RE: [asterisk-users] Spurious hangups on zaptel interface
Title: RE: [asterisk-users] Spurious hangups on zaptel interface I did have busydetect=yes in my config, but not the callprogress.I've commented out busydetect, and we'll try some of these same calls to see what happens. -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]] Sent: Wed 9/27/2006 6:51 PM To: Barry D. Hassler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spurious hangups on zaptel interface Barry D. Hassler wrote: We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of flash (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1 Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf ___ --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] E1 crossover system
I am setting up an E1 crossover system for a customer, with a Siemens Hipath Officecom 150 system. And it's not working - I get a red alarm from the outside world, and a yellow from the PABX. Any ideas? We have tried E1 crossovers and straight cables on both connections, with no more luck. Frame type? Cabling? Timimg? Any ideas at all? Anyone? PaulH ___ --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] Re: max number of devices in hint
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm glad you asked :-) If we had Shared Line Appearances, I would not have to do this. However, I could be at any of about 6 different phones, and on any of about 4 lines per phone. Therefore, to monitor whether or not I am on the phone would take a 24 BLF buttons or just one, if hinting allowed that many. How many hands/ears you have? ;)) -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.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
[asterisk-users] media stream count
Hello ppl, Is it possible to get a count of the number of calls, for which the media is passing thru Asterisk and for calls, which are bypassing Asterisk for media? Thanks in advance. Ben ___ --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 Hangups on PRI Interface
Hi Sahil, I have a similar problem with PRI where some numbers cannot be called. I'm still analyzing the problem. Does it happen randomly or you have some numbers you can as it is for me? TIA Giorgio Incantalupo Sahil Gupta wrote: Hi, I seem to be having an issue with a PRI at present whereby the call works fine for 90% of the users however, when a customer begins dialling DTMF tones over the channel with the ASTCC application - the call seems to disconnect from the PRI Interface end: -- Channel 0/5, span 1 got hangup request -- Hungup 'IAX2/voicevalley-7' == Spawn extension (context, 099092428, 1) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' I have upgraded to the latest versions and have also ensured that busydetect and callprogress are turned off. Any ideas? Regards, Sahil Gupta VoiceValley ___ --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] SMS Text Send working with BT Text in the UK??
We are using 1.4 trunk with sms - it got fixed recently (about 4-5 weeks ago I think). Julian Scott Stingel wrote: Hi all- In 2004, I set up a sms texting process for a UK customer, using the asterisk SMS command and BT's BT Text SMS facility. This has been running fine up until recently. A couple of weeks ago, I upgraded them from their old asterisk version (CVS 2004.07.25) to 1.2.9.1, and have been having trouble getting the SMS feature to work on this newer version. I'm connecting to BT via a BRI, running an updated bristuff. (was also running this configuration previously) I do note the differences called out in the documentation, mainly that smsq is used to set up parameters for the text to be sent, and I've changed my code appropriately. Here is what I try: smsq --motx-channel=Zap/g3/17094001 --motx-retries=0 0111222 Hello! This seems to start things happening, as I observe the following on the asterisk console: --- -- Attempting call on Zap/g3/17094001 for application SMS(0) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH Sep 27 20:47:51 NOTICE[13661]: channel.c:2455 __ast_request_and_dial: Don't know what to do with control frame 15 Channel Zap/7-1 was answered. Launching SMS(0) on Zap/7-1 -- SMS RX 93 00 6D -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- Channel 0/1, span 3 received AOC-E charging 0 units -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- Hungup 'Zap/7-1' Sep 27 20:47:59 NOTICE[13661]: pbx_spool.c:279 attempt_thread: Call completed to Zap/g3/17094001 --- From looking at the app_sms.c code, I seem to be connecting to BT ok, but it appears that the 92 code received from them indicates an error in the format. As other posts have suggested,I have tried the following: (a) going back to version 1.2.7.1 (same symptoms) and (b) increasing the wait for response delay (h-opause) -no effect either. I've also tried reverting to my 2 year old app_sms.c, which no longer compiles (as expected) Does anyone have asterisk SMS texting via BT working in the UK, using a recent asterisk version, and if so, can you please shed some light on this? Many thanks Scott Stingel www.evtmedia.com ___ --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] includes in realtime ??
Hello ppl, follow up on a somewot old post. I set rtcachefriends=no and voila! changes to codecs, etc are immediately reflected! now.. that duz raise some issues .. hmmm cheerz Ben. Douglas Garstang wrote: If you want to use MWI, and I imagine most people would, you have to cache your realtime data, which means that changes to the tables do not become effective immediately. They become effective after you prune the entry in memory. Doug. -Original Message- From: RR [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 05, 2006 12:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] includes in realtime ?? Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Voip Buster - CID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... There are not many that will allow you to set your own CID even then they normally ask for proof of the numbers you wish to use. Hi Chris! So, you are saying that I can't set outgoing CID number on Voip Buster? Do you know for any VoIP provider that allows that? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.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
[asterisk-users] Re: ASTTAPI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. Hi Mike! I have been using ASTTAPI, but it takes time to configure it and I'm not sure it's developing any more. Now I'm using SNAP for several days but it seams that it has some bugs. I'm using Snap's forum to check with developer about this, but it's going slowly. I don't think that Snap is for business production yet. If developer doesn't solve those problems with Snap, I'll try Etelescript. Is Etelescript free? Is it open source? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.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
[asterisk-users] pay as you go t.t38 fax termination and origination
i can't for the life of me find a pay as you go termination and origination service.there's garfachi, but they don't offer DID's in anywhere else other than CA. Any suggestions? Thanks. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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] Re: RPID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Has anyone successfully gotten rpid working between two phones through asterisk? Hi Aaron! Can you please tell me what is RPID? Wikipedia and Google - define: RPID didn't help me. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.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
[asterisk-users] corrupt faxes
Hello, Since our telco messed with our PRI in some way, we get corrupt faxes like these: http://zenon.apartia.fr/stuff/corrupt_fax.pdf We use the lastest asterisk with a TE410P and spandsp. (for some strange reason, our neighbour company has a traditional pbx fed by 7 BRI's and sees the same problem) Now the telco is trying to racket us with some audit to solve the problem. They are claiming our pbx clockrate might be responsible. What could interefere with faxing in such a way? Could the telco have enabled some echo cancellation on their side? Thanks in advance for any insight, ___ --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] Multiple asterisk same GUI
Hi there, Im wondering, is it possible to have single GUI on same DB but write to different asterisk server? Means assuming you have 3 asterisk server with same configurations. Therefore with the same DB but it write to different asterisk server conf files. where is the connection that we should focus? Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --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: ASTTAPI
-Ursprüngliche Nachricht- Von: Tomislav Parcina [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 28. September 2006 09:10 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Re: ASTTAPI In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. Hi Mike! I have been using ASTTAPI, but it takes time to configure it and I'm not sure it's developing any more. Now I'm using SNAP for several days but it seams that it has some bugs. I'm using Snap's forum to check with developer about this, but it's going slowly. I don't think that Snap is for business production yet. If developer doesn't solve those problems with Snap, I'll try Etelescript. Is Etelescript free? Is it open source? After spending many many hours on asttapi and other tapisolutions, we found Tapi for Asterisk here: http://www.phonesuite.de/de/produkte/ast_tsp/phonesuite_tapi_for_asterisk.ht m It works like a charm and the licensefee with 25.-€/10 Clients is really fair. We couldn't find any bugs in the software and in combination with tapicall www.tapicall.de it's our preferred link-up to Outlook/Exchange in all of our asterisk installations. Since it's language is in german, you might have a closer look on some german dictionaries, but after configuration is done (5 minutes) you can forget about ever installed it. ;-) Hope, these informations saved you some time, money and nerves Guido ___ --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] importance of crc4 in zaptel.conf?
Hello, We have a TE410P connected to an EuroISDN E1 with these span definitions: span=1,1,0,ccs,hdb3 span=2,1,0,ccs,hdb3 span=3,1,0,ccs,hdb3 span=4,1,0,ccs,hdb3 Why should we add crc4 to these definitions? What does it do? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTTAPI
On 9/27/06, Mike Hammett [EMAIL PROTECTED] wrote: Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. You may want to take a look at ActivaTSP here: http://activa.sourceforge.net/index.html I had only a quick play with it, but it looked promising and is developing pretty fast. If you want to try AstTAPI, then I suggest that you checkout the CVS version, and build it yourself - This had several additions and fixes beyond the pre-built binary. Cheers, Steve ___ --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 Hangups on PRI Interface
Hi I have a similar problem with PRI where some numbers cannot be called. I'm still analyzing the problem. Does it happen randomly or you have some numbers you can as it is for me? We have the same problem you're reporting, Giorgio. Some numbers, mostly PBXs of other companies and sometimes some mobile phones, can't be reached through Asterisk, while they do answer when called from a mobile or an external line (I mean, not managed by Asterisk). The logs show a channel hangup, just like in Sahil's report. Some data: - Debian Sarge, default 2.6.8-3-686-smp kernel - Digium's TE210P, one interface connected to a Telefonica de España's T1 line and the other one to our legacy PBX - Asterisk 1.2.12.1, Zaptel 1.2.9.1, all compiled from scratch (have always had the issue, with several previous versions) - FreePBX 2.1.3 + some manual tweaks Any idea as to tests to do in order to pin-point and solve the problem will be appreciated. Regards, -- Vicente Aguilar [EMAIL PROTECTED] Departamento de Sistemas Tlf.: 965 98 71 92 Recursos en la Red, S.L.U. http://www.renr.es ___ --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] importance of crc4 in zaptel.conf?
Ask your Telco what you should use, they could be using CRC4 framing, in that case you have to match your config. Louis-David Mitterrand wrote: Hello, We have a TE410P connected to an EuroISDN E1 with these span definitions: span=1,1,0,ccs,hdb3 span=2,1,0,ccs,hdb3 span=3,1,0,ccs,hdb3 span=4,1,0,ccs,hdb3 Why should we add crc4 to these definitions? What does it do? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Erik Versaevel ___ --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] UK Colocation services
Hi,We can provide UK toll-free inbound and can recommend a co-lo provider where you'll have single-hop access to our network. Feel free to contact me off list.Kind regards,Simon On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Can anyone direct me to a colo provider in the UKwhere I can park an asterisk server and buy UK toll free inbound services over SIP? Thanks Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___--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] Asterisk Hangups on PRI Interface
Hi Vicente, I do not know if Sahil has a legacy PBX but you havethis is really interesting because I do have an old legacy PBX connected to Asterisk (via PRI trunk) which is connected to PRI telco. The problem arises when I call from an analog phone connected to the legacy PBX passing thru Asterisk to the telco. In other words if I call number 12345 from legacy PBX I receive the Channel 0/5, span 1 got hangup request but it works if I use a SIP phone connected to Asterisk PBX. I think the secret is in zapata.conf configuration..infact I changed some values and magically I could make outbound calls!!! The only drawback was I couldnt' receive any inbound call!! Do you have the same problem? If yes have you tried to call those bad numbers from legacy phones and from SIP/IAX? TIA Giorgio Incantalupo Vicente Aguilar wrote: Hi I have a similar problem with PRI where some numbers cannot be called. I'm still analyzing the problem. Does it happen randomly or you have some numbers you can as it is for me? We have the same problem you're reporting, Giorgio. Some numbers, mostly PBXs of other companies and sometimes some mobile phones, can't be reached through Asterisk, while they do answer when called from a mobile or an external line (I mean, not managed by Asterisk). The logs show a channel hangup, just like in Sahil's report. Some data: - Debian Sarge, default 2.6.8-3-686-smp kernel - Digium's TE210P, one interface connected to a Telefonica de España's T1 line and the other one to our legacy PBX - Asterisk 1.2.12.1, Zaptel 1.2.9.1, all compiled from scratch (have always had the issue, with several previous versions) - FreePBX 2.1.3 + some manual tweaks Any idea as to tests to do in order to pin-point and solve the problem will be appreciated. Regards, ___ --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] quadbri + tdm400p + modem-fax
Hello, I'm trying to send/receive faxes with a modem connected to a FXS port of a TDM400P. 50% of the faxes are sent/received bad. The installation is this: QuadBRI - Asterisk - TDM400P (FXS) - modem/fax - Zetafax Things I have tried: - reduce the speed of modem/fax to 9600bps = no changes - disable echo cancel in all channels = no changes - Answer() before Dial (to detect fax tone and disable echo cancel) = no changes Somebody has tried a configuration like this? Thanks in advance. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --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] PRI Backup
On Sun, 2006-09-24 at 16:47 +0100, adebayo omo-dare wrote: I don't know if this may at sometime help mr Wood, but BT, with their ISDN30* actually offer something called Site Assurance - the problem is that it does not automatically fail over, and according to the last memo I read - failover takes about 1 hr. Yes, I was thinking about ISDN2e. With ISDN30 BT gives you more options. On ISDN2e BT only offers Back in Business which essentially means you call them up and they divert the number to somewhere else. Helpful, but not really quick enough. A problem is that, due to outsourcing, product ranges, size issues, etc, a lot of people on BT's frontline are not really keyed up to their product offerings. Who knowns, maybe the failover process has been automated at this point in time. True. We buy PRIs from a wholesale retailer who resell BTs pris. It's technically the same thing and they seem to be quite clued on what the line can do and what it cannot. (Apart from being a fraction of the price). I found the experience dealing directly with BT quite frustrating. Conrad ___ --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 does SIP work?
(I'm sorry to ask this question here, but I didn't get a reply in VoIP-related forums and I figured there's a lot of people here who are knowledgeable about VoIP and SIP, and could help me see the light. Please replace Axon PBX server with Asterisk in SIP mode if you will :-) ) I finally got to have a working set up using an Axon Windows PBX software, Linksys 3102 gateway, a GrandStream IP phone and an X-Ten softphone over the Net... but I don't know _why_ it works :-) Here's how I think the whole thing works: 1. I set up the router to map UDP 5060 to the host where the PBX is installed, and I launch the Axon server 2. Remote phones connect through the Net into the Axon server to register their IP address and extension 3. When a call comes in from the PSTN network into the Linksys, the 3102 sends an SIP notification to the PBX. The PBX checks what extensions it must ring, and sends out SIP notifactions to all extensions involved. For this to work, all remote routers must also forward SIP messages to the IP phones that registered (UDP 5060 by defaullt, but each phone needs its own port to be reachable, eg. UDP 5060 for the first phone in the LAN, UDP 5061 for the second phone, etc.) 4. Once a phone goes off-hook, a connection is set up between the phone and the Linksys gateway. During the connection, each device tells the other what UDP ports it will use for RTP, ie. data packets. Provided this is correct so far, here's where things begin to blur: - If I don't set up remote phones to use STUN, connections are made, but I don't get sound in one direction: Is it because without STUN, the misconfigured phone sends its private IP in the data part of an SIP message, eg. 192.168.0.1, and since this is an unroutable address the other device won't be able to route data packets? - I didn't forward any ports for RTP, but calls still work: Is it because I happen to have UPnP-capable routers, hence RTP ports are automagically opened to make things happen? Thanks much for any hint :-) -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.407 / Virus Database: 268.12.9/458 - Release Date: 27/09/2006 ___ --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 Hangups on PRI Interface
El jue, 28-09-2006 a las 11:06 +0200, Giorgio Incantalupo escribió: Do you have the same problem? If yes have you tried to call those bad numbers from legacy phones and from SIP/IAX? Haven't tried. We're still using mostly analog phones connected to the legacy PBX, and starting to play around with a couple of VoIP phones. But I've experienced other problems when both analog and SIP phones are involved in the same call (transfering calls from an analog phone to a SIP one usually fails). I'll try as soon as possible. -- Vicente Aguilar [EMAIL PROTECTED] Departamento de Sistemas Tlf.: 965 98 71 92 Recursos en la Red, S.L.U. http://www.renr.es ___ --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] get the value from CDR
How can I get the value of the field (say billsec) in CDR after the
call is terminated? I have a dial plan below to get the billsec after
one of the party hangup the call. However, the value of billsec is
always 0.
exten = s,2,Hangup
exten = h,1,NoOp(channel=${CHANNEL})
exten = h,2,Set(billsec=${CDR(billsec)})
exten = h,3,MacroExit
___
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[asterisk-users] Sangoma a301
I am currently looking at designing a testsystem that incorporates Sangoma's a301 with either Tyan or Supermicro,and AMD combinations.I was wondering if anyone here has any experience of running a301s with*.In addition, if they would possibly have any experience specifically with the above type of setup.I am most interested in problems I may face, all/any conflicts involving the above,performance shortfalls, etc.Any and all information/help would be most appreciated.Thank youBayo. Inbox full of spam? Get leading spam protection and 1GB storage with All New Yahoo! Mail.___ --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 with cisco 7935
On Tue, Sep 26, 2006 at 01:13:14PM -0500, Ryan Amos arranged a set of bits into the following: =urn:schemas-microsoft-com:office:smarttags xmlns=http://www.w3.org/TR/REC- html40 I spent quite a bit of time debugging the 7935/7936, and it is an issue inside the firmware that Cisco knows how to work around in CallManager. There are better conference phone options available, and development on chan_sccp is basically dead at this point anyway, so I dont see this one ever being fixed. I would recommend a Polycom IP4000, its the exact same phone body but is much cheaper MSRP, and its SIP. Mine works fine with chan_sccp, and there's now SIP firmware out for it (which I haven't tried) I'm also considering taking back chan_sccp to get it working with 1.4, but can't do that until some of my contract work clears up. signature.asc Description: Digital signature ___ --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] 407 Proxy Authentication Required
There is a problem between 2 asterisk servers in message 407. In the normal flow, it should something like diagram1. However, in my case, I got the situation as diagram 2 and the call dropped finally. Does anyone have the same problem with me? How to solve the problem? Anyone can help? UA1 AST1AST2UA2 INVITE-- ---407 ---ACK-- INVITE-- -INVITE-- -INVITE-- ---continue as normal-- diagram1 UA1 AST1AST2UA2 INVITE-- ---407 ---ACK-- INVITE-- -INVITE-- -407- -ACK-- --call drop after timeout- diagram2 ___ --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] SER with multiple asterisk deployment
Adi Simon ha scritto: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). record_route() and loose_route() should help you, AFAIK. They don't? Cheers, Simone. ___ --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] No answer time
Hi, I use asterisk as gateway between PSTN and SIP users. It have install E1 card with CAPI interface. When I call from or to Asterisk and user don't pick up phone, Asterisk after 1 minute return NO ANSWER. Is it possible, that change this value to another (3 minutes)? -- Regards Domin ___ --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] 7940 vs. 7941
Any pros / cons on getting one over the other ? I was wondering what the main differences were. Julian. ___ --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 enable jingle in 1.4beta2?
Hi everyone, I'm going to compile this version of asterisk with jingle support, but I need to know how to enable this feature with the ./configure --enable option the sources chan_jingle.c and res_jabber.c aren't compiled by defalut. Thanks everyone ___ --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] 407 Proxy Authentication Required
I had some difficulty with this and an insecure=very in the appropriate section of sip.conf fixed it for me -- very annoying while it was happening. on Thursday 09/28/2006 unplug([EMAIL PROTECTED]) wrote There is a problem between 2 asterisk servers in message 407. In the normal flow, it should something like diagram1. However, in my case, I got the situation as diagram 2 and the call dropped finally. Does anyone have the same problem with me? How to solve the problem? Anyone can help? UA1 AST1AST2UA2 INVITE-- ---407 ---ACK-- INVITE-- -INVITE-- -INVITE-- ---continue as normal-- diagram1 UA1 AST1AST2UA2 INVITE-- ---407 ---ACK-- INVITE-- -INVITE-- -407- -ACK-- --call drop after timeout- diagram2 ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is this phone any good?
Does anyone know if the Gnet VP320S phones are any good? My supplier has them on sale until Friday at close (Sep 29th). I have two Gnet VP168S adapters and they are just good enough for testing purposes. Tim ___ --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] quadbri + tdm400p + modem-fax
Paco Brufal wrote: Hello, I'm trying to send/receive faxes with a modem connected to a FXS port of a TDM400P. 50% of the faxes are sent/received bad. The installation is this: QuadBRI - Asterisk - TDM400P (FXS) - modem/fax - Zetafax Things I have tried: - reduce the speed of modem/fax to 9600bps = no changes - disable echo cancel in all channels = no changes - Answer() before Dial (to detect fax tone and disable echo cancel) = no changes Somebody has tried a configuration like this? Lots have tried it. it doesn't work. Steve ___ --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] E1 crossover system
Paul Hales wrote: I am setting up an E1 crossover system for a customer, with a Siemens Hipath Officecom 150 system. And it's not working - I get a red alarm from the outside world, and a yellow from the PABX. What's your zaptel.conf? My suggestion: a. Do one step at a time. b. Asterisk - PBX: use a cross-cable. You probably need to enable CRC on the span. You need to take timing from the span connected to the telco. c. Asterisk - PSTN: should be straight cable. You need to take timing from the PSTN. Red alarm means not connected. Hope this helps. Leo ___ --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 = E1 = Alcatel OXO
Hi list! Has anyone of you connected Asterisk (Digium TE205) with Alcatel OXO thru E1 lines? I need to configure Asterisk so that every call from Alcatel OXO passes thru it. Asterisk will be between my provider (T-com in Croatia) and Alcatel. Thing is that, probably next week, I'll go on site to install Asterisk. And I need to prepare as best I can to make it work. And as far as I'm concern, best preparation would be working configuration. So, if anyone of you has done it, please send me your zapata.conf, zaptel.conf and extensions.conf files. Thank you. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.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
Re: [asterisk-users] How to enable jingle in 1.4beta2?
Afer running ./configure with whatever options you need, you should run make menuselect That will give you a menu to select the required modules. Modules marked with XXX are disabled, mostly because of a missing dependency. I think jingle requires iksemel. Good luck! Koen On 9/28/06, Raffaele Porzio [EMAIL PROTECTED] wrote: Hi everyone, I'm going to compile this version of asterisk with jingle support, but I need to know how to enable this feature with the ./configure --enable option the sources chan_jingle.c and res_jabber.c aren't compiled by defalut. Thanks everyone___--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] Asterisk with cisco 7935
Julien Goodwin wrote: Mine works fine with chan_sccp, and there's now SIP firmware out for it (which I haven't tried) Going into a conference room will cause Asterisk to segfault. After dialing out twice, the 7935 stops responding to key presses. I'll have to look at the Cisco site for the SIP firmware. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple asterisk same GUI
On 15:52, Thu 28 Sep 06, Sharon Lim wrote: Hi there, Im wondering, is it possible to have single GUI on same DB but write to different asterisk server? Means assuming you have 3 asterisk server with same configurations. Therefore with the same DB but it write to different asterisk server conf files. where is the connection that we should focus? What GUI ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] 7940 vs. 7941
At 05:39 AM 9/28/2006, you wrote: Any pros / cons on getting one over the other ? I was wondering what the main differences were. New phones (7941) support 802.3af POE. Old phones only Cisco special POE. New phones don't work with old SIP images. Only new unified SIP/SCCP images. New phones have a higher resolution display. New phones have some lighted buttons. Tom Julian. ___ --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] SMS Text Send working with BT Text in the UK??
Thanks Julian - will update and see if it works. regards, Scott Julian Lyndon-Smith wrote: We are using 1.4 trunk with sms - it got fixed recently (about 4-5 weeks ago I think). Julian Scott Stingel wrote: Hi all- In 2004, I set up a sms texting process for a UK customer, using the asterisk SMS command and BT's BT Text SMS facility. This has been running fine up until recently. A couple of weeks ago, I upgraded them from their old asterisk version (CVS 2004.07.25) to 1.2.9.1, and have been having trouble getting the SMS feature to work on this newer version. I'm connecting to BT via a BRI, running an updated bristuff. (was also running this configuration previously) I do note the differences called out in the documentation, mainly that smsq is used to set up parameters for the text to be sent, and I've changed my code appropriately. Here is what I try: smsq --motx-channel=Zap/g3/17094001 --motx-retries=0 0111222 Hello! This seems to start things happening, as I observe the following on the asterisk console: --- -- Attempting call on Zap/g3/17094001 for application SMS(0) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH Sep 27 20:47:51 NOTICE[13661]: channel.c:2455 __ast_request_and_dial: Don't know what to do with control frame 15 Channel Zap/7-1 was answered. Launching SMS(0) on Zap/7-1 -- SMS RX 93 00 6D -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- SMS RX 92 01 01 6C -- SMS TX 91 13 01 00 0B 81 70 97 22 63 86 F2 00 F1 06 C8 32 9B FD 96 01 AB -- Channel 0/1, span 3 received AOC-E charging 0 units -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- Hungup 'Zap/7-1' Sep 27 20:47:59 NOTICE[13661]: pbx_spool.c:279 attempt_thread: Call completed to Zap/g3/17094001 --- From looking at the app_sms.c code, I seem to be connecting to BT ok, but it appears that the 92 code received from them indicates an error in the format. As other posts have suggested,I have tried the following: (a) going back to version 1.2.7.1 (same symptoms) and (b) increasing the wait for response delay (h-opause) -no effect either. I've also tried reverting to my 2 year old app_sms.c, which no longer compiles (as expected) Does anyone have asterisk SMS texting via BT working in the UK, using a recent asterisk version, and if so, can you please shed some light on this? Many thanks Scott Stingel www.evtmedia.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100M location in circuit requirement?
I have just added a A1200P+FXO port to my home line for testing. In the interest of saving time, I wired it to the Phone port of my Fujitsu Speed Port DSL Modem. So in total, I have the line from the telco going to my Fujitsu, which goes to the FXO port. In parallel at the POP I have a DSL line filter in series with the rest of my house phones (2 phones, one modem, and the ADT alarm system). So I look at my console this morning as see all these events. Judging by the period, I am going to guess that my ADT alarm panel is calling home to check in on the parallel existsing phone system and Asterisk is seing that. Would that be correct? And.. When it comes to Asterisk, does it function fine as a second system to the same line as the house phones? (Also, can anyone point me to the list of configuration options for an X100M FXO module for the asterisk conf files?) Sep 27 21:50:16 NOTICE[17729]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 27 21:50:23 WARNING[17729]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 00:20:23 NOTICE[18249]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 00:20:30 WARNING[18249]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 01:50:23 NOTICE[18565]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 01:50:31 WARNING[18565]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 02:50:23 NOTICE[18874]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 02:50:31 WARNING[18874]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 04:20:25 NOTICE[20057]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 04:20:32 WARNING[20057]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 05:20:24 NOTICE[20387]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 05:20:31 WARNING[20387]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Sep 28 05:50:25 NOTICE[20556]: chan_zap.c:6073 ss_thread: Got event 17 (Polarity Reversal)... Sep 28 05:50:33 WARNING[20556]: chan_zap.c:6149 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (inbound_qwest, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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] Master.csv has stopped writing call logs.
I just checked my Master.csv file and found that no logs had been written to this file since 7:30 yesterday morning. The system is heavily used and nothing is being recorded. Has anyone ever seen this before? Does the Master.csv fill up? It's current size is 8591056 but there are no other files in the directory ( such as Master.1 or .2 ) as one might expect with traditional logging.Does anyone know what could be going on?Richard How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] New ptlib dependency-requirement in SVN-trunk?
Suddenly I can't get SVN-trunk to build anymore; the configure script is looking for something related to ptlib I don't have: checking for /root/pwlib/include/ptlib.h... no checking for /usr/local/include/ptlib.h... yes checking for ptlib-config... no checking for ptlib-config... no Cannot find ptlib-config - please install and try again Starting ./configure --without-ptlib does no good. I had never even heard of ptlib; the header file it found says it's a Portable Windows Library. Anyone with a clue on this I'd be grateful to get things to build again. Thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 asterisk same GUI
Michiel van Baak wrote: On 15:52, Thu 28 Sep 06, Sharon Lim wrote: Hi there, Im wondering, is it possible to have single GUI on same DB but write to different asterisk server? Means assuming you have 3 asterisk server with same configurations. Therefore with the same DB but it write to different asterisk server conf files. where is the connection that we should focus? What GUI ? PBXware can do above. In fact it was designed to do that from the day one. http://www.bicomsystems.com/products/ PLEASE NOTE: We are having some ISP routing issues, some parts of the world are not able to access the site hence please try later. Regards, Senad ___ --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: RPID
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Has anyone successfully gotten rpid working between two phones through asterisk? Hi Aaron! Can you please tell me what is RPID? Wikipedia and Google - define: RPID didn't help me. -- Tomislav Parčina Tomislav, RPID is short for Remote-Party-ID. Basically, Remote-Party-ID is a way, using a header (Remote-Party-ID) to completely separate caller id presentation from authentication information with SIP. I should point out that in standards tracks, Remote-Party-ID has been replaced by PAI (P-Asserted-Identity). Gotta love those standards :). -- Kristian Kielhofner ___ --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] SER with multiple asterisk deployment
Mainly I have a problem of figuring out how to use them with dispatcher or any other mean of switching between asterisks. Do you have any configuration example of such? On 9/28/06, Simone Ricci [EMAIL PROTECTED] wrote: Adi Simon ha scritto: Hi, Did anyone actually manage setting up a single SER with multiple Asterisk boxes? I particulary have a problem of keeping the session alive and by that I mean directing all the following sip messages to the same asterisk box the first signal was sent (randomally). record_route() and loose_route() should help you, AFAIK. They don't?Cheers,Simone.___--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
[asterisk-users] asterisk on 2.4 kernel ... scheduler problem?
Hello people! I have an inquiry (not a doubt ;D ). Actually, two. I am trying to run asterisk on an embedded Power PC platform on which we have a linux with a 2.4.2x kernel. In there, the linux scheduler runs at 100Hz. On a 2.6 kernel, the scheduler is at 1KHz. I just take this from a colleague ... hope it is true :) I only need to run the VOIP part, thus no POTS or external hardware. Actually, I just need SIP and H323 (channels/h323). Is there any problem to be expected from the scheduler difference? Or any other from running on a 2.4 kernel? Some colleague said that asterisk needs the 1KHz scheduler, but i cannot believe that it won't run on a 2.4 kernel ... Anyway, that is why i am asking. The other inquiry ... as the system is embedded we have not so much disk space available. So, i need a minimal asterisk installation. When compiled and stripped, the biggest amount of space is taken by the modules. My question is, can asterisk work with just the chan_sip.so, chan_h323.so and the codec_*.so? is there any other module needed? I need only be able to bridge sip to h323, no extra fancy stuff needed (parking, echo, blah, blah, ... ) Tks a lot! Cesc ___ --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] RE:T1 timing errors Nortel 61C with TE110P
One thing to note - changes to the timing parameter in zaptel.conf do not take effect on an asterisk 'reload' , you need to unload and load the zaptel driver. I've found it useful (on occasion) to power cycle the asterisk box too, as this _forces_ the far end of the E1 (T1 in your case) to start afresh. Tim. Tim Panton Thanks guys!! Thank you all for the response. The fix was power-cycling the entire server. All my settings were correct. Runs great now. Ronnie Jones Engineer - ICT Clay Electric Cooperative, Inc 352-473-8000 ext. 8272 352-473-1929(F) 352-745-0910(C) ___ --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] New ptlib dependency-requirement in SVN-trunk?
Brian Capouch wrote: Suddenly I can't get SVN-trunk to build anymore; the configure script is looking for something related to ptlib I don't have: checking for /root/pwlib/include/ptlib.h... no checking for /usr/local/include/ptlib.h... yes checking for ptlib-config... no checking for ptlib-config... no Cannot find ptlib-config - please install and try again Starting ./configure --without-ptlib does no good. ./configure --without-pwlib ___ --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] unable to call ATT audio conference bridge
Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead of getting the automated prompts to join the conference, it takes me to the customer support and then I got to give them the bridge number and pincode to add me into the conference call. The reason given by ATT was that their conference system is unable to identify our tone. This happens only with ATT conference bridges... not sure what the problem is. This problem started after I installed trixbox on a new hardware. Previous setup with [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] did not have this issue and I even switched back to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (a different box) and called the same conf bridge... that worked fine. I am running trixbox with the following versions: asterisk - 1.2.9.1 zaptel - 1.2.8 libpri - 1.2.3-1.349 using zap over a 8 channel pri Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contacts for Chan_gsm_bt maintainer?
Boris Bakchiev wrote: Anyone knows how to contact maintainers of Chan_gsm_bt? They http://changsmbt.free.fr/ site has no contact details. now this site is down. anyone knows why? can someone share lastest source code? ___ --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] txfax reliability on TDM cards
Hi all, What is the reliability of sending faxes with txfax? I am sending a 4 page fax. I have received 1 and 2 pages but never the whole thing? Do I have to have T1 or something different to reliably send faxes with a TDM card? Jerry ___ --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] Master.csv has stopped writing call logs.
Nevermind this post. The machines time was incorectly updated. The Master.csv is fine and performing dutifuly.Richard Reina [EMAIL PROTECTED] wrote: I just checked my Master.csv file and found that no logs had been written to this file since 7:30 yesterday morning. The system is heavily used and nothing is being recorded. Has anyone ever seen this before? Does the Master.csv fill up? It's current size is 8591056 but there are no other files in the directory ( such as Master.1 or .2 ) as one might expect with traditional logging.Does anyone know what could be going on?Richard How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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 on 2.4 kernel ... scheduler problem?
On Thu, Sep 28, 2006 at 03:24:27PM +0200, Cesc wrote: Hello people! I have an inquiry (not a doubt ;D ). Actually, two. I am trying to run asterisk on an embedded Power PC platform on which we have a linux with a 2.4.2x kernel. Still uses 2.4 today? Not a very good sign. In there, the linux scheduler runs at 100Hz. On a 2.6 kernel, the scheduler is at 1KHz. I just take this from a colleague ... hope it is true :) I only need to run the VOIP part, thus no POTS or external hardware. Actually, I just need SIP and H323 (channels/h323). Is there any problem to be expected from the scheduler difference? Or any other from running on a 2.4 kernel? Some colleague said that asterisk needs the 1KHz scheduler, but i cannot believe that it won't run on a 2.4 kernel ... Anyway, that is why i am asking. If you really want a 1kHz timing source for 2.4, build zaptel. But you'll need a USB UHCI chip. The other inquiry ... as the system is embedded we have not so much disk space available. So, i need a minimal asterisk installation. When compiled and stripped, the biggest amount of space is taken by the modules. My question is, can asterisk work with just the chan_sip.so, chan_h323.so and the codec_*.so? is there any other module needed? I need only be able to bridge sip to h323, no extra fancy stuff needed (parking, echo, blah, blah, ... ) Don't autoload modules in modules.conf . Load only the modules you need. Use 'load' from the CLI to manually load modules to see if you need them. One shortcut you may take is to use DeStar. It is an Asterisk configuration generator that generates a configuration with explicit load in modules.conf, rather than loading everything... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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 asterisk same GUI
Sharon,
pbxware.bicomsystems.com
U: [EMAIL PROTECTED]
P: pbxware
All standard.
Steve
steve {at] bicomsystems [dot} com
- Original Message -
From:
Sharon Lim
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, September 28, 2006 9:52
AM
Subject: [asterisk-users] Multiple
asterisk same GUI
Hi there, Im wondering, is it possible to have single
GUI on same DB but write to different asterisk server? Means assuming you have
3 asterisk server with same configurations. Therefore with the same DB but it
write to different asterisk server conf files. where is the connection that we
should focus? Thanks-- Regards, Sharon Lim
*Good memories are to be folded neatly and tucked away into the back
pocket *
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[asterisk-users] extensions.conf strangeness
Hello,
I have an anomoly that I am unable to explain.
My entire extensions.conf is attached. You can see that the [from-sip] and
[internal] dial plans are identical, each including 4 other contexts in the
same order:
[internal]
include = extensions
include = outbound
include = invalid
include = test
[from-sip]
include = extensions
include = outbound
include = invalid
include = test
I can place calls between extensions as expected.
Now here's the problem. If I dial '611' on a zaptel line (context
internal), I get the Hello World message from the test context as
expected. However if I dial '611' on a SIP phone (context from-sip), I get
I am sorry, that is not a valid extension
If I modify extensions.conf as follows:
[from-sip]
include = extensions
include = outbound
;include = invalidcomment out this line
include = test
then it works: dialling 611 from a SIP phone gives Hello World.
Could someone please explain to me why the dialplan seems to behave
differently for calls originating from SIP and zaptel lines in this instance?
I am running Asterisk from SVN trunk, compiled two weeks ago (September
13th)
Thanks,
Brian Candler.
[general]
autofallthrough=no
[incoming]
; incoming calls from the FXO port are directed to this context from zapata.conf
include = extensions
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Background(enter-ext-of-person)
exten = i,1,Background(pbx-invalid)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup()
[macro-ext]
exten = s,1,Dial(${ARG1},10)
exten = s,2,Playback(vm-nobodyavail)
exten = s,3,Hangup()
exten = s,102,Playback(tt-allbusy)
exten = s,103,Hangup()
[extensions]
exten = 101,1,Macro(ext,Zap/1)
exten = 102,1,Macro(ext,Zap/2)
exten = 301,1,Macro(ext,SIP/test301)
[outbound]
exten = _9.,1,Dial(Zap/4/${EXTEN:1})
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()
[invalid]
exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)
[test]
exten = 611,1,Answer()
exten = 611,2,Playback(hello-world)
exten = 611,3,Hangup()
[internal]
include = extensions
include = outbound
include = invalid
include = test
[from-sip]
include = extensions
include = outbound
include = invalid
include = test
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Re: [asterisk-users] How to enable jingle in 1.4beta2?
It worked, thanks!2006/9/28, Koen Van Impe [EMAIL PROTECTED]: Afer running ./configure with whatever options you need, you should run make menuselect That will give you a menu to select the required modules. Modules marked with XXX are disabled, mostly because of a missing dependency. I think jingle requires iksemel. Good luck! Koen On 9/28/06, Raffaele Porzio [EMAIL PROTECTED] wrote: Hi everyone, I'm going to compile this version of asterisk with jingle support, but I need to know how to enable this feature with the ./configure --enable option the sources chan_jingle.c and res_jabber.c aren't compiled by defalut. Thanks everyone___--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 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] Spurious hangups on zaptel interface
Title: RE: [asterisk-users] Spurious hangups on zaptel interface Commenting out the busydetect=yes seems to have resolved this annoying issue! Thanks Eric! On Thu, 2006-09-28 at 02:26 -0400, Barry D. Hassler wrote: I did have busydetect=yes in my config, but not the callprogress.I've commented out busydetect, and we'll try some of these same calls to see what happens. -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]] Sent: Wed 9/27/2006 6:51 PM To: Barry D. Hassler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spurious hangups on zaptel interface Barry D. Hassler wrote: We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of flash (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1 Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf Barry D. Hassler President HCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED] +1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED] Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] txfax reliability on TDM cards
Hello Use IAXmodem+hylafax instead. It works a lot more stable than rxfax and txfax. Probably something to do with hylafax being more accepting of errors. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Jerry Geis Sendt: 28. september 2006 15:51 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] txfax reliability on TDM cards Hi all, What is the reliability of sending faxes with txfax? I am sending a 4 page fax. I have received 1 and 2 pages but never the whole thing? Do I have to have T1 or something different to reliably send faxes with a TDM card? Jerry -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.9/457 - Release Date: 26-09-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.9/457 - Release Date: 26-09-2006 ___ --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] unable to call ATT audio conference bridge
On 9/28/06, asterisk-user [EMAIL PROTECTED] wrote: Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead of getting the automated prompts to join the conference, it takes me to the customer support and then I got to give them the bridge number and pincode to add me into the conference call. The reason given by ATT was that their conference system is unable to identify our tone. This happens only with ATT conference bridges... not sure what the problem is. This problem started after I installed trixbox on a new hardware. Previous setup with [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] did not have this issue and I even switched back to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (a different box) and called the same conf bridge... that worked fine. I am running trixbox with the following versions: asterisk - 1.2.9.1 zaptel - 1.2.8 libpri - 1.2.3-1.349 using zap over a 8 channel pri Thanks in advance. ATT's IVR to collect the passcode is coming through as early media and since you haven't signaled to the phones that the phone is answered they're probably not letting you send DTMF through the bridge that isn't technically supposed to be there yet. Put an Answer() in your dial plan prior to sending the call out to the Dial() application to reach the bridge for these types of calls and this generally fixes your problems caused by someone else not signaling correctly. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Polycom 501 One-way Audio
I have a site running an up-to-date version of Asterisk from the 1.2 trunk. We have a dozen Polycom 501 units and one of them (none of the others) is having recurring one-way-audio problems. As Murphy's Law dictates, it's the bosses phone! The user gets a few calls a day where the caller can hear her fine but she hears dead silence. It happens when she calls out sometimes too. Even internal voicemail and extension-to-extension calls are affected. I just called her three times from another extension; the first two were affected, the third got through. None of the other units seem to have the problem. They're all running the same firmware and are loading central configs that are identical except for line-button text and registration info. I've been running * with lots of debug/verbose logging enabled and have yet to see it complain about anything when she reports the problem. I'm about to replace her phone with a spare to see if that fixes it. Wondering if anyone has seen something like this and might be able to tell me what to look for as a potential cause. TIA, Paul Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. signature.asc Description: This is a digitally signed message part ___ --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] TDM2400P vs Sangoma A200
I had the exact same issue with the TDM2400E, choppy voice on the SIP side. I never called Digium Support, I simply removed the hardware echo can and everything was fine. I finally decided to buy a Sangoma A200. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]igium.com] De la part de Sampath Ganji Envoyé: 25 septembre 2006 23:44 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [asterisk-users] TDM2400P vs Sangoma A200 I have used both Sangoma and Digium cards. As far as TDM2400p with echo canceller is concerned, we had issues with static and choppy voice on the SIP side. However PSTN receivers had no issues. We called digium and they logged in remotely and found no issues in configuration. We are forced to move away from TDM2400p due to quality issues. We were also stuck with analog lines and approached Sangoma. It had no problems right from the start. Voice quality was excellent when compared to digium's card. Regards Sampath On 9/25/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Sep 25, 2006 at 09:27:12PM -0400, Dave Fullerton wrote: I never really considered one. I've never used one for that matter. This system is really only a testbed. If it works out, then in 6 months to a year we'll put asterisk in main site and all the phones in this location will slave off that system. But, I went to atacomm.com and tried to spec out a channel bank solution: Rhino Chasis: $750 2 4port FXO cards: $599 TE110P: $499 Echo canceler: Total: $1848 A TDP2400 8 FXO with echo can module: $1093 Mine didn't come out cheaper, what kind of equipment do you recommend? I think he meant something more like this: http://www.thevoipconnection.com/store/catalog/product_16317_AudioCodes_Analog_Gateway_MP1188_FXS.html Which I can't *recommend*, since I haven't used one... but like that. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] DesignerBaylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you.-- Jennifer Crusie; _Fast_Women_ ___ --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] extensions.conf strangeness
Brian Candler wrote: Hello, I have an anomoly that I am unable to explain. My entire extensions.conf is attached. You can see that the [from-sip] and [internal] dial plans are identical, each including 4 other contexts in the same order: [internal] include = extensions include = outbound include = invalid include = test [from-sip] include = extensions include = outbound include = invalid include = test I can place calls between extensions as expected. You need the [general] and [global] sections ___ --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] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]
I concur with your approach, but Tier 1 means as little here as it does when evaluating Internet backbone carriers. could you expand on what evaluation criteria you use? I'm going to be pre-speccing some stuff myself this month... Sorry I should have been more clear. A good Asterisk install needs a holistic approach to use a hippy dippy phrase. A Tier 1 server, which is a midrange to high end name brand server from the Big 3 (Dell, HP/Compaq, IBM, am I missing someone?) is usually highly optimized for bus bandwidth although that design was intended for a different use - usually massive disk I/O. As well, a Tier 1 server will have two seperate, independent PCI buses and this to me is a critical feature - it allows you to completely separate your TDM traffic from network, disk I/O etc. On my big production Netfinity, I took great care to ensure the Digium cards were all on their lonesome on a single bus, and everything else on the other bus. This is how I can run two TE110's in a single box with no problems. zttest does not give me 100% all the time, but on the other hand it *never* drops below 99.9987%, even under load. I selected this Netfinity because of the obvious care put into it's design, but the specs are unimpressive: quad Xeon 700's. CPU is over rated for Asterisk, IMO unless you are doing tons of transcoding and if you are doing that, then your design is flawed. Anyway, the holistic approach (to go on a small rant for the newbie lurkers) be summed up as follows: 1. Good box, see above 2. Good LAN - this is so critical and so often overlooked in the day and age of guys crimping their own cables and running $150 switches. You can't do that, and if you do, you do so at your own peril. Managed swiches, professional cable installation. This is not a problem for me since I *am* a professional cable installer but I have actually witnessed people making patch cables with a flat blade screwdriver and a hammer! 3. Tuning of the LAN - VLAN's are good. QoS packets are good. Switches that honor the QoS packets are good. 4. Handset selection - this is another biggie. I've selected Snom 360's, and yes they have warts, but they are feature rich for the price and Snom is really good about revising firmware. When you select handsets, GET YOUR USERS INVOLVED. 5. Tuning of Asterisk box itself - this cannot be under emphasized. This is a very important step and tuning methodologies vary according to distro, skill of the admin, and particular circumstances. I've learned *way* more than I ever wanted to about processor affinity sinc I started using Asterisk. 6. Termination of PSTN. Basically I would never do an Asterisk install where I was forced to do something stupid like aggregate a dozen Centrex lines or some mickey mouse deal with FXO ATA's or whatever except for a hobby or prototype install. PRI, BRI, IAX or SIP, don't mess around with anything else. 7. Relationship with provider. What is their SLA? Is it the incumbent or the clec? An incumbent will be more expensive and more difficult to deal with but they will tend to be more reliable. A clec will be cheaper and they will be way more accomodating but you will most likely not get five 9's from them. A VoIP provider should never be trusted, period. You will not get five nines from them, ever. Plan failover situations accordingly. 8. Plan plan plan plan. A good install of ANYTHING is 80% planning 20% doing it. What is your plan when your primary PSTN provider fails? What is your plan if your Asterisk box goes pear shaped? My dialplan can survive either PSTN, WAN or LAN failure (albeit with reduced functionality). I also keep a cold spare, an identically configured box that I can literally throw into the rack, turn it on, plug in the PRI's and no problem. 9. Internet bandwidth and latency. I am fortunate enough to have a great IP provider. Ask for demos - most guys will install a 90 day trial or something like that. Do not believe the brochure, get the product installed and put it under load. 10. Traffic prioritization at the IP demarc - total no brainer. 11. Constant, constant user feedback and remediation. If you are not talking to your users, your install will ultimately fail even if you have the best of everything. Underpromise and overdeliver. Never loose sight of the basics - they have to pick up the phone, and it has to work. Always. ___ --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] Is this phone any good?
Does anyone know if the Gnet VP320S phones are any good? My supplier has them on sale until Friday at close (Sep 29th). I have two Gnet VP168S adapters and they are just good enough for testing purposes. Tim (Sorry if this comes in twice, but I got an error when sending the first time.) ___ --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] quadbri + tdm400p + modem-fax
On sep/28/2006, Steve Underwood wrote: Lots have tried it. it doesn't work. With Sangoma cards it will work? Thanks. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --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] E1 crossover system
Dunno about your country but I have the same setup here in Germany. I have 1:1 Cables (I think). I have in zapata.conf: [channels] progzone=nl callgroup=1 pickupgroup=1 loadzone=nl,us defaultzone=nl context=zap-in priindication=inband switchtype=euroisdn ... group=1 context=telekom signalling=pri_cpe channel = 1-15,17-31 loadzone=nl group=2 signalling=pri_net callgroup=1 pickupgroup=1 context=alcatel channel = 32-46,48-62 loadzone=nl and in /etc/zaptel.conf: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46, 48-62 dchan=47 I am setting up an E1 crossover system for a customer, with a Siemens Hipath Officecom 150 system. And it's not working - I get a red alarm from the outside world, and a yellow from the PABX. Any ideas? We have tried E1 crossovers and straight cables on both connections, with no more luck. Frame type? Cabling? Timimg? Any ideas at all? Anyone? PaulH ___ --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 -- GATWORKS GmbH [EMAIL PROTECTED] Internetloesungen vom Feinsten Fon. +49 2166 9149-32 Fax. +49 2166 9149-10 ___ --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 - Tekelec T6000 (Vocaldata, voiss)
Does anyone have a working sip.conf for a SIP trunk to a Tekelec T6000 switch. I can get everything to work except the DTMF. The t6000 requires RFC1833 and I have that in the sip.conf but it still doesn't seem to work. Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --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] Early rtp bridge and reINVITE in 1.4b2
Hi, I just test the reinvite feature in this version and I realized that the SDP is changed in the early stages of the SIP call and thus * is not sending the reINVITEs. Is there any way to disable the early rtp bridge but still having the reINVITEs? (may be some parameter in sip.conf ). Let's say the device is behind a NAT router and knows its public IP, if the RTP port choosen by the NAT-router is not the same port the device set in the SDP then the early bridge is not going to work. I'm not asterisk savvy, but may I can comment something out in chan_sip.c Any ideas? Thanks for your help, Humberto __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Re: Voip Buster - CID
You can try VoipJet (http://www.voipjet.com)A simple configuration in you extensions.conf as below will solve your problem.exten = _X.,1,SetCIDNum(1341212)
exten = _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})- Buki
-- Forwarded message --From:Tomislav Parčina [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comDate:Thu, 28 Sep 2006 08:55:33 +0200Subject:[asterisk-users] Re: Voip Buster - CIDIn article
[EMAIL PROTECTED], [EMAIL PROTECTED] says... There are not many that will allow you to set your own CID even then they
normally ask for proof of the numbers you wish to use.Hi Chris!So, you are saying that I can't set outgoing CID number on Voip Buster? Do you know for any VoIP provider that allows that?
--Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148Mob.: +385(91)1212148SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hrhttp://www.lama.hr
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Re: [asterisk-users] Polycom 501 One-way Audio
Apologies around for posting HTML. My bad. --Paul Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. signature.asc Description: This is a digitally signed message part ___ --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] RPID
Thanks... I did some research and found that it's actually not what I was wanting (unless I missed something lol). I'm actually looking for a way to forward caller id information to the called party on a forwarded call. I may just need to dig deeper. On another note, I did find a patch in mantis that is considered experimental that does get it to where you can see the caller id of who you're calling based in the dialplan. Back to the drawing board though :) Aaron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Wednesday, September 27, 2006 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RPID DANIEL, AARON MATTHEW wrote: Has anyone successfully gotten rpid working between two phones through asterisk? Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 Aaron, RPID is supported in Asterisk but many phones do not support it. Try adding the following to sip.conf: sendrpid=yes trustrpid=yes If it is going to work with your phones, it will just work. If not, chances are your phone does not support RPID. You can always look at a SIP debug to make sure it is getting sent. Even if your phones do not support RPID, From: usually works just fine :). -- Kristian Kielhofner ___ --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] Polycom 501 One-way Audio
Check the handset cords. They can get loose and cause this exact issue. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Dugas Sent: Thursday, September 28, 2006 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom 501 One-way Audio I have a site running an up-to-date version of Asterisk from the 1.2 trunk. We have a dozen Polycom 501 units and one of them (none of the others) is having recurring one-way-audio problems. As Murphy's Law dictates, it's the bosses phone! The user gets a few calls a day where the caller can hear her fine but she hears dead silence. It happens when she calls out sometimes too. Even internal voicemail and extension-to-extension calls are affected. I just called her three times from another extension; the first two were affected, the third got through. None of the other units seem to have the problem. They're all running the same firmware and are loading central configs that are identical except for line-button text and registration info. I've been running * with lots of debug/verbose logging enabled and have yet to see it complain about anything when she reports the problem. I'm about to replace her phone with a spare to see if that fixes it. Wondering if anyone has seen something like this and might be able to tell me what to look for as a potential cause. TIA, Paul Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 One-way Audio
It looks like a NAT issue on RTP. How many ports are u running on rtp.conf? Are you using static or dynamic nat translation for rtp? Redouane De: Paul Dugas [mailto:[EMAIL PROTECTED] Envoyé: jeudi 28 septembre 2006 16:32À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: [asterisk-users] Polycom 501 One-way Audio I have a site running an up-to-date version of Asterisk from the 1.2 trunk. We have a dozen Polycom 501 units and one of them (none of the others) is having recurring one-way-audio problems. As Murphy's Law dictates, it's the bosses phone!The user gets a few calls a day where the caller can hear her fine but she hears dead silence. It happens when she calls out sometimes too. Even internal voicemail and extension-to-extension calls are affected. I just called her three times from another extension; the first two were affected, the third got through. None of the other units seem to have the problem. They're all running the same firmware and are loading central configs that are identical except for line-button text and registration info.I've been running * with lots of debug/verbose logging enabled and have yet to see it complain about anything when she reports the problem. I'm about to replace her phone with a spare to see if that fixes it. Wondering if anyone has seen something like this and might be able to tell me what to look for as a potential cause.TIA,Paul Paul DugasComputer Engineer Dugas Enterprises, LLC522 Black Canyon ParkCanton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 One-way Audio
On Thu, Sep 28, 2006 at 10:32:12AM -0400, Paul Dugas wrote: I have a site running an up-to-date version of Asterisk from the 1.2 trunk. We have a dozen Polycom 501 units and one of them (none of the others) is having recurring one-way-audio problems. As Murphy's Law dictates, it's the bosses phone! Does swapping the physical locations of the phones move the problem? Swapping IP addresses in the original locations? Did you mention whether they've all got identical FW versions? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] 7940 vs. 7941
Any pros / cons on getting one over the other ? I was wondering what themain differences were. Others have had better luck, but I haven't had any luck getting the 79x1 series to work with Asterisk. Maybe I just haven't invested enough time since I only have one 7961 and the rest are 7960s. ___ --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] quadbri + tdm400p + modem-fax
Paco Brufal wrote: On sep/28/2006, Steve Underwood wrote: Lots have tried it. it doesn't work. With Sangoma cards it will work? Thanks. Every time. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --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] T1 incoming connects, but no sound
Hi everybody, When I call my asterisk box, connected via a T1 line, it connects, logs various things, supposedly plays back the message defined in extensions.conf, and then disconnects. Seems all fine and dandy other than the fact that no sound is being heard on the phone placing the call. I'm upgrading my PBX from an intertel-axxess to asterisk. In zaptel.conf and zapata.conf I set the incoming T1 line exactly the same as the intertel box has it set: zaptel.conf: span=1,1,0,esf,b8zs em=1-8 fxsls=9-16 em=17-24 loadzone = us defaultzone=us zapata.conf: [channels] usecallerid=yes hidecallerid=no transfer=yes echocancel=yes echotraining=yes immediate=no ; span 1 group=1 context=from-ptsn signalling=em channel = 1-8,17-24 signalling=fxs_ls channel = 9-16 Ztcfg, zttool, and asterisk all give the green light on this configuration, but when an incoming call is received (haven't tested outgoing yet, one piece at time), the following is logged: Sep 27 17:39:30 VERBOSE[31181] logger.c: Asterisk Ready. -- Starting simple switch on 'Zap/1-1' Sep 27 17:39:35 DEBUG[31181] chan_zap.c: Exception on 17, channel 1 Sep 27 17:39:35 DEBUG[31181] chan_zap.c: Got event On hook(1) on channel 1 (index 0) Sep 27 17:39:35 DEBUG[31181] chan_zap.c: disabled echo cancellation on channel 1 Sep 27 17:39:35 WARNING[31181] chan_zap.c: getdtmf on channel 1: Success Sep 27 17:39:35 DEBUG[31181] chan_zap.c: Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Sep 27 17:39:35 DEBUG[31181] chan_zap.c: disabled echo cancellation on channel 1 Sep 27 17:39:35 DEBUG[31181] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Sep 27 17:39:35 DEBUG[31181] chan_zap.c: Updated conferencing on 1, with 0 conference users Sep 27 17:39:35 VERBOSE[31181] logger.c: -- Hungup 'Zap/1-1' No sound is ever heard on the calling phone, and the call is quickly terminated from the asterisk end. As I figured this was a configuration problem, I also tried zapata.conf and zaptel.conf as such: zaptel.conf: span=1,1,0,esf,b8zs fxsks=1-24 defaultzone=us zapata.conf: [channels] usecallerid=yes hidecallerid=no transfer=yes echocancel=yes echotraining=yes immediate=no ; span 1 group=1 context=from-ptsn signalling=fxs_ks channel = 1-24 Doing this caused the phone call to appear to be handled correctly, but still no sound was heard: Sep 27 17:43:52 VERBOSE[31391] logger.c: Asterisk Ready. -- Starting simple switch on 'Zap/1-1' Sep 27 17:43:57 NOTICE[31391] chan_zap.c: Got event 18 (Ring Begin)... Sep 27 17:43:57 VERBOSE[31391] logger.c: -- Executing Answer(Zap/1-1, ) in new stack Sep 27 17:43:57 DEBUG[31391] chan_zap.c: Took Zap/1-1 off hook Sep 27 17:43:57 DEBUG[31371] channel.c: Avoiding initial deadlock for 'Zap/1-1' Sep 27 17:43:57 DEBUG[31391] chan_zap.c: Enabled echo cancellation on channel 1 Sep 27 17:43:57 DEBUG[31391] chan_zap.c: Engaged echo training on channel 1 Sep 27 17:43:57 VERBOSE[31391] logger.c: -- Executing Playback(Zap/1-1, vm-goodbye) in new stack Sep 27 17:43:57 DEBUG[31391] channel.c: Scheduling timer at 160 sample intervals Sep 27 17:43:57 VERBOSE[31391] logger.c: -- Playing 'vm-goodbye' (language 'en') Sep 27 17:43:58 DEBUG[31391] channel.c: Scheduling timer at 0 sample intervals Sep 27 17:43:58 DEBUG[31391] channel.c: Scheduling timer at 0 sample intervals Sep 27 17:43:58 VERBOSE[31391] logger.c: -- Executing Hangup(Zap/1-1, ) in new stack Sep 27 17:43:58 VERBOSE[31391] logger.c: == Spawn extension (from-ptsn, s, 3) exited non-zero on 'Zap/1-1' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '(null)' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '(null)' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is 's' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is 'from-ptsn' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is 'Zap/1-1' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '(null)' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is 'Hangup' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '(null)' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '2006-09-27 17:43:57' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '2006-09-27 17:43:57' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '2006-09-27 17:43:58' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '1' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '1' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is 'ANSWERED' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is 'DOCUMENTATION' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '(null)' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '1159400632.0' Sep 27 17:43:58 DEBUG[31391] pbx.c: Function result is '(null)' Sep 27 17:43:58 DEBUG[31391] chan_zap.c: Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Sep 27 17:43:58 DEBUG[31391] chan_zap.c: disabled echo cancellation on channel 1 Sep 27 17:43:58 DEBUG[31391]
RE: [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]
OK, pardon my ignorance -- but what can you tune on such a system? How does Linux handle separate buses? Thanks. on Thursday 09/28/2006 Colin Anderson([EMAIL PROTECTED]) wrote I concur with your approach, but Tier 1 means as little here as it does when evaluating Internet backbone carriers. could you expand on what evaluation criteria you use? I'm going to be pre-speccing some stuff myself this month... Sorry I should have been more clear. A good Asterisk install needs a holistic approach to use a hippy dippy phrase. A Tier 1 server, which is a midrange to high end name brand server from the Big 3 (Dell, HP/Compaq, IBM, am I missing someone?) is usually highly optimized for bus bandwidth although that design was intended for a different use - usually massive disk I/O. As well, a Tier 1 server will have two seperate, independent PCI buses and this to me is a critical feature - it allows you to completely separate your TDM traffic from network, disk I/O etc. On my big production Netfinity, I took great care to ensure the Digium cards were all on their lonesome on a single bus, and everything else on the other bus. This is how I can run two TE110's in a single box with no problems. zttest does not give me 100% all the time, but on the other hand it *never* drops below 99.9987%, even under load. I selected this Netfinity because of the obvious care put into it's design, but the specs are unimpressive: quad Xeon 700's. CPU is over rated for Asterisk, IMO unless you are doing tons of transcoding and if you are doing that, then your design is flawed. Anyway, the holistic approach (to go on a small rant for the newbie lurkers) be summed up as follows: 1. Good box, see above 2. Good LAN - this is so critical and so often overlooked in the day and age of guys crimping their own cables and running $150 switches. You can't do that, and if you do, you do so at your own peril. Managed swiches, professional cable installation. This is not a problem for me since I *am* a professional cable installer but I have actually witnessed people making patch cables with a flat blade screwdriver and a hammer! 3. Tuning of the LAN - VLAN's are good. QoS packets are good. Switches that honor the QoS packets are good. 4. Handset selection - this is another biggie. I've selected Snom 360's, and yes they have warts, but they are feature rich for the price and Snom is really good about revising firmware. When you select handsets, GET YOUR USERS INVOLVED. 5. Tuning of Asterisk box itself - this cannot be under emphasized. This is a very important step and tuning methodologies vary according to distro, skill of the admin, and particular circumstances. I've learned *way* more than I ever wanted to about processor affinity sinc I started using Asterisk. 6. Termination of PSTN. Basically I would never do an Asterisk install where I was forced to do something stupid like aggregate a dozen Centrex lines or some mickey mouse deal with FXO ATA's or whatever except for a hobby or prototype install. PRI, BRI, IAX or SIP, don't mess around with anything else. 7. Relationship with provider. What is their SLA? Is it the incumbent or the clec? An incumbent will be more expensive and more difficult to deal with but they will tend to be more reliable. A clec will be cheaper and they will be way more accomodating but you will most likely not get five 9's from them. A VoIP provider should never be trusted, period. You will not get five nines from them, ever. Plan failover situations accordingly. 8. Plan plan plan plan. A good install of ANYTHING is 80% planning 20% doing it. What is your plan when your primary PSTN provider fails? What is your plan if your Asterisk box goes pear shaped? My dialplan can survive either PSTN, WAN or LAN failure (albeit with reduced functionality). I also keep a cold spare, an identically configured box that I can literally throw into the rack, turn it on, plug in the PRI's and no problem. 9. Internet bandwidth and latency. I am fortunate enough to have a great IP provider. Ask for demos - most guys will install a 90 day trial or something like that. Do not believe the brochure, get the product installed and put it under load. 10. Traffic prioritization at the IP demarc - total no brainer. 11. Constant, constant user feedback and remediation. If you are not talking to your users, your install will ultimately fail even if you have the best of everything. Underpromise and overdeliver. Never loose sight of the basics - they have to pick up the phone, and it has to work. Always. ___ --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 -- Your life is like a penny. You're going to lose it. The
Re: [asterisk-users] Asterisk with cisco 7935
Julien Goodwin wrote: Mine works fine with chan_sccp, and there's now SIP firmware out for it (which I haven't tried) I don't find it on the Cisco site. Do you have a link? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Tekelec T6000 (Vocaldata, voiss)
Matthew Crocker wrote: Does anyone have a working sip.conf for a SIP trunk to a Tekelec T6000 switch. I can get everything to work except the DTMF. The t6000 requires RFC1833 and I have that in the sip.conf but it still doesn't seem to work. RFC2833 not RFC1833 ___ --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] Voicemail callback bug?
Hello everyone, I'm having a problem with voicemail callback (option 3 after message, option 1 to send a reply). Here is what happens: -- Playing '/var/spool/asterisk/voicemail/default/400/INBOX/msg0001' (language 'en') -- Playing 'vm-prev' (language 'en') -- Playing 'vm-advopts' (language 'en') -- Playing 'vm-toreply' (language 'en') -- Playing 'vm-tohearenv' (language 'en') -- Playing 'vm-starmain' (language 'en') == Parsing '/var/spool/asterisk/voicemail/default/400/INBOX/msg0001.txt': Found -- Leaving voicemail for '3@' in context 'starbox_11' Sep 28 16:14:03 WARNING[1749]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '3' -- Playing 'vm-prev' (language 'en') -- Playing 'vm-advopts' (language 'en') -- Playing 'vm-toreply' (language 'en') msg0001.txt looks like this: [message] origmailbox=400 context=vm-in macrocontext= exten=vmu priority=106 callerchan=SIP/vm-082b9f78 callerid=Buck Aneer 300 origdate=Thu Sep 28 04:18:36 PM UTC 2006 origtime=1159460316 category= duration=7 No entry in voicemail config file for 3 leads me to think that Asterisk is parsing 300 from the callerid line above as 3, which obviously isn't correct. There is a 300 in the voicemail config file, there just isn't a three. The interesting thing is if you go the other direction (400 leaving vm for 300), Asterisk parses the callerid as 40 instead of 400. Closer, but still no cigar... Any thoughts? Thanks! -- Kristian Kielhofner ___ --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: Voip Buster - CID
Naija Man wrote:
You can try VoipJet (http://www.voipjet.com)
A simple configuration in you extensions.conf as below will solve your
problem.
exten = _X.,1,SetCIDNum(1341212)
exten = _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
1 is not valid as the first digit in a NANPA phone number. Only put
the area code and phone number in the Caller*ID Information,
___
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Re: [asterisk-users] RPID
Aaron Daniel wrote: Thanks... I did some research and found that it's actually not what I was wanting (unless I missed something lol). I'm actually looking for a way to forward caller id information to the called party on a forwarded call. I may just need to dig deeper. On another note, I did find a patch in mantis that is considered experimental that does get it to where you can see the caller id of who you're calling based in the dialplan. Back to the drawing board though :) Aaron Aaron, Usually, the way that I do that is to save CALLERIDNUM in a variable someplace when a call first comes in. Then later on, if you need to forward the call, you can set the CALLERID to your custom variable (I use KKFROMCID) to always set the original callerid. Works quite well in macros. -- Kristian Kielhofner ___ --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] Good Book on Asterisk
Michel Vaillancourt wrote: Norbert Zawodsky wrote: Hi everybody! I have some Linux experience but I'm completely new to asterisk. I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk (1.2.12) preinstalled and some basic configuration (Wiht a few extensions). Now I want to implement something more, fox example voicemail (storing voicemail data in an extern mysql DB) and so on. And since I don't want to waste your time with stupid questions ... can someone of you recommend a really good book on Asterisk? (To buy or for download) ... or another online source of information which would be helpful for someone like me? I searched Amazon with Asterisk and got 21 hits.. Thanks Norbert Hi, Norbert ... The O'Reily Book for Asterisk: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Enjoy! Thanks for the book. I read it last night (or nearly all of it) and now i think i understand a bit more. But now the next question: Where can I find the documentation of the applications and functions I can use in the dialplan? (For example how to use the mysql add-on, ...) Thanks Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 One-way Audio
Had the same issue and it was the handset cord. Rick From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, September 28, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 501 One-way Audio Check the handset cords. They can get loose and cause this exact issue. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Dugas Sent: Thursday, September 28, 2006 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom 501 One-way Audio I have a site running an up-to-date version of Asterisk from the 1.2 trunk. We have a dozen Polycom 501 units and one of them (none of the others) is having recurring one-way-audio problems. As Murphy's Law dictates, it's the bosses phone! The user gets a few calls a day where the caller can hear her fine but she hears dead silence. It happens when she calls out sometimes too. Even internal voicemail and extension-to-extension calls are affected. I just called her three times from another extension; the first two were affected, the third got through. None of the other units seem to have the problem. They're all running the same firmware and are loading central configs that are identical except for line-button text and registration info. I've been running * with lots of debug/verbose logging enabled and have yet to see it complain about anything when she reports the problem. I'm about to replace her phone with a spare to see if that fixes it. Wondering if anyone has seen something like this and might be able to tell me what to look for as a potential cause. TIA, Paul Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --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: Voip Buster - CID
At 11:55 PM 9/27/2006, you wrote: So, you are saying that I can't set outgoing CID number on Voip Buster? Do you know for any VoIP provider that allows that? I use SellVOIP and Voxee which both seem to allow that. Ira ___ --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] Cisco CAll Manger and H323
Hi, I recently had to hook up to Cisco Call Manager 4.1.3, and it only supports H323. SO I used ooh323, and a strange thing happens. When a Cisco IP user calls from his phone, the call gets sent from Call Manager to Asterisk, but the phone will ring once only, then it seems asterisk will drop the call, and int he debug it says: stopped from reciving frames from OOH323/cisco , bridging is being stopped. What is wrong? What RTP ports must I be using? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 - Tekelec T6000 (Vocaldata, voiss)
On Thu, 28 Sep 2006, Matthew Crocker wrote: Does anyone have a working sip.conf for a SIP trunk to a Tekelec T6000 switch. I can get everything to work except the DTMF. The t6000 requires RFC1833 and I have that in the sip.conf but it still doesn't seem to work. I get my incoming calls from a Tekelec. The SIP User-Agent says Tekelec-7000/r4.0. I don't know how different a 6000 configuration is compared to a 7000 configuration. Here's my sip.conf in its entirety: [general] disallow = all allow = ulaw allowguest = yes allguest= yes context = block-ani host= dynamic ; ; for debugging ; dumphistory = yes ; recordhistory = yes ; sipdebug= yes ; ; (end of /etc/asterisk/sip.conf) The application involves a bunch of DTMF as callers jump around the dial plan a lot. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 One-way Audio
On Thu, 2006-09-28 at 16:34 +0100, Redouane Doumer wrote: Are you using static or dynamic nat translation for rtp? No NAT. Everyone on the same switch and subnet. -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. signature.asc Description: This is a digitally signed message part ___ --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] AstriCon 2006 Reminder / Hotel Selling Out
Greetings Asterisk Users, Only a few weeks remain until AstriCon 2006 and we are getting very close to selling out of space at the hotel. If you want to attend and take advantage of the discount rate, please register and book your room ASAP. Register for AstriCon: http://www.astricon.net Register for a room:http://www.starwoodmeeting.com/Book/astricon If you have any questions, please email us at [EMAIL PROTECTED] or call us at +1 816 256 8916. Thanks, Steve -- Steven Sokol AstriCon 2006 Asterisk Training: http://www.sokol-associates.com/ AstriCon 2006: http://www.astricon.net/ ___ --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] Polycom Queues, Login, Logout etc
Someone was working on integrating the Polycom Agent Login/Logout/status functionality withAsterisk. Who was that? Do we know if it made it into Asterisk 1.4? I can perform a login on the phone, and Asterisk gets a subscription for it... Peer User Call ID Extension Last state Type Mailbox xxx.yyy.128.18 80014101 dc665304-c8 [EMAIL PROTECTED] Idle pidf+xml none 1 active SIP subscription However, when I hit the 'Unvail' or 'Avail' soft key, Asterisk responds with a '489 Bad Event' to the update. Doug. ___ --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] Yet another processor question
Here's my setup, I'd like to hear what everyone suggests is the best route: I have three sites, a main and two outlying buildign, each has a Mitel SX200 system with a PRI interface. The outlying sites currently each have a PTP T1 line back to the main site. Right now I have a P2 700mhz Asterisk server at each site with a Digium 4 port T1 card. Each Asterisk server sites between its site's external T1 line and Mitel switch. Each Asterisk box is also connected to the PtP T1 line. I am using ZaptelPPP to send data on 12 channels, and the rest are left free for voice calls. The Asterisk boxes do some limited LCR, and route outbound calls over the PtP T1 to whichever site has the most affordable outbound calling plan for the call in question, as well as sending site to site calls over the PtP. It works, but using ZaptelPPP is a bit of a pain, and it makes upgrading a hassle, since I haven't found ppp packages for a recent OS so now I have to have the compiler on each box, etc. Making changes to the asterisk servers means taking down the data connections as well, which is unpopular with the users. One site has never worked, despite having identical configuration to the other site. The data bandwidth is always limited to 12 channels, even when the voice channels are idle. So what route would everyone recommend? I can move the PtP connections to independent WAN routers with QOS, then the Asterisk boxes will need to do TDM to SIP and back. Right now everything is TDM switching. Will the slow P2s have the horsepower to handle 12 or 13 TDM to SIP calls? Or should I stick to my current system and just deal with the upgrade hassle? (I wouldn't even upgrade, except that Caller ID Name information doesn't work on my current 1.0 version) Mark ___ --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] T1 incoming connects, but no sound
Nathan Bell wrote: extensions.conf: [from-ptsn] exten = s,1,Answer() exten = s,2,Playback(vm-goodbye) exten = s,3,Hangup() You might try adding a wait(3) command after the answer. Some analog lines do not pass audio immediately after being answered. (Something to do with how toll processing is handled) Mark Testing my setup from a channel bank seems to work just fine (slightly different zaptel.conf and zapata.conf). ___ --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] RPID
Hi, Here is how I have it working: If Alice calls Bob and Bob's phone diverts the call to Carol. You want Bob to pay for the call and the callerid shown to Carol to be 'Alice' On Bob's server exten _X.,1,sipaddheader(Divertion:[EMAIL PROTECTED];user=phone The proxy that routes bob's call to Carol will then charge Bob for the call and the From: field will be Alice If you are an ITSP using Asterisk then you must look for the 'Divertion' header in incoming SIP invite's yourselves with a sipheader(Divertion) command I have this working in a few different scenarios I think that the right thing todo would be setting the RDNIS if the 'Divertion' is present on inbound side but I am not sure about this so I am using a private variable and doing this outside the SIP channel b.r. Freddi Thanks... I did some research and found that it's actually not what I was wanting (unless I missed something lol). I'm actually looking for a way to forward caller id information to the called party on a forwarded call. I may just need to dig deeper. On another note, I did find a patch in mantis that is considered experimental that does get it to where you can see the caller id of who you're calling based in the dialplan. Back to the drawing board though :) Aaron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Wednesday, September 27, 2006 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RPID DANIEL, AARON MATTHEW wrote: Has anyone successfully gotten rpid working between two phones through asterisk? Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 Aaron, RPID is supported in Asterisk but many phones do not support it. Try adding the following to sip.conf: sendrpid=yes trustrpid=yes If it is going to work with your phones, it will just work. If not, chances are your phone does not support RPID. You can always look at a SIP debug to make sure it is getting sent. Even if your phones do not support RPID, From: usually works just fine :) . -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] You are not the next caller
Ok... I have heard this on Digium's PBX in the past, but can't seem to find it anymore. There was an IVR that you could dial into and Allison had recorded one of the funniest messages I have ever heard... you are not the next caller, hang up not, spend time with your children... it was hilareous... Does anyone have that recording? or know where I can find it? Sean ___ --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 with cisco 7935
Doug Lytle wrote: Julien Goodwin wrote: Mine works fine with chan_sccp, and there's now SIP firmware out for it (which I haven't tried) I don't find it on the Cisco site. Do you have a link? Doug I saw the 7936, didn't see a 7935 though. ___ --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: Voip Buster - CID
Eric ManxPower Wieling wrote:
Naija Man wrote:
You can try VoipJet (http://www.voipjet.com)
A simple configuration in you extensions.conf as below will solve your
problem.
exten = _X.,1,SetCIDNum(1341212)
exten = _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
1 is not valid as the first digit in a NANPA phone number. Only put
the area code and phone number in the Caller*ID Information,
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We've been getting a lot of telemarketing calls with the 1 at the
beginning of the number. I wonder if these calls are from Naija Man's
company? :-)
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