Re: [asterisk-users] _ALERT_INFO replacement in 1.4?
12 mar 2007 kl. 23.33 skrev Nikhil Jogia: Bruce Reeves wrote: Does SIPAddHeader(Alert-Info:) not do it? No, but from another thread, setting the _SIPADDHEADER variable works. You misunderstand. The prefered way is to use SIPAddHeader(Alert- Info: slakfj aslkfjaklsdf) But in the situation of the manager call setup used in the other mail, that was not possible. /O ___ --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: AMI - DBPut
Lee Jenkins wrote: Try putting quotes around the value. I played with it a while back only a little, but I can't remember if quotes did it or I ended up having stripping the quotes off myself when I retrieved the value ... My first mail was copy/paste, so I'm positive I didn't make any error typing. Now I have tried again with the similar input and it works. Action: DBPut Family: checkin Key: 319 Val: yes Response: Success Message: Updated database successfully I really don't know why it didn't work yesterday. Has anybody head similar error/problem? Now I wonder, is it stable enough for production use? -- Tomislav Parcina [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] Update Asterisk 1.2.12 to 1.4.1 ?
Hi i have a big change or bproblems to update a asterisk 1.2.12 server to asterisk 1.4.1 ? Thanks bye ___ --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] Update Asterisk 1.2.12 to 1.4.1 ?
13 mar 2007 kl. 09.53 skrev Noc Phibee: Hi i have a big change or bproblems to update a asterisk 1.2.12 server to asterisk 1.4.1 ? There won't be any problems if you take some time to read the available documentation to see what changes you need to do in your configuration. Make sure you read UPGRADE.txt and the doc/ and configuration files. They all contain a lot of information that is very useful. Regards, /Olle ___ --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: warble on registration?
Ken D'Ambrosio wrote: Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Are you talking about the warble that the phone makes every sip registration if there are messages waiting??? 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] Call load balancing
On 9 Mar 2007, at 17:51, Octavio Ruiz (Ta^3) wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Another approach: what about load-balance (in terms of redundancy and scalability) the AGI app's and just the AGIs with FastAGI? So your IVR application can be separated from your * boxes and they (the * boxes) dont have to ve overloaded with your AGI apps. Your head system receive the two PRIs and in dial-plan logic you can (maybe using RANDOM() or something more deterministic like a counter) Assuming the head box takes all the calls you could just use setgroup and getgroupcount on the pri box and use them to count the calls. Using groups has the advantage of dealing with hangup right. The only tricky bit would be implementing min(group) in the dialplan. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Single sign on PC + phone?
On Mon, 2007-03-12 at 22:12 -0700, Trevor Peirce wrote: Patrick wrote: Hi all, Does anyone have any experience with creating a Single sign on (SSO) concept where if someone logs in on their PC the phone next to that PC is also automatically assigned to that user? Yup, I've done this on a small proof-of-concept scale. I basically created a script that runs on login which updates a MySQL database. This database is in turn queried by the Realtime application so the dial plan logic can route the call accordingly. An expansion of this idea will let you also manage outgoing caller id and voicemail notifications, but those weren't needed in my project. This was a single day project with Fedora Core 5. Thanks for the info Trevor. Was your proof of concept also with Windows PCs or *nix PCs? I haven't played with realtime yet so I might be in for a bit of a learning curve. Regards, Patrick ___ --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] MusicOnHold stops after upgrade from 1.4.0 to 1.4.1
Hello I have following problem. After upgrade from 1.4.0 to 1.4.1 my musiconhold stops immediately after start. Bellow some logs from 1.4.0 and 1.4.1 (same configs and situations) First, the one from 1.4.0 (everything works) [Mar 12 13:44:00] -- Executing [EMAIL PROTECTED]:1] SetMusicOnHold(SIP/1036690-b74004b8, mymusic) in new stack [Mar 12 13:44:00] -- Executing [EMAIL PROTECTED]:2] Dial(SIP/1036690-b74004b8, SIP/halo/0502xxx|20|mgA(pol800)) in new stack [Mar 12 13:44:00] -- Called halo/0502xxx [Mar 12 13:44:00] -- Started music on hold, class 'mymusic', on SIP/1036690-b74004b8 [Mar 12 13:44:00] -- SIP/halo-081c3340 is making progress passing it to SIP/1036690-b74004b8 and now from 1.4.1 Mar 12 13:45:21] -- Executing [EMAIL PROTECTED]:1] SetMusicOnHold(SIP/1036690-081cc0b8, mymusic) in new stack [Mar 12 13:45:21] -- Executing [EMAIL PROTECTED]:2] Dial(SIP/1036690-081cc0b8, SIP/halo/0502xxx|20|mgA(pol800)) in new stack [Mar 12 13:45:21] -- Called halo/0502xxx [Mar 12 13:45:21] -- Started music on hold, class 'mymusic', on SIP/1036690-081cc0b8 [Mar 12 13:45:21] -- Call on SIP/halo-081de798 left from hold [Mar 12 13:45:21] -- Stopped music on hold on SIP/1036690-081cc0b8 [Mar 12 13:45:21] -- SIP/halo-081de798 is making progress passing it to SIP/1036690-081cc0b8 What that 'Call on SIP/halo-081de798 left from hold' means ?? I have no clue... sh0t ___ --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: SIP unicode support ?
KD == Klaus Darilion [EMAIL PROTECTED] writes: KD Hi! Is there unicode support in Asterisk for SIP? E.g. How can I KD have a displayname with special characters? KD E.g. if I want to have the Umlaut ä in the display name: KD callerid=Jeff Gräser 11 Is your sip.conf UTF-8-encoded? /Benny ___ --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: Number of SIP messages per minute
Mark Davies wrote: I’ve just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me. Put Ethereal and count :) -- Tomislav Parcina [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] CDR and CallerID
Hi, Is there a way to unlink CallerID and the CDR values? I'd like my CDR to have, in the src column, the extension of the person calling, for my records (let's say 201). But if that person is calling outside the company, I want the callerid to show 555-555-1234). At first sight, the two values must be identical. Is there any way to change that? Mike ___ --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: 1.4 compile issue
Wai Wu wrote: I am use Fedora 3, and run into a 1.4 compile issue. I recommend you to start using Cent OS 4.4 - it's basically RHEL. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTalk/Jabber passing audio in 1.4.1!
Dear Lewis, Can you please post you gtalk.conf and jabber.conf for me? I also make it under Fedora Core 6. But I got no audio at all. I use X-Lite as SIP client (under NAT). 2007/3/7, Ronald Lewis [EMAIL PROTECTED]: I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got two-way audio between Google Talk and Asterisk! This IS an exciting moment today in VoIP! This is just GREAT! - Ronald Lewis http://ronaldlewis.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 -- Best Regards Charles ___ --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] Number of SIP messages per minute
That does sound low, especially if you have multiple devices behind a NAT. I have customers with 8 analog lines going into their analog phone system and just have 4 ATAs with 2 lines each. Of course, all of this traffic would seem to come from the same IP! On 3/8/07, Mark Davies [EMAIL PROTECTED] wrote: Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me. Thanks in advance, 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 ___ --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 scenario
Hi all, i need help to implement a voicemail scenario. What i am trying to do is the following. user X dials a direct access for user Y voicemail and is asked to enter a number (e.g 12345678) and then leaves a message. Then asterisk sends a notification with attachement. The problem is that i need the number entered (e.g 12345678) in the subject. Is that possible. thx in advance. Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --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] French PRI channel - exact signaling used
hello, We encountered signaling problem with a french national carrier. They ask us, which signaling is configured on our single E1. I need to know if it's ETSI, VN4 or VN6. I know what ccs, and hdb3 mean but I do not succeed to make the link between the signaling type. I searched through RFC Q.921 and Q.931 It would be great to obtain some help. cedric /etc/zaptel.conf # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan= 16 # Span 2: WCTDM/0 Wildcard TDM400P REV I Board 1 fxoks=32 fxoks=33 fxoks=34 fxoks=35 # Global data loadzone= fr defaultzone = fr /etc/asterisk/zapata.conf [channels] language=fr context=fromE1 switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=800 relaxdtmf=yes rxgain=-3.0 txgain=-4.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived accountcode=carteE1 adsi=no busydetect=no callprogress=no musiconhold=vt channel = 1-15,17-31 ___ --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 many outgoing phone line/voip account do I need?
On 3/12/07, Dave Cotton [EMAIL PROTECTED] wrote: On Mon, 2007-03-12 at 20:52 +1100, Paul Hales wrote: More importantly, how many calls per day and how long per call. Then you can figure out the other bits. He wants to make 50 simultaneous calls. What difference does the length and frequency make. His vindictive dialer isn't playing while it is listening to rings or busy signals. So there is an impact on CPU usage from the length of time it takes the average victim to hang up. -HJC ___ --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] IAX2 Question (Asterisk 1.4 tarball)
I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the following message on asteriskm's cli: [Mar 13 11:34:33] NOTICE[6024]: chan_iax2.c:7840 __iax2_poke_noanswer: Peer 'asterisk1' is now UNREACHABLE! Time: 0 [Mar 13 11:36:13] WARNING[6029]: chan_iax2.c:3792 iax2_send: No private structure for packet? The warning repeats every 30 seconds, what am I doing wrong? Asteriskm config: **iax.conf** [general] bindaddr=192.168.0.160 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asterisk1] type=peer username=asteriskm auth=plaintext secret=asgard host=192.168.0.161 qualify=yes **extensions.conf** [general] [1ST-T1] exten = _X,1,AGI(rexx.agi) exten = 12345,1,Dial(IAX2/asterisk1/80483) exten = 12345,n,Hangup() Asterisk1 config: **iax.conf** [general] bindaddr=192.168.0.161 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asteriskm] type=user context=incoming-iax auth=plaintext secret=asgard host=192.168.0.160 qualify=yes trunk=yes **extensions.conf** [general] [incoming-iax] exten = _X,1,AGI(rexx.agi) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How many outgoing phone line/voip account do I need?
His vindictive dialer isn't playing while it is listening to rings or busy signals. Forgive my ignorance, but what on earth's a vindictive dialler? Is it one with a strong sense of revenge? :-) Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP hardphones with good jitter tolerance
Greetings list, Quite a few of our users seem to be experiencing poor voice quality when they're using internet connections over which we have little or no control (i.e. they're using their own router with no QoS, etc.). Some of these connections are giving a qualify time within asterisk of 130ms+. Are there any recommendations as to phones with particularly good buffering that might iron out at least some of the poor network performance? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] FW: Seamless Multi Office Asterisk Deployment
For startes I will keep it on the list and we can discuss some major concepts, and I will possibly make some contact off list later for the nitty-gritty :) In-reply to Steve: I did have a look at the bicomsystems product and it does appear to do everything I am looking for. However, I have looked in to vendor systems and have decided to go with an Asterisk system. Hench asking for assistance on the Asterisk mailing list ;) On the discussion at hand: At this time I am not going to worry about the QoS with my T1 network lines, I have been wondering what the quality will be like. I do not plan to have more than maybe three calls on a line at peak times. But I know that there will be more in the future. I am working with a total employee base of around 30, and the remote offices have two to four employees at a time, not a huge traffic demand. What I am most curious about at this time is the methods used to move from server to server. *Ideally* I would like to sit down at a phone, enter my extension/password and have that phone ring as my extension. Essentially, I would like a log in system on the phone. This presents me with two issues: I have to make my phones allow simple logon as a SIP device, and I need to get my credentials to move between Asterisk servers. What methods have others used, or where should I look for more information? At this point I have two Polycom phones (430 and 501) for testing, they seem to be talked about as very flexible. If they will not allow me to add a user friendly login prompt, maybe I need to find alternatives though. But this is the Asterisk list and I don't want to go too far off topic, so the main concern is how I would synchronize my information between asterisk servers. One final topic on this message I would like to cover is time frame. I am thinking maybe around 6 months to have at least a partial functioning system up and tested. By partial I mean deployable with a basic infrastructure feature set. I don't know if this is too little time or too much time. My co-workers are excited about what Asterisk has to offer. Any other thoughts on time frames? P.S. I want to thank everyone who replied so quickly, surprised my co-worker and I :D -- Thanks, Brandon Comouche IT Administrator Sno Falls Credit Union -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Monday, March 12, 2007 9:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment I'm more then happy to share my experiences with anyone, there is just a lot to be said about the things Brandon is trying to accomplish. Take the automatic fail over he mentioned, there are a number of ways to do that and everyone has an opinion. I just want make myself available to help other get from playing with Asterisk like I did to really putting it to use so that people sit back and say wow, my cisco/avaya/nortel can't do that. On 3/12/07, Sean Bright [EMAIL PROTECTED] wrote: Why does everyone want to go off-list? Is this not information that could benefit others? On 3/12/07, Bruce Reeves [EMAIL PROTECTED] wrote: Brandon Your on the right track with what is can do. It will also be good to look into what kind of QOS you can do on the T-1 connections between offices. I have an 8 office setup similar to this and many of your goals I have achieved and would be glad to offer ideas and such if you want to email me off list. On 3/12/07, Brandon Comouche [EMAIL PROTECTED] wrote: Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone - hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers J Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) -
Re: [asterisk-users] New to Asterisk
On Mon, 2007-03-12 at 23:51 +0400, NetSys Admin wrote: Hi everyone, I'm completely new to Asterisk and before I buy any card, I would like to ask for some information. 1. We'll be using analog PSTN phone lines. Is there anything that I should ask the telecom company before I buy the card? What I mean is whether the card will be compatible with the line? In the US, and several other countries, you are OK, the card is compatible. 2. What about the hardware on the PC? I will be using at least a Pentium 3 with a 600 or 700 MHz processor with at least 256 MB. Is there a way to know how much traffic or calls it can handle? Sound OK to me. 3. Suppose I buy a TDM04B card. It has 4 FXO modules and 0 FXS module. Later I decide that I need a TDM13B configuration. Can I just buy 1 FXS S110M module and replace one existing FXO module myself and reconfigure Asterisk? Yes, you could. I have configured systems with 2 TDM 4 cards, no problem. I have also put together a system with 1 tdm 4 card, and two old (100p) FXO cards, worked fine. 4. Does fax work fine with Asterisk? Should I use one FXS module for each fax machine? I've had good luck. I set up the main context with the fax extension, and dial my fax machine, for automatically routing incoming fax calls to the fax machine. Works great. 5. Is the power connector on the card identical to the power connectors inside PCs? Yes, the standard 4-pin power connector you would use for disk drives, cd drives, fans, etc. Thank you for any help. smime.p7s Description: S/MIME cryptographic 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
Re: [asterisk-users] Call Back
Ivo Zivkov wrote: Sorry, I can only give you a general outline, because the code is proprietary. Call anywhere *from* anywhere... for just 12 cents a minute! (Some restrictions apply; see 47 page contract for details) -Stephen- ___ --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] DST and VM timestamp
Who is tired of dealing with DST changes? I have asterisk running on FC4, FC4 has been patched and shows the correct MDT timezone and time. Email notifications of voicemail show the message time an hour early (standard time, not daylight). This si the time in the message body, not the email delivery time, so it is coming form asterisk wrong. I did a reload after correcting the time/timezone, but not a restart. Does the system need to be rebooted (or asterisk restarted?) or is there another way to get the application in sync with the OS. Could I be missing something else? 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] French PRI channel - exact signaling used
can you tell me about your physical layer cable.. i know that in frensh (I m talking about France Telecom) that they use 1,1,0,ccs,hdb3,crc4 and euroisdn pri_cpe 2007/3/13, Cedric MILLET [EMAIL PROTECTED]: hello, We encountered signaling problem with a french national carrier. They ask us, which signaling is configured on our single E1. I need to know if it's ETSI, VN4 or VN6. I know what ccs, and hdb3 mean but I do not succeed to make the link between the signaling type. I searched through RFC Q.921 and Q.931 It would be great to obtain some help. cedric /etc/zaptel.conf # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan= 16 # Span 2: WCTDM/0 Wildcard TDM400P REV I Board 1 fxoks=32 fxoks=33 fxoks=34 fxoks=35 # Global data loadzone= fr defaultzone = fr /etc/asterisk/zapata.conf [channels] language=fr context=fromE1 switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=800 relaxdtmf=yes rxgain=-3.0 txgain=-4.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived accountcode=carteE1 adsi=no busydetect=no callprogress=no musiconhold=vt channel = 1-15,17-31 ___ --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] How many outgoing phone line/voip account do I need?
On 3/13/07, Chris Bagnall [EMAIL PROTECTED] wrote: His vindictive dialer isn't playing while it is listening to rings or busy signals. Forgive my ignorance, but what on earth's a vindictive dialer? Is it one with a strong sense of revenge? :-) A normal predictive dialer determines from agent behavior when will be the most convenient time to deliver the next call to them. A vindictive dialer uses arcane arts to determine the least convenient time to deliver the call to the target. Is it when they are about to sit down for dinner, when they are about to step out or when they are taking a bath? Many factors have to be adjusted to maximize the inconvenience of the call. The dinosaur telephone companies are the main users, but the free vacation seminar companies are stepping up their deployments. Personally I never answer calls from area code 666 anymore. ;-) -HJC ___ --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] 120 concurrent ZAP connections in asterisk open edition. Is that possible?
Hi all, In your experience, what is the maximum number of *concurrent* zap channels that you've ever tried with one box of Asterisk open edition? In my case, the max that I've tried was 63 simultaneous connections in a Quad T1/E1 card installed on a Intel Pentium D 3.4Ghz 2GB ram system. Your comments will be really appreciated. Regards, Héctor. ___ --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] great problem with sounds and ztdummy
My solution. With Zaptel 1.4 I change ztdummy.c comment lines 47 to 56, for rtc config. Compile and without rtc module, load ztdummy. It work good with usbcore and uhci_hcd modules. I have installed libusb-dev for my debian etch system. Now, my kernel is 2.6.20.2, but it work good with 2.6.18 Regardsss Germán Aracil Boned escribió: Hello System: Debian etch with kernel 2.6.18-4-686 or 2.6.18 custom. Asterisk Version: SVN-branch-1.4-r55483M Zaptel Version: SVN-branch-1.4-r2302 modules all ok in compilation time. And modules loaded: ztdummy 5928 0 rtc13364 1 ztdummy zaptel181540 1 ztdummy crc_ccitt 3200 1 zaptel In /dev/zap directory I have: crw-rw 1 root dialout 196, 254 2007-03-13 08:16 channel crw-rw 1 root dialout 196, 0 2007-03-13 08:16 ctl crw-rw 1 root dialout 196, 255 2007-03-13 08:16 pseudo crw-rw 1 root dialout 196, 253 2007-03-13 08:16 timer crw-rw 1 asterisk asterisk 196, 250 2007-03-13 08:16 transcode (Asterisk runing with user and group root) All ok, no error messages, but when I call and play backgroud or speak, asterisk do not play nothing. I can call to meetme, see: -- Executing Goto(SIP/5060-081e9db0, pbx9|10|1) -- Goto (pbx9,10,1) -- Executing Answer(SIP/5060-081e9db0, ) -- Executing meetme(SIP/5060-081e9db0, |iMs) -- SIP/5060-081e9db0 Playing 'conf-getconfno' (language 'es') But can't get sound. If I quit ztdummy module meetme don't work, but I can get sound. Computer as Dell server. Any idea ? very thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DST and VM timestamp
Damon Estep wrote: Who is tired of dealing with DST changes? I have asterisk running on FC4, FC4 has been patched and shows the correct MDT timezone and time. Email notifications of voicemail show the message time an hour early (standard time, not daylight). This si the time in the message body, not the email delivery time, so it is coming form asterisk wrong. I did a reload after correcting the time/timezone, but not a restart. Does the system need to be rebooted (or asterisk restarted?) or is there another way to get the application in sync with the OS. Could I be missing something else? Thanks! At the minimum you'll need to restart asterisk. The safest bet is to reboot the computer. Each app usually gets its timezone info when it's started, so to get each program running on the machine to get the new DST info each process will need to be restarted (ie a reboot). -Dave ___ --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] great problem with sounds and ztdummy
The problem is rtc module. My servers don't have a standard pc chip for it. I like a ztdummy working with genrtc. Exist this option ? Now My ztdummy work with usb clock Germán Aracil Boned escribió: And If I execute: ./zttest -v I can see: Opened pseudo zap interface, measuring accuracy... But, command don't show nothing. If I press ctrl+C after +-30 seconds: --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 I think, ztdummy don't work good.. ¿? Where is the problem ?¿ Germán Aracil Boned escribió: Hello System: Debian etch with kernel 2.6.18-4-686 or 2.6.18 custom. Asterisk Version: SVN-branch-1.4-r55483M Zaptel Version: SVN-branch-1.4-r2302 modules all ok in compilation time. And modules loaded: ztdummy 5928 0 rtc13364 1 ztdummy zaptel181540 1 ztdummy crc_ccitt 3200 1 zaptel In /dev/zap directory I have: crw-rw 1 root dialout 196, 254 2007-03-13 08:16 channel crw-rw 1 root dialout 196, 0 2007-03-13 08:16 ctl crw-rw 1 root dialout 196, 255 2007-03-13 08:16 pseudo crw-rw 1 root dialout 196, 253 2007-03-13 08:16 timer crw-rw 1 asterisk asterisk 196, 250 2007-03-13 08:16 transcode (Asterisk runing with user and group root) All ok, no error messages, but when I call and play backgroud or speak, asterisk do not play nothing. I can call to meetme, see: -- Executing Goto(SIP/5060-081e9db0, pbx9|10|1) -- Goto (pbx9,10,1) -- Executing Answer(SIP/5060-081e9db0, ) -- Executing meetme(SIP/5060-081e9db0, |iMs) -- SIP/5060-081e9db0 Playing 'conf-getconfno' (language 'es') But can't get sound. If I quit ztdummy module meetme don't work, but I can get sound. Computer as Dell server. Any idea ? very thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?
On Tue, Mar 13, 2007 at 12:18:50PM -0500, Héctor Maldonado wrote: Hi all, In your experience, what is the maximum number of *concurrent* zap channels that you've ever tried with one box of Asterisk open edition? With Zaptel, the limit is pretty clear: the number of channels your hardware supports... 120 is the capacity of a quad E1 card, or (if I may pitch our own hardware) 4 Astribank 32 units. A decent system that does not have much transcoding conferencing or other types of complicated processings should have no problem with such a load. In my case, the max that I've tried was 63 simultaneous connections in a Quad T1/E1 card installed on a Intel Pentium D 3.4Ghz 2GB ram system. That's half the capacity of the quad E1. What exactly did you try? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to match wild card inside a GoToIf?
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3)
exten = s,2,Hangup
Any ideas?
Regards,
Ricardo.
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RE: [asterisk-users] IAX2 Question (Asterisk 1.4 tarball)
The communication problem boiled down to iptables rules, but I'm still getting the No private structure for packet? error message. It doesn't seem to cause any problems and only occurs when an IAX2 peer has been unavailable for at least three minutes, but I would like to know why it happens if anyone knows. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Tuesday, March 13, 2007 11:38 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] IAX2 Question (Asterisk 1.4 tarball) I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the following message on asteriskm's cli: [Mar 13 11:34:33] NOTICE[6024]: chan_iax2.c:7840 __iax2_poke_noanswer: Peer 'asterisk1' is now UNREACHABLE! Time: 0 [Mar 13 11:36:13] WARNING[6029]: chan_iax2.c:3792 iax2_send: No private structure for packet? The warning repeats every 30 seconds, what am I doing wrong? Asteriskm config: **iax.conf** [general] bindaddr=192.168.0.160 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asterisk1] type=peer username=asteriskm auth=plaintext secret=asgard host=192.168.0.161 qualify=yes **extensions.conf** [general] [1ST-T1] exten = _X,1,AGI(rexx.agi) exten = 12345,1,Dial(IAX2/asterisk1/80483) exten = 12345,n,Hangup() Asterisk1 config: **iax.conf** [general] bindaddr=192.168.0.161 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asteriskm] type=user context=incoming-iax auth=plaintext secret=asgard host=192.168.0.160 qualify=yes trunk=yes **extensions.conf** [general] [incoming-iax] exten = _X,1,AGI(rexx.agi) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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 ___ --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] cisco sip firmware update for cisco 7970
I simply called the vendor I bought it from. Myriad. Call Andy: (212) 366-6996 x111 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Saturday, February 24, 2007 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco sip firmware update for cisco 7970 Tim Connolly wrote: You can buy smartnet on a single phone for something like $8 a year. This will get you in legally. Any idea about how specifically to get such a contract? It is rumored to be pretty tricky. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to match wild card inside a GoToIf?
Try
exten = s,1,GotoIf($[${MACRO_EXTEN} = 220408XXX]?2:3)
exten = s,2,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Tuesday, March 13, 2007 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to match wild card inside a GoToIf?
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3) exten = s,2,Hangup
Any ideas?
Regards,
Ricardo.
___
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Re: [asterisk-users] DST and VM timestamp
On Tue, Mar 13, 2007 at 10:44:08AM -0600, Damon Estep wrote: Who is tired of dealing with DST changes? I have asterisk running on FC4, FC4 has been patched and shows the correct MDT timezone and time. Email notifications of voicemail show the message time an hour early (standard time, not daylight). This si the time in the message body, not the email delivery time, so it is coming form asterisk wrong. Email in the headers has a timezone information in it. If you fixed the system time by etting the clock, rather than fixing the timezone definitions, you may get errors. OTOH, the recipient of the email may have done that error. I did a reload after correcting the time/timezone, but not a restart. No reload or restart should be needed. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] great problem with sounds and ztdummy
On Tue, Mar 13, 2007 at 06:25:37PM +0100, Germán Aracil Boned wrote: My solution. With Zaptel 1.4 I change ztdummy.c comment lines 47 to 56, for rtc config. Compile and without rtc module, load ztdummy. It work good with usbcore and uhci_hcd modules. I have installed libusb-dev for my debian etch system. Now, my kernel is 2.6.20.2, but it work good with 2.6.18 can ztdummy / 2.6 take timing from a UHCI USB controller? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting 7970 to update
I'm having issues with a Cisco 7970. It seems to ignore minor changes in its config file. Is there something like the versionstamp or some other setting I need to increment in order to get the 7970 to update each time? It does seem to download the file from the TFTP server, but it never updates the display or its settings. Thanks 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] How to match wild card inside a GoToIf?
Try
exten = s,1,GotoIf($[${ARG1:0:5}=220408]?2:3)
This looks at the first 5 digits of ARG1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Tuesday, March 13, 2007 12:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to match wild card inside a GoToIf?
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3) exten = s,2,Hangup
Any ideas?
Regards,
Ricardo.
___
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RE: [asterisk-users] How to match wild card inside a GoToIf?
I don't believe this will work. He wants it to goto if EXTEN =
220408235 or 220408743 or any other digits for the last 3 of the
extension block 220408xxx. When Asterisk processes both his and your
line it's going to look to see if the EXTEN is exactly 220408XXX, which
of course it will never be.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Connolly,
Tim
Sent: Tuesday, March 13, 2007 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] How to match wild card inside a GoToIf?
Try
exten = s,1,GotoIf($[${MACRO_EXTEN} = 220408XXX]?2:3) exten =
s,2,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Tuesday, March 13, 2007 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to match wild card inside a GoToIf?
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3) exten = s,2,Hangup
Any ideas?
Regards,
Ricardo.
___
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RE: [asterisk-users] DST and VM timestamp
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, March 13, 2007 12:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DST and VM timestamp On Tue, Mar 13, 2007 at 10:44:08AM -0600, Damon Estep wrote: Who is tired of dealing with DST changes? I have asterisk running on FC4, FC4 has been patched and shows the correct MDT timezone and time. Email notifications of voicemail show the message time an hour early (standard time, not daylight). This si the time in the message body, not the email delivery time, so it is coming form asterisk wrong. Email in the headers has a timezone information in it. If you fixed the system time by etting the clock, rather than fixing the timezone definitions, you may get errors. OTOH, the recipient of the email may have done that error. I did a reload after correcting the time/timezone, but not a restart. No reload or restart should be needed. The tzdata was update on the FC4 box, the OS shows the correct time and time zone (MDT), the app is still running with the old tzdata. Putting a tz= in the voicemail config for each use as well as pointing that same entry to the /America/Denver zone info file corrects the issue, but I am assuming a reboot will also. The system is too busy to reboot in the middle of the day, so will confirm after an early morning restart. It seems (as stated by a previous responder) that asterisk (and many other apps) reads time zone info at startup, so a restart is required. ___ --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] 120 concurrent ZAP connections in asterisk open edition. Is that possible?
Using an octal(8 T1 ports) card I have kept an average of 150 concurrent Zap channels open on a single server over 8 T1s. It's all a matter of what the hardware will support. Pure Zap channel conversations isn't always the limiter, what else are you doing on this server? MATT--- On 3/13/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Mar 13, 2007 at 12:18:50PM -0500, Héctor Maldonado wrote: Hi all, In your experience, what is the maximum number of *concurrent* zap channels that you've ever tried with one box of Asterisk open edition? With Zaptel, the limit is pretty clear: the number of channels your hardware supports... 120 is the capacity of a quad E1 card, or (if I may pitch our own hardware) 4 Astribank 32 units. A decent system that does not have much transcoding conferencing or other types of complicated processings should have no problem with such a load. In my case, the max that I've tried was 63 simultaneous connections in a Quad T1/E1 card installed on a Intel Pentium D 3.4Ghz 2GB ram system. That's half the capacity of the quad E1. What exactly did you try? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 48
From: Ricardo Carvalho [EMAIL PROTECTED]
Subject: [asterisk-users] How to match wild card inside a GoToIf?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3)
exten = s,2,Hangup
You are going to need a substring of the original. I'm thinking
something like the following although I haven't tested it.
exten = s,1,GotoIf($[${ARG1:3} = 220408]?2:3)
dbc.
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Re: [asterisk-users] Number of SIP messages per minute
Just how many SIP packets do you think it takes to set up a call? Remember AUDIO IS NOT SIP! SIP is for call control, setup, and teardown. Do a sip debug in the CLI and see just how many packets it takes to setup a call. Matt wrote: That does sound low, especially if you have multiple devices behind a NAT. I have customers with 8 analog lines going into their analog phone system and just have 4 ATAs with 2 lines each. Of course, all of this traffic would seem to come from the same IP! On 3/8/07, Mark Davies [EMAIL PROTECTED] wrote: Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me. ___ --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 1.2.15 fax
Is there any way to implement t38 in asterisk 1.2.15 Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Getting 7970 to update
I went back to a simplified config. Although it sits at registering now forever.. Can't dialout once it does give up. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Connolly, Tim Sent: Tuesday, March 13, 2007 2:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Getting 7970 to update I'm having issues with a Cisco 7970. It seems to ignore minor changes in its config file. Is there something like the versionstamp or some other setting I need to increment in order to get the 7970 to update each time? It does seem to download the file from the TFTP server, but it never updates the display or its settings. Thanks 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 ___ --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 match wild card inside a GoToIf?
[default]
exten = _220408XXX,1,Hangup
exten = _2,1,Macro(outcall,${EXTEN})
Ken Williams wrote:
I don't believe this will work. He wants it to goto if EXTEN =
220408235 or 220408743 or any other digits for the last 3 of the
extension block 220408xxx. When Asterisk processes both his and your
line it's going to look to see if the EXTEN is exactly 220408XXX, which
of course it will never be.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Connolly,
Tim
Sent: Tuesday, March 13, 2007 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] How to match wild card inside a GoToIf?
Try
exten = s,1,GotoIf($[${MACRO_EXTEN} = 220408XXX]?2:3) exten =
s,2,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Tuesday, March 13, 2007 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to match wild card inside a GoToIf?
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3) exten = s,2,Hangup
Any ideas?
Regards,
Ricardo.
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[asterisk-users] RE: In Asterisk 1.4.x, Why Digium has two H323 channels?
Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more, and studing the chan_h323, it's the old chan_oh323 I not wanna work with add_ons but in Asterisk 1.4.x, will I have work? Somebody confirm it, have the same opinion. Or this new chan_h323 work fine without the problems that had the H323 or OH323 channels. Thanks in Advanced. Thiago Maluf Resende. -- Date: Mon, 12 Mar 2007 15:57:42 +0100 From: Pavel Jezek [EMAIL PROTECTED] Subject: Re: [asterisk-users] In Asterisk 1.4.x, Why Digium has two H323Channels To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed as I know, ooh323 is external project from objective systems, anyway, for 1.4 I prefer chan_h323 from asterisk tree. Thiago Maluf wrote: Now, the H323 Channels is updated and your bugs fixed. But Digium still develop your OOH323 Channel. My question is why? What is the better in Asterisk 1.4.x.? I know that in Asterisk 1.2.x OOH323 is very better than H323 or OH323. Thanks in advanced. Thiago. -- -- -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 120 concurrent ZAP connections in asterisk open edition. Is that possible?
2007/3/13, Matt Florell [EMAIL PROTECTED]: Using an octal(8 T1 ports) card I have kept an average of 150 concurrent Zap channels open on a single server over 8 T1s. It's all a matter of what the hardware will support. Pure Zap channel conversations isn't always the limiter, what else are you doing on this server? Actually a couple of daemons to control asterisk and nothing else. 63 concurrent is the maximum traffic that I've experienced, and my concern is in how much additional traffic could I handle without having problems. BTW, what is the configuration of your server with the octal card? memory? cpu? 150 concurrent zap channels is a lot.. at least for me :) 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] 120 concurrent ZAP connections in asterisk open edition. Is that possible?
On 3/13/07, Héctor Maldonado [EMAIL PROTECTED] wrote: 2007/3/13, Matt Florell [EMAIL PROTECTED]: Using an octal(8 T1 ports) card I have kept an average of 150 concurrent Zap channels open on a single server over 8 T1s. It's all a matter of what the hardware will support. Pure Zap channel conversations isn't always the limiter, what else are you doing on this server? Actually a couple of daemons to control asterisk and nothing else. 63 concurrent is the maximum traffic that I've experienced, and my concern is in how much additional traffic could I handle without having problems. BTW, what is the configuration of your server with the octal card? memory? cpu? 150 concurrent zap channels is a lot.. at least for me :) I was doing VICIDIAL capacity testing on this server with a quad processor(dual core) server, so the load was high and the call volume was also very high, not a good comparison to what you are doing. I have handled more than 100 concurrent channels before with two quad-T1 cards in a singl P4 3.2GHz server before with no issues, So I don't really know why you would be having issues with more than 63 channels on a similar machine. What does top show when this happens? MATT--- ___ --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] voicemail scenario
Hi, i finally managed to get it work using GlobalVar. I still have a question. I have several context in my voicemail.conf like [default] [customer_1] [customer_2] [customer_3] How can i set a different emailsubject for each context? thx --- richard Coco [EMAIL PROTECTED] wrote: Hi all, i need help to implement a voicemail scenario. What i am trying to do is the following. user X dials a direct access for user Y voicemail and is asked to enter a number (e.g 12345678) and then leaves a message. Then asterisk sends a notification with attachement. The problem is that i need the number entered (e.g 12345678) in the subject. Is that possible. thx in advance. Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --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 The fish are biting. Get more visitors on your site using Yahoo! Search Marketing. http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php ___ --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] 120 concurrent ZAP connections in asterisk open edition. Is that possible?
I was doing VICIDIAL capacity testing on this server with a quad processor(dual core) server, so the load was high and the call volume was also very high, not a good comparison to what you are doing. I have handled more than 100 concurrent channels before with two quad-T1 cards in a singl P4 3.2GHz server before with no issues, So I don't really know why you would be having issues with more than 63 channels on a similar machine. What does top show when this happens? Thank you Matt for your response.. Actually, my concern wasn't so big as it was for the asterisk ability to process such a load.. If only one box of asterisk (open edition) could process more than 120 concurrent calls.. thats really cool!.. ___ --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] Asterisknow with video and X-Lite not quite working
Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can see me. When I make a call and then click 'Start' (sending video) in the X-Lite client, nothing seems to happen on the other side, but here it says that a video transmission has begun. According to 'sip show codecs', both the h.263 and h.263p codec are supported, and those are also set on either X-Lite clients. I have enabled 'canreinvite' for both users as well, but still the other user can not see me. I can, however, see the cameras view on my computer, so that seems all properly set up. Could anyone help me sort this out? 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] RE: In Asterisk 1.4.x, Why Digium has two H323 channels?
for 1.4 you have only two choices chan_h323 and chan_ooh323, chan_oh323 from inaccessible networks, is death project, more than year unmaintained, I'm using chan_h323 both from 1.2 and 1.4 without problems (opposite site to chan_h323 is ci$co gateway or callmanager) also, chan_ooh323 isn't maintained as good as chan_h323, if you will have some issues with chan_h323, you can report through digium bugtrack and reply come quite quickly, not so with chan_ooh323 thats my reasons, why I can recommend original chan_h323. PJ Thiago Maluf wrote: Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more, and studing the chan_h323, it's the old chan_oh323 I not wanna work with add_ons but in Asterisk 1.4.x, will I have work? Somebody confirm it, have the same opinion. Or this new chan_h323 work fine without the problems that had the H323 or OH323 channels. Thanks in Advanced. Thiago Maluf Resende. -- Date: Mon, 12 Mar 2007 15:57:42 +0100 From: Pavel Jezek [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Subject: Re: [asterisk-users] In Asterisk 1.4.x, Why Digium has two H323Channels To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed as I know, ooh323 is external project from objective systems, anyway, for 1.4 I prefer chan_h323 from asterisk tree. Thiago Maluf wrote: Now, the H323 Channels is updated and your bugs fixed. But Digium still develop your OOH323 Channel. My question is why? What is the better in Asterisk 1.4.x.? I know that in Asterisk 1.2.x OOH323 is very better than H323 or OH323. Thanks in advanced. Thiago. -- -- -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] mailto:[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 ___ --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] Number of SIP messages per minute
Just how many SIP packets do you think it takes to set up a call? Probably around 8 - 10 per call, excluding any ReINVITES, DTMF, etc. INVITE, Authentication Required, ACK INVITE w/AUTH INFO, TRYING, RINGING, OK BYE, OK --Luki ___ --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] great problem with sounds and ztdummy
Tzafrir Cohen escribió: On Tue, Mar 13, 2007 at 06:25:37PM +0100, Germ�n Aracil Boned wrote: My solution. With Zaptel 1.4 I change ztdummy.c comment lines 47 to 56, for rtc config. Compile and without rtc module, load ztdummy. It work good with usbcore and uhci_hcd modules. I have installed libusb-dev for my debian etch system. Now, my kernel is 2.6.20.2, but it work good with 2.6.18 can ztdummy / 2.6 take timing from a UHCI USB controller? Yes. Download from svn version 1.4 of zaptel, edit ztdummy.c and comment lines 47 to 56: /* #if defined(__i386__) || defined(__x86_64__) #if LINUX_VERSION_CODE = VERSION_CODE(2,6,13) #define USE_RTC #else #if 0 #define USE_RTC #endif #endif #endif */ compile all, install all, unload rtc module and load ztdummy module. This work good You need libusb headers see line 69 from ztdummy.c: #include linux/usb.h ___ --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] Press quotes needed from ENUM users on Asterisk
I'm helping with an article in New Scientist on the use of ISN (http://www.freenum.org/) and the reporter with whom I'm working is trying to get some quotes from users of normal ENUM services (e164.arpa, please) from a telco perspective as a comparative basis. If you run an SS7 interconnected telephone switching system that incorporates VoIP via ENUM (hopefully via Asterisk, but any platform will do), please let me know via email and I'll pass your information along to the reporter. Deadline is tomorrow morning, so fast reactions appreciated. This is somewhat off topic, but possibly still the best place to ask, so my apologies for the s/n disruption. Both of these methods are built into Asterisk in the ENUMLOOKUP function, and I suspect that Asterisk is one of the most common places to find users of these technologies, along with SER, OpenSER, and SIPxchange. JT ___ --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] Number of SIP messages per minute
Ok so let's go with 10. Now say you have a busy call-center behind a NAT with 8 lines. That's 80 SIP messages. And if you have short calls, you could easily exceed that, especially if you are placing calls on hold, forwarding, etc. On 3/13/07, Luki [EMAIL PROTECTED] wrote: Just how many SIP packets do you think it takes to set up a call? Probably around 8 - 10 per call, excluding any ReINVITES, DTMF, etc. INVITE, Authentication Required, ACK INVITE w/AUTH INFO, TRYING, RINGING, OK BYE, OK --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Back
--- Klaverstyn, David C [EMAIL PROTECTED] wrote: Can you provide some specific details as I would like to implement something like this. I wrote this application a while ago for FreePBX, maybe it helps: http://samyantoun.50webs.com/asterisk/callback/callback.gif http://samyantoun.50webs.com/asterisk/athome/callback.htm We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Back
Check out Nerd Vittles at nerdvittles.com. There's an article on this kind of scenario On 3/14/07, Samy Antoun [EMAIL PROTECTED] wrote: --- Klaverstyn, David C [EMAIL PROTECTED] wrote: Can you provide some specific details as I would like to implement something like this. I wrote this application a while ago for FreePBX, maybe it helps: http://samyantoun.50webs.com/asterisk/callback/callback.gif http://samyantoun.50webs.com/asterisk/athome/callback.htm We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265 ___ --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] voicemail scenario
I dont think you can but you can use a variable. Have a look at
voicemail.conf. You can edit the message the asterisk sends out. If you want
the CID to be in the subject you can use the variable ${CALLERID(number)} .
- Original Message -
From: richard Coco [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 13, 2007 10:53 PM
Subject: Re: [asterisk-users] voicemail scenario
Hi,
i finally managed to get it work using GlobalVar.
I still have a question. I have several context in my
voicemail.conf like
[default]
[customer_1]
[customer_2]
[customer_3]
How can i set a different emailsubject for each
context?
thx
--- richard Coco [EMAIL PROTECTED] wrote:
Hi all,
i need help to implement a voicemail scenario. What
i
am trying to do is the following.
user X dials a direct access for user Y voicemail
and
is asked to enter a number (e.g 12345678) and then
leaves a message. Then asterisk sends a notification
with attachement. The problem is that i need the
number entered (e.g 12345678) in the subject. Is
that
possible.
thx in advance.
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html
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[asterisk-users] Re: Asterisknow with video and X-Lite not quite working
benedikt, * 1.4 does no video codec translation... it is just a pass through so using the same unit on both ends is a plus. you might try adding this codec too. allow=h264 I assume that audio is ok, just no video, right!? there may be a nat problem, try nat=no I have some video experience but not with x-lite... which version of x-lite are you using? I would recommend getting the full release of 1.4.1... and reinstalling over * Now. daveC -- Message: 16 Date: Tue, 13 Mar 2007 22:10:35 +0100 From: Benedikt Franz [EMAIL PROTECTED] Subject: [asterisk-users] Asterisknow with video and X-Lite not quite working To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can see me. When I make a call and then click 'Start' (sending video) in the X-Lite client, nothing seems to happen on the other side, but here it says that a video transmission has begun. According to 'sip show codecs', both the h.263 and h.263p codec are supported, and those are also set on either X-Lite clients. I have enabled 'canreinvite' for both users as well, but still the other user can not see me. I can, however, see the cameras view on my computer, so that seems all properly set up. Could anyone help me sort this out? Thanks. -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium S101i - Adapter DTMF works perfeclty
Does anybody know what DTMF coding does S101i adapter using? I've been testing one for over a week and here are my observations: - DTMF signaling is working perfectly with Asterisk, much better than Sipura 3K Though, I think the Asterisk iaxy firmware is buggy, the unit is using auto-update feature; so I have Asterisk 1.2.13 and iaxy firmware version is: 23 When enable in provisioning for both options: server and altserver initial connection goes through fine but there is some problems with provisioning routine after. I lose the connection frequently (during conversation, after, or when phone is on hook) and the unit S101i will not initialize itself again, usually the orange light stays ON (even though there is no conversation) or there is no light at all. When I enable in provisioning only server option the unit works perfectly even over the Internet (it has be up for over a week): S101i -- Firewall -- Inernet -- Firewall -- Asterisk Server 2.) Does anybody know how to tune up DTMF in Sipura 3K ? sip.conf is set for: dtmfmode=rfc2833 -- #Joseph ___ --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] Strange issue SIP URI to follow me busy signal
Ok, I'm going to have to lay out how we have this set up so you can understand. :) We're using a VOIP provider (ViaTalk) and have our main trunk provisitioned with them. We also have 2 of what they call virtual numbers ... we have one set up to do a SIP forward to [EMAIL PROTECTED] and [EMAIL PROTECTED] We're using AAH 2.7. Extensions 801 and 802 both have follow me set up on them so they'll ring our cell phones if we're not in the office. An example of the problem: If someone calls our main number and dials extension 801, it will ring my desk phone and then roll over to my cell phone if I don't answer. No problems here, it works fine. I'm at extension 801 ... if someone calls my virtual number it rings my SIP phone on my desk just fine (an Aastra 480i CT)... however, when it tries to ring my cell phone after I miss the call on my desk phone, the caller receives a busy signal. My cell phone still rings, but the caller has already hung up by that point because they got the busy signal. Has anyone seen anything like this? Any thoughts on what might be causing this? Regards, George A. Roberts IV President and CEO, Interjuncture Corp. | http://www.interjuncture.com/ [EMAIL PROTECTED] | +1 630 364-4100 x801 HostingCon 2007 - The largest gathering of hosted services professionals in the world! Historic Navy Pier, Chicago | July 23-25, 2007 | http://www.hostingcon.com/2007/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Back
Vernier Umali wrote: Check out Nerd Vittles at nerdvittles.com. There's an article on this kind of scenario proprietary :P -Stephen- ___ --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 Integrator Birch
I'm thinking about replacing my Birch T1 integrator with an Asterisk box. The Integrator has 12 voice 768k data, so the Asterisk box would become a router PBX. Has anyone done anything similar, what experiences have you seen /or read about. TIA ___ --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
