ${x} /dev/null; then
echo -n $x
sleep 1
fi
done
echo
# Reload the modules again
echo -n Loading zaptel hardware modules:
for x in $MODULES; do
if modprobe ${x} ${ARGS} /dev/null; then
echo -n $x
sleep 1
fi
done
echo
action Running ztcfg: /sbin/ztcfg
RETVAL=$?
[ $RETVAL -eq 0 ] touch /var/lock/subsys/zaptel
;;
stop)
# Stop daemons.
echo -n Unloading zaptel hardware drivers:
for x in $RMODULES; do
if rmmod ${x} /dev/null; then
echo -n $x
fi
done
echo
action Removing zaptel module: rmmod zaptel
RETVAL=$?
[ $RETVAL -eq 0 ] rm -f /var/lock/subsys/zaptel
;;
restart)
$0 stop
$0 start
RETVAL=$?
;;
reload)
action Reloading ztcfg: /sbin/ztcfg
;;
*)
echo Usage: zaptel {start|stop|restart|reload}
exit 1
esac
exit $RETVAL
Hopes this helps anybody else trying to implement on a FC3 base.
Raymond McKay
President
RAYNET Technologies LLC
http://www.raynettech.com
(860) 693-2226 x 31
Toll Free (877) 693-2226
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Re: [Asterisk-Users] Is this good packet latency/jitter ? (ping resultsfor BabyTel...)
On Jan 11, 2005, at 12:15 PM, Kim Lux wrote: I'm about to order an account with BabyTel. They are based in Montreal and have line access in most Canadian centers. Does this look good enough for VOIP ? Easily. I have connections where the latency is up to 300ms but a consistent 300ms without loss. The key to clear VoIP isn't always the latency but more of an issue of packet loss and ordering. As long as your packets arrive constantly and in order, most times you are going to find that the connection is good enough for VoIP. Mind you higher latency will equal more delay in communications and echo problems, but those can be dealt with. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com Phone: (860) 693-2226 x 31 Toll Free: (877) 693-2226 ___ 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] kind of urgent
In the zaptel directory, find the file README.udev. Find the # Section for zaptel device and take those five lines (KERNEL=...) and stick them in the file /etc/udev/rules.d/50-udev.rules and then reboot. If it helps any, I have Asterisk Running on FC3 with no issues. The udev thing was initially tricky for me, but I'm used to it now. Raymond McKay ___ 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] Status of SNOM Intercom
Using the new firmware is there still the issue with needing to patch chan_sip.c, or does it work out of the box? Do you have details on how it should be implemented within *? As of now, the hack still applies. It would be wonderful though if somebody could implement a command line variable that allows you to append anything to the SIP URI in the form of variable=variable. Right now the patch essentially breaks the VXML_URL functionality right now as stands. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ 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] Status of SNOM Intercom
It seems the current issue is that the Snom phones in current firmware don't want to accept intercom=true on the SIP URI. When passed this is the result and Asterisk retries and retries before exceeding a maximum # of retries to get the phone to connect. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.200.2.4:5060;branch=z9hG4bK196bc91f From: Cordless RAYNET sip:[EMAIL PROTECTED];tag=as138bec77 To: sip:[EMAIL PROTECTED]:5060;line=cg88tguw;intercom=true;tag=9vtji625gf Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060;line=cg88tguw WWW-Authenticate: Digest realm=10.200.2.4, nonce=11b09f72a06e02f6, algorithm=MD5 Content-Length: 0 I have a support call open on this too. A short fix on this would be to allow the configuration of AutoAnswer to be by line NOT by phone as it is now. I would think this would be a simple firmware fix and could get temporary intercom service to everybody who needs it for now. The fix for the intercom=true issue looks to be a bit more complicated as apparently the SNOM phones are looking for a bit more authentication that Asterisk just doesn't do at this point. So I guess my question is, Nils, any chance of getting this Auto Answer update change into the next firmware release? Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ 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 in a mixed phone environment
Hi, How difficult is to setup and maintain an Asterisk PBX with phones from multiple vendors? Is it even worth considering or is it safer to pick one vendor for phones and stick with them? I am more concerned about proprietary DHCP extensions, firmware upgrades etc..If anyone has any thoughts or experiences they would like to share I would be more than happy to hear from them. Having implemented quite a few mixed and non mixed vendor systems, my two cents is to stick with one vendor especially in a production environment. Here are my main reasons. 1) Standardization of features: If all the phones have the same feature set, there is less of a fear of incompatibility between endpoints. While you can minimize this though careful configuration and dialplans, for large systems, you need to almost be God to think of everything. At least with one vendor, you know a set of features to work with that will work across the board. 2) Easier Configuration and Maintenance: Pick one vendor type, be it SNOM, Cisco or others and you now only have one type of configuration file to maintain. Most vendors solutions can be configured TFTP so you can create configuration templates that will work across all the phones with only minor modifications. I have currently been recommending to people the SNOM phones for the simplest rollout of a large number of phones. They support config files via HTTP. More specifically, you can use a scripting language to generate the config files on the fly from a database. If you setup a database driven asterisk config, this essentially would ELIMINATE any individual file maintenance but it requires a single vendor, in this case SNOM, to work (BTW, for anyone who is interested, I hope to be releasing the fully dynamic phone code within the next few months once I actually have the time to sit down and fully write it) 3) Simpler troubleshooting: I'd hate to count the number of SIP traces I have had to do to figure out if a problem I was having was one endpoint, asterisk, or another endpoint. Having phones from the same vendor usually removes a step in the whole process. 4) Price: Buy more phones from one vendor, and you are likely to get a better price. By the time you are done combining high end desk phones and lower end phones, you might have been able to get a better bulk price on the higher end phones thus negating any cost savings by going with lower end phones. 5) TCO: All of these reasons are likely to lead to a system that costs lower to maintain long term. Remember, you can buy a really cheap car and add a whole bunch of third party options, but if one of those options fails, you have to remember who installed it, deal with varying degrees of support, and mostly spend more time figuring out who is going to resolve the problem rather than getting the problem resolved. The same occurs in most IT and Telecommunications installations. Having too many vendors in the pot tends to lead to the its not our problem syndrome I think we have all experienced one time or another. Remember that there is more cost to any system than what you pay upfront for it. With that said, the choice of the vendor for the system is especially critical. Is this vendor going to be in business next week? Do they provide the level of support you need? (Maybe you don't need any support, maybe you need high level support. Is that level available to you?) Does the vendor have a long term plan for integration with Asterisk? These are some tough questions and will vary based on your needs for future flexibility and upgradeability. With any luck, I'm hoping to have some time soon to write another article similar to http://voip-info.org/wiki-Asterisk+setup+soho+4+CO+12+extensions describing a low maintenance, and high reliability system config recommendation for a VoIP setup. (BTW, thanks to all who have written me thanking me for the clear recommendations) Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ 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] Sprint Vision Phones ReadyLink=SIP?
I was playing with a Sprint Vision phone recently and noticed when viewing the low level ReadyLink configuration screens that there are references to SIP registrars and the like. Does anyone happen to know if Sprint's implementation of ReadyLink truly is SIP based, and if so, managed to get it to interoperate with Asterisk. If so, it would prove to be an interesting paging mechanism and I would think would have immense value to any organization with multiple mobile individuals. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ 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] Over 10,000 lines. Will asterisk manage?
So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. I'm guessing, and I'd can't say for sure without seeing the actual physical layout of all of this, that the final solution would probably be a combination of SER and Asterisk with Asterisk getting used for endpoint connections and SER as a routing solution. There are really two virtual topologies that need to be considered to make such a judgment though. First, the actual network structure has to be finely analyzed. You need to know where your bottlenecks exist, latency issues within the network, and other such factors that could cause network issues. During the same time, its also probably a good idea to consider your potential network points of failure so you can plan on strategies should something go wrong. Second, you need to look at the virtual telephone exchange you are creating to understand how and where traffic is going to flow. In certain cases, you may want SIP devices talking to each other such as backend connections, but you really aren't going to want to have SIP endpoint devices doing this as 1) Some countries may and probably will start implementing wiretap requirements that will force you to redesign your entire network. 2) Accounting and control of devices is much harder when your devices are talking P2P. Just look at all the problems the RIAA has when trying to regulate P2P networks. 15,000 endpoints may sound like a lot, but realistically, never more than about 1/8 - 1/4 will be inuse at the same time depending on the environment. Realistically, I see this kind of size system being more of a network design issue than a VoIP one so the key is to make sure you have a good network engineer planning the network and knowing what that network is going to really get used for. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 833-9720 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] snom gsm codec
does anonybody know what is the status of gsm codec in snom phones ? they were some issuses in archives, some problems so i would like to know what is the actual status. best regards Marian I had problems with the SNOM phones with GSM when they first came out. One of the first couple firmware releases after the initial release seemed to fix the problem pretty early on though. I have a SNOM phone sitting at a remote office I use once in a while with GSM now and do not seem to have any problems with at this point. Regards, Raymond McKay President RAYNET Technologies LLC (860) 833-9720 http://www.raynettech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 RedHat Enterprise
Are their any issues with Asterisk and Redhat Enterprise? I have see one or two posts with issues concerning compiling zaptel ? drivers but that is about it. Just looking for some consensus to if any problems exist with it. I have one production system running with the T1 card with no issues. Its been running for about 3 months now and I haven't had any issues to speak of. Raymond McKay President RAYNET Technologies LLC (860) 833-9720 http://www.raynettech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 + Cisco 7920 + chan_sccp or chan_skinny
Greetings, I have seen a few postings in the past regarding the interop of Asterisk and the Cisco 7920 WiFi phone. To date, I have not seen a definitive method to getting the phone working. Assuming someone has this actually working, can that person step up and answer these questions. 1) What Channel is it working with (chan_skinny or chan_sccp)? 2) If code was used that is not a part of a current Asterisk CVS release, exactly where did that code come from? 3) What specifically is needed in the config files (skinny.conf or sccp.conf, SEPMAC.cnf.xml etc)? An example would be great. 4) Are any specific firmware versions required for the Cisco 7920 for the interop to work? 5) What codecs have been successfully tested? 6) What is the proper structure for the TFTP server root and files? If anyone can step up on this one and provide information, I would be happy to put it all together in a fully documented HOWTO so that anyone else attempting this configuration will have a clear and concise guide. Regards, Raymond McKay President RAYNET Technologies LLC (860) 833-9720 http://www.raynettech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Forwarding a call to another FXO port
If this is going to be more of a mainstay installation, I would highly recommend that you get a T100P card and channelbank. They work like champs and I've had virtually no complaints from any of those installs I've done. Thats actually what I'm using already. I had some issues with the X100P and SBC's lines so I put in a Adtran TA750 and a T100P a while back. This is the first time I have tried to do an FXO to FXO call though. FXO to FXS and vice versa are fine with no problems. Raymond McKay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Forwarding a call to another FXO port
-- Original Message -- From: Tim Thompson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 3 Dec 2003 13:22:25 -0600 I would change the option number to something else because 9 is often picked up in another context as 9NXXNX You might have to make a sub menu in order to get there, but try using 2-8 for the menu options. Actually I don't use 9 for anything in any context. For small dialplans I find it more confusing for users to have to dial 9 for anything. I don't think the problem is as much with the dialplan, though, as it is with the bridging of the two FXO ports Zap/1+Zap/2. When 9 is pressed by the caller, the call is made out the available FXO port and the cell phone ends up ringing. The problem really is that once that connection is made, the only audio that passes is a loud feedback type noise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding a call to another FXO port
Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten = 9,1,Dial(Zap/g1/CELLPHONENUMBER where CELLPHONENUMBER is the number it is calling out to. When option 9 is selected, a horrible feedback noise is heard and caller cannot hear anything else. The cell phone that the call is going to does ring and can be answered and hears the same noise. Hardware on this is Asterisk Box - T100P - Adtran750 FXO channels are 1 and 2 set in group = 1 Both channels otherwise operate normally echocancel = 64 echocancelwhenbridged = no Any ideas? Raymond McKay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT Troubles (SIP) - 407 Proxy Authentication Required?
For some reason, this 407 Proxy Authentication Required seems to be getting in the way... Any ideas? The UID and PW are fine in the 186 (it works great when it isn't behind NAT). I'd like to add that I have seen this same problem using Arrayvox Voxphones (SIP phones also) both behind and not behind NAT. Figured it was something wrong with the phones seeing how they are cheapies ($149 new). Maybe not. Raymond McKay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clear ADSI Configuration?
Greetings, There seems to be a bit of buzz on the list about ADSI phones and their configuration, but no clear progression of what really needs to exist to have a basic config. Could someone please post what they had to do to get an unlocked ADSI phone to work? Thanks Raymond McKay President RAYNET Technologies [EMAIL PROTECTED]
Re: [Asterisk-Users] Message waiting light on Cisco 7960
Will that work in the zapata.conf file also for phones that support MWI? - Original Message - From: Benjamin Miller [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 28, 2003 10:13 AM Subject: RE: [Asterisk-Users] Message waiting light on Cisco 7960 Just put a mailbox=XX in as part of the phone's definition in sip.conf where XX is the voicemail box number for that phone. -Original Message- From: Lenny Tropiano / asterisk.org Mailing list [mailto:[EMAIL PROTECTED] Sent: Thursday, February 27, 2003 11:28 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Message waiting light on Cisco 7960 Can I get the voicemail application turn on / off the MWI (message waiting indicator) on the Cisco 7960? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
