Re: [asterisk-users] multi tenant
Yes, you can do this. You should point the trunks to the right context and done. Op 30-10-2012 8:15, Darin Iv schreef: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debugging Sip
Elliot, I am installing by default wireshark/thark on my asterisk machines. This allows to do protocol traces from the linux commandline and to store the traces into a file. You can find more information at www.wireshark.org Henk Elliot Murdock schreef: Hello, When debugging SIP in Asterisk is it possible to send the SIP debug log to a specific file instead of the general log file, or even better, send each call into its own file for easier analysis? Thanks, Elliot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP Trigger on incoming call request
Daniel, Have you thought about using CURL from the Dialplan? Henk Daniel Isenmann schreef: Hi, is it possible to configure a TCP trigger to a predefined address and port on a incoming call request? Some background: For example “Client 1” tries to call “Client 2”, “Client 1” is sending the call request to Asterisk (SIP-Server). Asterisk open a connection to the predefined address and port and send a simple TCP message (trigger) with caller and callee ID and close the connection afterwards. Is this scenario possible without any plugin like the Java API or similar? Thanks, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: Re: Large issue - having trouble diagnosing.]
Try to find a pattern. Looks that you are able to reproduce the problem. You mention after 4 minutes. Is this also the case for internal calls? If so then I would say that the E1 is ok. If not then I would step more into E1 related issues. Have you looked at the ethernet cards. Collisions, errors. Cameron Hissey schreef: thankyou both very much for your swift responses and helpful insight... While my knowledge of administrating Asterisk is fairly decent, i must say my knowledgebase and ability in troubleshooting it is fairly lousy... all these wonderful suggestions you have had about turning this log on here, etc sounds great, but i don't know where to begin on that! how do you recommend i turn these on or obtain them, else is there a site you can point me to to save your precious time? as for the network, we have two cisco routers, one is PoE and the other is standard. we have tried to keep things constant whereby phones are connected to the PoE and the data devices are connected to the standard switch, however the cabling was a bit of a rush job and consequently the PoE has proven unstable on many of the points, with some of them not even supplying data packets. this has meant i have had to share a single port for some desks, where the ethernet cable plugs into the phone and the computer's ethernet connection is routed through the phone also. The issues that we are having however are not confined to any single desk; they occur sperratically on all phones and with any number of call volumes (small business so only max 6 calls at once incl internal-internal) from 1 through to maximum capacity. I've been told that it usually happens around the 4 minute mark but i wouldn't hold him to that... im happy to setup whatever you think is going to fix this, however unfortunately with having to share network points, i dont really know how VLANS and segmentation are going to go... Thanks so much everyone for your support! Sincerely, Cameron Hissey On Jan 21, 2008 4:13 PM, Paul Hales [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Forwarded Message *From*: Paul Hales [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Reply-To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:[EMAIL PROTECTED] *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:[EMAIL PROTECTED] *Subject*: Re: [asterisk-users] Large issue - having trouble diagnosing. *Date*: Mon, 21 Jan 2008 15:47:54 +1100 Generally, E1 is pretty rock solid so my guess is more inside the network. We found an issue at a site a while ago which was pretty bad (calls cutting off randomly) and we fixed it by disconnecting the voice and data networks. We could have troubleshot it properly, but fitting an extra network card in the server was cheaper and faster. Is there anything ugly in the logs? If not, you could look at turning o debugging in logger.conf . later, PaulH On Mon, 2008-01-21 at 15:04 +1100, Cameron Hissey wrote: Hello, I am having a lot of trouble with my deployment of Asterisk. I am running the PBX-In-a-flash turnkey of Asterisk and ever since deployment I have had many different problems. I have managed to get all issues sorted out as I go along, until this one that randomly began last week. We are using Grandstream GXP 2000 Handsets in the office, and at TE110P card to interface to our ISDN OnRamp10 connection (10 Channels of PRI). The problem arising seems to happen roughly 4minutes into a call. Basically all of a sudden the caller just starts to no longer be understood (sounds like morse code, only milliseconds of voice packets getting through in either direction). naturally this could be a number of non-asterisk related things such as a carrier fault, bad network wiring (even more possible as we are using PoE), even badly configured QoS. However things being as they are my boss has taken it upon himself to absolve himself of any possible blame for any system that he manages (everything but the asterisk box) and lumped it all on me in such a way that its basically my job if i cannot get this working. With all of this, i need to do everything i can to rule out the Asterisk box, so i can go back to him with confidence and clear asterisk of any wrongdoing. Has anyone here ever heard of this sort of problem, and if so did you find a solution? If not, what steps would you recommend i take to diagnose the issue and rectify it as quickly as possible? Thankyou very much, Cameron Hissey
Re: [asterisk-users] Definity G3R and MWI
I have been playing with this some time ago. We used the so called mode code integration. This worked fine. It works simular as described for other Avaya Product. http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration Henk BJ Weschke schreef: I've just spent the last two hours Googling and searching the Wiki. I' trying to find if there are any listings of codes for the Avaya Definity G3R, to allow for an Asterisk system to turn on/off a phones MWI that is attached to a G3. We are looking to use an Asterisk system as a voice mail server. I think you're going to need to integrate via the SMDI feature of Asterisk and figure out what the Definity needs as well to work with an SMDI connection. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Definity G3R and MWI
Doug, Have you checked the feature access code that is defined in the definity. That is the code that needs to be dialed. I always checked the codes from a definity phone to make sure that I was using the right codes. Henk Doug Lytle schreef: Henk Dick wrot I have been playing with this some time ago. We used the so called mode code integration. This worked fine. It works simular as described for other Avaya Product. http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration Yes, I saw the page. The Definity wouldn't accept *53 for on, and #*53 for off. For a test, I was using extension 5574, so I did a Dial(ZAP/g1/*535574) from a Asterisk console with no results on the test phone. This system is attached via a PRI. Thanks! Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??
Did you install mpg123? You can check from linux prompt by just typing mpg123 Bhrugu Mehta schreef: no , not at all, there is no need to install sound card in asteirsk system. I am using asterisk server without soundcard. so there may be antoher problem may in configurtion of zapata or other. cheers!!! Bhrugu mehta On Dec 3, 2007 11:31 PM, Stefan Guenther [EMAIL PROTECTED] wrote: Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten = 202,1,ANSWER() exten = 202,2,PLAYBACK(tt-monkeys) exten = 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/user1-0827ebe8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Playback(SIP/user1-0827ebe8, tt-monkeys) in new stack -- SIP/user1-0827ebe8 Playing 'tt-monkeys' (language 'de') Yes, the file tt-monkeys exist in /var/lib/asterisk/sounds and the subdirectory de. No, there is no error message even if turn on debugging. :-( Besides this strange behaviour, I was wondering whether the asterisk server needs an soundcard to send the output of e.g. the playback application to the phone. BTW, this is asterisk 1.4.13 I would be really happy, if someone has an idea how to solve this problem. Thanks in advance, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Check if SIP user is available or not ?
You can use sip show peers. If an IP address is shown then the user will be available. Henk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400P and Junghanns QuadBRI issue
I would check: Cat /proc/zaptel/ To make shure that the cards are activated in the order that you programmed them. Henk _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Backup e-mail Sent: maandag 30 april 2007 13:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM400P and Junghanns QuadBRI issue Hi List, I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns QuadBRI on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, libpri-1.2.4 and asterisk-1.2.17 When it's the time for ztcfg to do its job it complains with ZT_SPANCONFIG failed on span 2: No such device or address (6) I'm out of ideas what to do to make it to work. Your help is very much appreciated. The config files and results from various commands follow. Thanks, Costa. --- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXO Loopstart (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: D-channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: D-channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: D-channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) 16 channels configured. ZT_SPANCONFIG failed on span 2: No such device or address (6) --- The /etc/modprobe.d/blacklist file contains, amongst others, the following lines: -- blacklist hisax blacklist hisax_fcpcipnp blacklist 8139cp blacklist hfc4s8s_l1 -- The /etc/zaptel.conf file looks like this: - zaptel.conf - # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxsks=1 fxols=2 fxsks=3 fxols=4 # Span 2-5: Junghans span=2,1,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami bchan=5,6 dchan=7 bchan=8,9 dchan=10 bchan=11,12 dchan=13 bchan=14,15 dchan=16 # Global data loadzone= fr defaultzone = fr --- The lsmod | grep zap command gives the following: --- zaptel 182820 8 wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm crc_ccitt 6337 2 zaptel,irda --- The lspci -vv command returns the following info in relation to the Junghanns card: 00:0a.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST] Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at d400 [size=8] Region 1: Memory at e2001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- --- _ Ahhh...imagining that irresistible new car smell? Check out new http://us.rd.yahoo.com/evt=48245/*http:/autos.yahoo.com/new_cars.html;_ylc= X3oDMTE1YW1jcXJ2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDbmV3LWNhcnM- cars at Yahoo! Autos. ___ --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] CPU motherboard for 100+ simultaneouse calls onDigium Quad E1 TE411p
Which codec do you plan to use? Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of umar tarar Sent: woensdag 7 februari 2007 20:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] CPU motherboard for 100+ simultaneouse calls onDigium Quad E1 TE411p hi! anyone please recommend/guide me of purchasing a resonably high performance server system regarding processor(s) motherboard (+ other compulsary peripherals i.e. VGA, Soundcard). Mentioning up-to-date vendor+model will be more helping I've to use Digium TE411p Quad E1 card signalling on the E1 is SS7 no. of simultaneouse calls is 100+ 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
RE: [asterisk-users] TE110P and TDM400P
I think that you are loading the drivers in the wrong order. You can change the order of loading are first define the E1 followed by the TDM400 Hope this helps, Henk _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln Zuljewic Silva Sent: woensdag 22 november 2006 20:51 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TE110P and TDM400P Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four X100P - FXS). Both boards are recognized by the operating system as showed above: :08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 64, IRQ 169 I/O ports at e800 [size=256] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 :08:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 193 I/O ports at e400 [size=256] Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 The problem is that I cant make the both cards to work together in the same server. Here is my /etc/zaptel.conf: ### fxsks=1-4 loadzone = us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 ### When I load the wctdm module, I get this error: ZT_CHANCONFIG failed on channel 5: No such device or address (6). Its sounds like the FXS module its tring to configure the channels 5 to 35 (E1 - ISDN Channels - TE110P card). Anybody already saw this ? Its possible to use this two cards in the same computer ? There is any separator that I can use in zaptel.conf to make the load of the modules dont mistakes itself ? Here is my versions: Debian kernel - 2.6.8 asterisk-1.2.12.1 libpri-1.2.4 zaptel-1.2.11 Thanks Lincoln ___ --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] TE110P and TDM400P
I would suggest the following - remove the drivers - load them manually (zaptel, wcte11xp, wctdm) Run: Zttools - should show unconfigured cards. Take: /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxsks=32-35 loadzone = us defaultzone=us run: ztcfg -vv See what it is saying Hope this helps ___ --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] Operator in Voicemail
This is what I am using: exten = o,1,Answer() exten = o,2,GoTo(default,3000,1) exten = o,3,Hangup() Hope this helps, Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Davis Sent: maandag 24 juli 2006 18:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Operator in Voicemail Im having the exact same problem here. I originally thought it was a context problem. However, to troubleshoot I tried placing the following in every context (default, from-inside, from-outside, etc) in extensions.conf with no luck: exten = o,1,DIAL(SIP/100,100) Like Kevin, it works fine for our internal users, just doesnt work for callers coming from the PSTN. Thanks, -AntD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy Sent: Monday, July 24, 2006 7:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Operator in Voicemail Ive got an odd problem. I have set in Voicemail.conf operator=yes as a default. This is so that when a caller is in the voicemail system they can press 0 and be sent to the operator. This works fine when the caller is internal to the system but NOT when the caller is calling in from the PSTN. Instead the caller gets the message Press 1 to accept the recording. Pressing 0 again deletes the message. How do I get this to work for outside callers calling in?? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Operator in Voicemail
It is an Oh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: maandag 24 juli 2006 21:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Operator in Voicemail Are you sure this is saying exten = 0 with a ZERO and not an Oh? Looks like a lowercase Oh to me below. Kevin Savoy wrote: This doesn't solve the problem. Still the same. Any other ideas? This is what I am using: exten = o,1,Answer() exten = o,2,GoTo(default,3000,1) exten = o,3,Hangup() Hope this helps, Henk *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony Davis *Sent:* maandag 24 juli 2006 18:20 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [asterisk-users] Operator in Voicemail I'm having the exact same problem here. I originally thought it was a context problem. However, to troubleshoot I tried placing the following in every context (default, from-inside, from-outside, etc) in extensions.conf with no luck: / exten = o,1,DIAL(SIP/100,100)/ Like Kevin, it works fine for our internal users, just doesn't work for callers coming from the PSTN. Thanks, -AntD *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin Savoy *Sent:* Monday, July 24, 2006 7:37 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Operator in Voicemail I've got an odd problem. I have set in Voicemail.conf operator=yes as a default. This is so that when a caller is in the voicemail system they can press 0 and be sent to the operator. This works fine when the caller is internal to the system but NOT when the caller is calling in from the PSTN. Instead the caller gets the message Press 1 to accept the recording. Pressing 0 again deletes the message. How do I get this to work for outside callers calling in?? ___ --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] Very bad quality with AVM Fritz!cardPCIandchan_capi
If you do not use USB then I would suggest to disable this in the bios. Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Harper Sent: donderdag 29 juni 2006 14:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Very bad quality with AVM Fritz!cardPCIandchan_capi That is possible, how could I check that? I can see that IRQ 17 is shared between eth0 and my fritz!card but I don't know if it changes anything: Can you try it (or eth0) in a different slot (or change IRQ's in the BIOS if possible) to see if it makes any difference? That's the only way to know for sure, anything else is just speculation :) James Unfortunately I can't, because the LAN board is integrated and I have no PCI device left anymore. You seem to have 2 wctdm adapters. Can you swap one of them with the fritz card? James ___ --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] Avaya 4610sw SIP setup problem
Did you try to manually to change the parameters of the phone? When you power the phone up then are you able to enter manually the parameter when you hit *. I am using a 4610 with Release 2.2 but I am not using the capability to upload the settings from the server. Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herchi Silviu Sent: donderdag 29 juni 2006 15:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Avaya 4610sw SIP setup problem I just tried serving the files off Apache, port 80, no change... Most parameters are taken into account by the phone, except for SIP proxy and SIP registrar... Coud someone post an excerpt from their 46xxsettings.txt where I could see the format they use? Thank you in advance, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lynn Sent: 29 June 2006 00:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since. On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom Lynn Sent: 28 June 2006 05:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying Registering for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR 204.140.111.200 SET DSTOFFSET 1 SET DSTSTART 1SunApr2L SET DSTSTOP LSunOct2L SET GMTOFFSET -5:00 SET DATESEPARATOR / SET DATETIMEFORMAT 3 SET DIALPLAN [234]xxx|55 SET DIALWAIT 3 SET MUSICSRVR SET MWISRVR SET PHNNUMOFSA 3 SET REGISTERWAIT 120 SET SIPDOMAIN sip.mycompany.com SET SIPPROXYSRVR 204.140.111.219 SET SIPPORT 5070 (this is not a typo) SET SIPREGISTRAR 204.140.111.219 SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o= avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support. avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___ --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] Avaya 4624 Ip phone
I dont think that this will work. It is Avayas own implementation of H323 and not standard H323. When you want to use the phone you have to migrate them to SIP. Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas Sent: woensdag 7 juni 2006 3:13 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Avaya 4624 Ip phone does anybody have any file of configuration of phones avaya? in H323 protocol? Regards On 6/6/06, Mark Phillips [EMAIL PROTECTED] wrote: Ig nore my last post. I had not seen this posting On Tue, 2006-06-06 at 22:33 +0200, Henk wrote: Have a look at the attached link. http://support.avaya.com/japple/css/japple?temp.documentID=283920temp.produ ctID=107755temp.bucketID=108025PAGE=Document Henk __ From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Gabriel Rosca Sent: dinsdag 6 juni 2006 21:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Avaya 4624 Ip phone Hi guys, I installed asterisk and it's working really well. For now I`m using soft phones IAX and SIP but now I want to use the regular IP phone, what I have now is Avaya 4624 and I didn't find a firmware for SIP for this particular phone I believe now is working with H.323, can please someone advice me if exist firmware for this phone to register SIP or IAX2 with my asterisk box, and to show me an example of config file for this phone. Thank you, Gabriel ___ --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] Avaya 4624 Ip phone
Have a look at the attached link. http://support.avaya.com/japple/css/japple?temp.documentID=283920temp.productID=107755temp.bucketID=108025PAGE=Document Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Rosca Sent: dinsdag 6 juni 2006 21:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Avaya 4624 Ip phone Hi guys, I installed asterisk and its working really well. For now I`m using soft phones IAX and SIP but now I want to use the regular IP phone, what I have now is Avaya 4624 and I didnt find a firmware for SIP for this particular phone I believe now is working with H.323, can please someone advice me if exist firmware for this phone to register SIP or IAX2 with my asterisk box, and to show me an example of config file for this phone. Thank you, Gabriel ___ --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] configuration
Create the 2 extensions in /etc/asterisk/extension.conf exten = 8,1,Answer() . Script 1 . exten = 9,1,Answer() . Script 2 . Make sure that the channel where the calls come in route the call to the context where you defined the scripts. Hope this helps, Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: donderdag 1 juni 2006 9:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] configuration hello I have 2 services with 2different numbers. the first is 88 and the second is 99. if a user call 88 I want to execute the script1 and if he call 99 I execute the script2. How can I do my configs files? big Thanks issam ___ --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] Need help with Junghanns Quadbri
Try to do ztcfg s before you run ztcfg -vv Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis curty Sent: woensdag 31 mei 2006 12:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need help with Junghanns Quadbri Hi everybody I hope that somebody can help me with the following I have 2 quadbri cards 2 - 1t0 cards 1 pabx alcatel 4200 I would like to connect my asterisk to the alcatel , I installed bristuff 0.3.0-1p , loaded the zaphfc driver in NT mode configured zaptel and zapata , it works great. then I removed the 1 t0 card, added the quadbri loaded qozap : insmod qozap.ko ports=15 ( 4 ports in NT ) adjusted the zaptel zapata, specified the right signalling, right context ran ztcfg -vv ( 12 channels configured ) started asterisk, I get layer1 down message on the 4 ports, leds remain red what ever I do in my conf , I am not able to get a reaction from the card ( I tried with my two quadbri, on 2 different pc's ) what can I check ? thanks jl ___ --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 strip a digit
This should do the job
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
Henk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: dinsdag 30 mei 2006 22:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to strip a digit
I have the following extension to dial outside via SIP
it's like this:
phoneasterisk-internet-SIP providerUSA
exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN},55,o)
exten = _91NXXNXX,3,Hangup
I want to strip the digit 9 before sending it to the SIP provider.
Also, any suggestions for the above definition?
Thanks,
Erick.
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RE: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI
I think that you are sending an outgoing caller id that is not part of the DID range. Most operators do not allow this. ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] Are you using caller id 1013 ? Change it to a number that is part of your trunks. Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sébastien Mortier Sent: maandag 20 maart 2006 11:52 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM -- OctoBRI -- Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h -- OctoBRI -- PABX e-Generis ISDN Phones | | SIP Phones France Telecom -- SIP Phones : Works France Telecom -- ISDN Phones : Works SIP Phones -- ISDN Phones : Works ISDN Phones - SIP Phones : Works SIP Phones -- France Telecom : DOESN'T WORK ISDN Phones - France Telecom : DOESN'T WORK Here are some characteristics of my Asterisk Setup OS Linux Gentoo 2.6.15-r1 zaptel 1.2.3 libpri 1.2.2 asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h ISDN Lines : EuroISDN not EuroISDN+ Junghanns OctoBRI PCI ISDN Card S/T 1+8 - S/T 2+7 : TE Mode S/T 3+6 - S/T 4+5 : NT Mode modprobe qozap ports=60 zaptel.conf --- loadzone=fr defaultzone=fr # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 --- zapata.conf --- switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres = yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 callprogress=yes context=isdn-incoming group = 1 ; S/T port 1,2,7,8 channel = 1-2 channel = 4-5 ;channel = 19-20 channel = 22-23 context=pbx-incoming group = 2 channel = 7-8 channel = 10-11 ;channel = 13-14 channel = 16-17 - Here's the output BRI debug when I try to make outbound calls from a SIP phone : -- Executing Dial(SIP/400-c8dc, Zap/1/1013) 1 -- Making new call for cr 137 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (Cool len=26 1 Call Ref: len= 1 (reference 9/0x9) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 811 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 051 411 801 341 301 301 ] 1 Calling Number (len= 7) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1 Presentation: Presentation permitted, user number not screened (0) '400' ] 1 [1 701 051 c11 311 301 311 331 ] 1 Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] -- Called 1/1013 1 Protocol Discriminator: Q.931 (Cool len=8 1 Call Ref: len= 1 (reference 137/0x89) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 871 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) 1 Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] 1 -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup, cause 100 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/400-c8dc, ) == Spawn extension (default, 1013, 2) exited non-zero on 'SIP/400-c8dc' 1 received TEI check request for TEI = 127 I've already tested several configurations for zapata.conf especially with the pridialplan and switchtype lines but without success. Could you help me to analyse and solve this odd problem ? Thank you in advance, -- Sébastien Mortier AbsysTech Tel : +33 3 20 50 99 02 Fax : +33 3 20 74 50 05 Gsm : +33 6 20 79 24 29 http://www.absystech.fr ___ --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
RE: [Asterisk-Users] Simple question about ringing multiple phones(extensions)?
Marty, Just remove the options for each technology. Dial(SIP/2005IAX/2010,25,tr) This should do the job Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: zaterdag 28 januari 2006 9:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Simple question about ringing multiple phones(extensions)? Hey Gurus, I have a very simple asterisk setup that basically lets me share a PSTN line from one location to another. I would like to have the phones at both locations ring when the PSTN # is dialed(inbound calls from PSTN to asterisk). I tried something like: exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr) I thought this might cause both 2005 and 2010 to ring when 2020 was dialed, but only 2005 rings? Thanks for ideas or suggestions on this. Marty ___ --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] Re: Some simple voicemail questions...
I think that you still should be able to use the voice mail system of our service provider. It will detect that all 3 lines are busy and reroute the call. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Brad Stockdale Verzonden: woensdag 23 februari 2005 20:31 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Re: Some simple voicemail questions... Hello all, I am working on setting up a basic Asterisk system (using Digium FXS and FXO cards, Polycom IP phones). I've got a few questions in regards to voicemail and was hoping that someone could give me some ideas... Right now we have three incoming POTS lines. There are times when all three lines are used. When the lines busy out, the incoming call is sent to a voicemail system that our local telco offers. The nice thing about it is that it never busy's out -- it always is available to take the message and store it for us... If I put this * system in place, as far as I can tell, if the POTS lines are all busy, the customer will get a busy signal and will have no chance to leave a message... My question is this -- Is there any way around this problem other than ordering more POTS line that would, most of the time, sit idle and be used only to take voicemail? If it helps at all, we have Centrex service on our lines here... Is there a way for * to somehow hook into the telco's voicemail system using Centrex? Any thoughts would be appreciated... It will be hard for me to use * here if we lose part of our voicemail capabilities... It would also be a hard sell to buy extra lines at $40.00/month per line (without centrex - with centrex that cost goes up to over 120 a month)... Getting a DCS line or a partial T isn't an option either -- DCS runs about $1,000 a month here... Voicemail is something that my company relies heavily upon, and until I can figure out some sort of solution to this problem or an interoperation between the CO's voicemail and our Asterisk machine, I can't really put this system into place. :-/ Thanks in advance, Brad Thanks, Brad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
