Re: [asterisk-users] MATH

does dtmf any any variable that i can capture and use w/ some logic like in the case of a gotoif Anyone have a clue what this means? Anyone? Anyone? How about this: does dtmf transmit any variable that i can capture and use w/ some logic like [in the case of a] gotoif Philipp --

Re: [asterisk-users] Data transfer

-- \\\|/// | ~ ~ | (- 0 0 -) +--oOOo-(_)-oOOo--+ | Philipp von Klitzing | | klitz...@pool.informatik.rwth-aachen.de | | Friesenstr.3, D-52062 Aachen, | |Tel/Fax: +49-241-4013340 | +-.oooO

Re: [asterisk-users] Linux-based hard phones?

Hi! Snom phones use Linux What hardware is it exactly on those phones? What CPU? How much memory? What size of NAND/flash? Which parts of that hardware are not supported in mainline kernel? Looks like someone out there is working on putting OpenWRT onto a snom 820. For the 3xx models

Re: [asterisk-users] Snom vs Polycom

Hey hey! Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more managerial phones * Let the customer test and decide himself * Polycom: great

Re: [asterisk-users] Siemens Gigaset + Asterisk Query?

Hi! I was wondering if you can use the base station as a outbound pots connection for asterisk. I currently have a tdm410 to do fxs/fxo ports and would like to get rid of it, I used to use a spa3102, but it only had 1 fxo (telephone connector). I like the idea of the siemans but I would

Re: [asterisk-users] Multicast RTP Paging

Hi! I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? Are you asking us if you are trying to do this? Only you would know. ;-) i have the same IP/Port to be listened on for multicast traffic on the Snom 320's. But when i make a call to 1234, the snom 320

Re: [asterisk-users] Restricting transfers between SIP phones

Hi! So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? It could probably work if you put a SIP proxy in between (ref. Kamilio). The only way to achieve what you want is to never allow a call to a

Re: [asterisk-users] DIDs PBX Multi-channel balanced audio output?

Hi! Is there a way to set this up using a single Asterisk server and the monitor process to send the various call streams to a multi-channel audio interface card? He wants Any ideas? Jackaudio? That would require 1.6. http://www.voip-info.org/wiki/view/Asterisk+cmd+jack Philipp

Re: [asterisk-users] SIP registry fails during night

Hi! Every morning I check my SIP registry to the SIP-provider. And I must conclude that during the night somewhere registry has failed. I must do a 'sip reload' to get registered again. Can you ALWAYS solve this with a SIP RELOAD, or is it sometimes necessary to restart Asterisk? Anyway,

Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered

Hi! I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Please check if this is

Re: [asterisk-users] DECT USB dongle - an Asterisk channel?

Hi! so who's writing the channel driver for it ? What for? Unless you can exploit some cool new CAT-iq future features (which exactly?) it is easier to buy a SIP-enabled DECT base station. No need to worry about another channel driver. Philipp

Re: [asterisk-users] To: Field

Hi! I have an incoming trunk that sends me calls from various usernames I have with them. Only trouble is they send invites as s...@my.ip.addr, not as the username I have with them You need to adjust your register = statement with them: Add /username to the end of it, then calls won't

Re: [asterisk-users] FritzBox 7270

, but its ISDN implementation isn't the best. Philipp von Klitzing ___ -- 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

Re: [asterisk-users] FritzBox 7270

Hi! looks interesting, indeed, but as the O.P. wanted to divert PSTN call, one would need chan_dahdi.so or chan_misdn.so/chan_capi.so (If the hardware of Fritz is capable of it) Divert-ing is a misleading term in this case. As I said, use the new firmware and register Asterisk to the

Re: [asterisk-users] FritzBox 7270

Hi! case-2 Incoming PSTN/ISDN are answered by Fritz, and then forwarded to your own Asterisk. Incoming VOIP-calls are answered by your own Asterisk. - the Fritz!Box usually doesn't answer unless you set it up for voicemail or fax - define forwarded: Do you mean normally an anlog phone

Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4

Hi Florian! yes, that's Asterisk's problem but it seems OP is talking here about something else that produces that particular message check_auth: username mismatch, have 7705, digest has 7736 I debugged the source code a bit and it is indeed an asterisk bug. However, I suspect that

Re: [asterisk-users] Wideband g711-HD vs. g711.1?

Hi! has anyone seen specifications of the codec g711-HD? This is right now spreading fast in the wake up CATiq (the DECT successor), for example in the AVM products (www.avm.de). Googling for G.711-HD only produces hits about AVM. The AVM web site is very vague. AVM support

[asterisk-users] Wideband g711-HD vs. g711.1?

Hi there, has anyone seen specifications of the codec g711-HD? This is right now spreading fast in the wake up CATiq (the DECT successor), for example in the AVM products (www.avm.de). Is this a re-branded g711.1 (rfc5391) and therefore compatiable with it, or a different animal? And what are

Re: [asterisk-users] Busy status of a snom connected to two asterisk servers?

Hi! Advanced solution: Use local channels as queue members and Custom hints. You could build a mechanism (outside of Asterisk) to sync the states of your Custom hints between both servers. I am already using local channels and will explore hints. I have not used it till now, any hints

[asterisk-users] OpenSky: Digium Skype gateway?

Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] Attacking DECT

It might not be such a big surprise after all, but recently serious flaws in DECT security were revealed and published here: http://dedected.org/ Summary in German: http://www.heise.de/security/25C3-Schwere-Sicherheitsluecken-beim- Schnurlos-Telefonieren-mit-DECT-update--/news/meldung/120988

Re: [asterisk-users] SIP request send me 482 error

There are two bug reports with patches that might (?) be able to help: http://bugs.digium.com/view.php?id=7403 http://bugs.digium.com/view.php?id=12215 Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 -

Re: [asterisk-users] Specific SIP answers on incoming calls?

Hi! specific SIP headers. Besides wrong number, I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to send a specific reply out of the dialplan? No. The dial plan does not provide such low-level access to the SIP stack.

Re: [asterisk-users] Asterisk 1.6 vs 1.4?

Hi! I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. [...] We are planning to accomodate about 5,000 users on this server. Many people on this list will advise you to use a SIP proxy like OpenSER in front of

Re: [asterisk-users] Receptionist SNOM-360

Hi! I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? Yes. One SIP account, has a limit on concurrent calls? No, not that I am

Re: [asterisk-users] Asterisk with lumenvox

Hi! I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. So what is that you'd like to know? Philipp ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Microsoft Office Communications Server

Hi! What is the logic of them using SIP over TCP? Is this a broad industry trend? Or just the latest attempt to get around SIP/NAT issues? I remember a quote of Henning Schulzrinne where he states that having designed SIP with UDP in mind was the biggest mistake he (and Mark Handle?) were to

Re: [asterisk-users] replace astdb with a cluster-capable sql database engine

Hi! So putting a translation layer so that ast_db_* API calls either go the normal route or translate to func_odbc (or another path) would improve functionality because both old and new apps would be able to seamlessly take advantage of the new database backend or keep using DB1 (the * admin

Re: [asterisk-users] Asterisk Scalability

Hi! Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls Use the Wiki, Luke!

Re: [asterisk-users] Summary: Upgrading to Asterisk 1.4

, the README file in asterisk-1.4 is outdated and refer to upgrade instructions from 1.0 to 1.2. Having said all of the above: Asterisk is coool and great, and everyone involved even more so - Olle included ;-) - thank you for all the effort! Cheers happy days, Philipp von Klitzing

Re: [asterisk-users] Hoteling

Hi! So you mean have a script rewrite the MAC-phone.cfg file, correct? If I do that, then i'll have to have the phone reboot (which i can do), but that really isn't a virtual extension anymore.. Then do it the other way around: Always use the same (virtual) voicemail box for a specific

Re: [asterisk-users] Suppressing certain queue announcement voice prompts

Hi! Short of replacing a sound file with a sound file containing only a short period of silence, is there any way to suppress certain sounds from playing during queue processing by configuring for example queues.conf or other similar files? Which announcements are you trying to

Re: [asterisk-users] Snom phones, blinking lights and call pickup

Hi! 1. Use group dial like in Dial(SIP/1SIP/2) and have your monitor phones each act as SIP/2 to SIP/6 with dedicated (!) lines that have their ringer set to silent. You might want to adjust the Caller ID name to prefix it with the called number like to 123: from 4567890. The SNOMs

Re: [asterisk-users] Snom phones, blinking lights and call pickup

Hi! 2. if the call is not answered after a short time, a light (phone layout is similiar to the Snom phones) begins to blink at the monitor phones; caller ID and the target number are displayed; if you press the button next to the blinking light, you pick-up the

Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?

Hi! 2. On the remaining locations we have a problem which I have been studying and trying to address... Faxing over IP. There might be yet another option for you to consider: Some of the bigger MFC printer/copy/fax combo devices by Brother (and maybe also other vendors?)

Re: [asterisk-users] Zaptel/mISDN and call transfer

Hi! I found an old feature-request bug in Zaptel which seems relevant: http://bugs.digium.com/3554 Not sure if this means that the feature is supported. Maybe ask Mathew Fredrikson or Digium support. by the way: Is this call deflection or ECT etc. only possible to be executed at ring

Re: [asterisk-users] Zaptel/mISDN and call transfer

Hi! I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). They keyword to search for is explicit call transfer (ECT). At least chan_capi-com (http://www.melware.org/ChanCapi) comes with

Re: [asterisk-users] Zaptel/mISDN and call transfer

Hi! I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). They keyword to search for is explicit call transfer (ECT). At least chan_capi-com (http://www.melware.org/ChanCapi) comes

Re: [asterisk-users] Sending text to a phone that's no in-use ...

Hi! Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, Hello World) but that doesn't exist ... Snom's or Grandstream GXP2000's I'm afraid... Sending text to them while in a call works fine (although

Re: [asterisk-users] Working softphone for poket PC

Hi! Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Windows Mobile 6 comes with a SIP client, however on my HTC device I still need to use the speaker phone or a headset,

Re: [asterisk-users] Call Pick Up

Hi! While a connection in progress on one extension, I would like to go to any other phone, dial some extension number, in order to ether pick up the call or join in an automatic conference. In other words, make it work like the old ma bell phone (when I want it to :-) ) Is this

Re: [asterisk-users] Querying channel variables via the Manager API

Hi! I was thinking that there must be a way to tell Asterisk give me a complete dump of all the available channel information including variables In Asterisk 1.4: show application DumpChan Cheers, Philipp ___ --Bandwidth and Colocation

[asterisk-users] regexten regcontext broken for SIP?

Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, show dialplan xxx reveals no

Re: [asterisk-users] Any Hardphone with VPNClient embedded?

Hi! Does any of you knows an Hardphone with VPN client embedded? Take a look at Zultys SIP phones. VPN enabled. www.zultys.com As I too am interested in IPsec capable hardphones (or ATA's), do you have a suggestion what to look at instead? I mean: It's nice to say the company may

Re: [asterisk-users] Auto retry on Busy

Look here: http://www.voip-info.org/wiki/view/Asterisk+tips+campon Also many so-called legacy hybrid PBX switches have had this for many a year. Hard to compete when well used features that have been around for 20 years are lacking I want something that will keep trying a busy number

[Asterisk-Users] Getting at SIP error with SIP_HEADER() ?

Hi, when attempting to dial an invalid number with Nikotel this is returned: SIP/2.0 400 Bad Request request uri is neighter my realm or a valid dns and Asterisk prints smth similar on the CLI. However it appears that I cannot get access to 400 Bad Request from the dialplan because this

[Asterisk-Users] SNOM, g722 and 16 kHz audio

Hi there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is g722/8000 which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to

[Asterisk-Users] Failing SIP registration brings * down

Hi there, this is now the second time I've seen an issue like this with 1.2.7.1, the first time it was a DNS hickup, today its some Internet congestion: When one (!) or more register statements in sip.conf fail the entire Asterisk becomes very unresponsive and does not accept registrations

Re: [Asterisk-Users] NO ringing tone while dialing

Hi! I have tried when the option 'm' , but I don?t want the default music on hold to be listened neither. I want nothing (silence) to be heard instead of ring, ring. Any idea how to do this??? The answer is in your question: Create a MOH music file with... silence in it. Then make a new MOH

Re: [Asterisk-Users] voicemailmain()

Hi! in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? A simple solution is to just edit/remove some of the voice prompts that announce the unwanted options, so the user will not be informed about their existence. Also I want to know if there

Re: [Asterisk-Users] any doc/example for app_sms.so ?

Hi! is there any documentation or simple example around for app_sms.so to get started with it and do two simple tasks: 1. send a message to an sms-capable phone connected to an ATA 2. receive a message from an sms-capable phone and so something simple with it, even just write it to

Re: [Asterisk-Users] When/whether to use SER?

Hi! 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I have yet to open my server to the Internet to be accessible to travelers without

Re: [Asterisk-Users] MeetMe Dialplan question

Hi! is the following possible? I would like to transfer a call to my personal MeetMe conference room and get transferred there automatically as well. Currently I can transfer the call to the conference, have to hangup and then call the conference number myself. I would love to have this in

[Asterisk-Users] D-Link announces Asterisk on Router/DSL-Modem

Found this today: D-Link has apparently positioned itself to challenge the AVM Fritz!Box (which is an analog ISDN home PBX including DSL modem, router, LAN WLAN). The PBX service of this device called HorstBox Professional DVA- G3342SB will be provided by Asterisk, and it should become

Re: [Asterisk-Users] Screening incoming calls.

Hi! The PBX I'm getting ready to replace has a really nifty feature -- one that I'm not even sure Asterisk -can- do -- though I'm hoping to be proven wrong. When a call goes to voicemail, the end-user can listen to the VM as it's being recorded, and can interrupt and answer the call if it's

Re: [Asterisk-Users] controlling SIP subscriptions from SNOM phones

Hi! Now, one user, not the receptionist, has gone in and set his personal numbers to these function keys thinking that DESTINATION meant setting a number to dial out. So now I have a ton of SIP SUBSCRIBE messages for his numbers. Indeed this situation is not ideal. The first thing to do in

Re: [Asterisk-Users] bristuff/zaphfc disturbing other ISDN phones

Hi! When I'm dialling through Asterisk/bristuff, and in the same time TA have some conversation (or maybe modem link) on channel 1, I can hear that conversation (or modem) very short period (0.2sec), and I also disturbing that conversation (modem). I've made a somewhat similar experience:

Re: [Asterisk-Users] Screening incoming calls.

Hi! Thanks for that post thats a good one :-) just one thing, what happens if the user doesn't want to connect to the caller? does it get saved as VM? Looking thru the code I couldn't see where that happens. The 1st MeetMe has three participants: caller AGI .call file --

Re: [Asterisk-Users] Transfer

Hi! Can * transfer call if I use canreinvite=yes in sip.conf? Can * start automon (recording) if I use canreinvite? If answers are no, then which one did you chouse for your configuration? Do you use canreinvite=yes so you can't do those stuff or you don't use this so you have high

Re: [Asterisk-Users] .call files on PRI not waiting for answer in de sired context --ResponseTimeout the best answer?

Hi! Upgarde to 1.2.1 and try again - 1.2.0 (and maybe the beta) had a bug concerning .call files and the non-passing on of variables that might affect you as well. Cheers, Philipp Hmmm seems like every dialplan snippet I've seen so far relies on ResponseTimeout and looping back to s,1. Is

Re: [Asterisk-Users] cdr_addon_mysql can't find libmysqlclient.so

Hi! Dec 13 12:19:29 WARNING[4112]: loader.c:325 __load_resource: libmysqlclient.so.15: cannot open shared object file: No such file or directory Mine is in /usr/lib/libmysqlclient.so, so how about just adding a symlink? Philipp ___ --Bandwidth

Re: [Asterisk-Users] Unable to find key

Hi! Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] Look like you have enabled

Re: [Asterisk-Users] chan_capi AVM C2

Hi! currently i running * 1.0.9 with chan_capi 0.3.5 Try chan_capi-cm instead and see if it helps. Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Softphone with Hint support?

Hi! Are there any Windows-based softphones (SIP or IAX based) that support the new Hint system in Asterisk 1.2? I don't mind evaluating commercial options, if they're available. Try the SNOM softphone: http://www.snom.com/snom360softphone.html The only other softphone I am aware if is

Re: [Asterisk-Users] Mechanisms for Implementing a Common Contact Database

Hi Douglas! 2. It'd be cool if the regcontext command actually did something. There's a myth out there that it does something like execute a command upon registration. Even the O'Reilly The Future of Telephony seems to think this. After reading some posts in the developer discussion I can

Re: [Asterisk-Users] SRV Lookups *ARRGH!*

Dear Douglas! Asterisk is really pissing me off. Can someone tell me why this doesn't cause SRV lookups to be done on outbound calls: In general: If you are missing documentation then you are warmly invited to write and enhance the existing one (e.g. the Wiki) wherever you see fit. In

RE: [Asterisk-Users] SRV Lookups *ARRGH!*

Hey Douglas, The link to the Wiki is woefully indadequate. I have no problem adding to the documentation, as soon as I bloody understand it myself. Excellent, that's what I wanted to hear! :-) http://bugs.digium.com/view.php?id=1805 http://bugs.digium.com/view.php?id=2081 The two bug

Re: [Asterisk-Users] Asterisk Dial Failover

Hi! I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are broken. If I issue a series of Dial commands, such as this: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) How about you use ChanIsAvail() before each dial

Re: [Asterisk-Users] IAX Jitterbuffer and trunking

Hi! 2) PolycomSIP - *1 -IAX- *3 - Zap TDM400 Analog 3) H323 Gateway - *2 -IAX- *3 - Zap TDM400 Analog Here is what you want to enable jb for ZAP: http://bugs.digium.com/view.php?id=3854 Cheers, Philipp ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] multiple line registrations on attendant console

Hi Dionisis, please search the Wiki/ Google for hint in connection with asterisk and you will find. Philipp I've noticed that Polycom and Snom each offer attendant console expansions. As far as I understand, the point in using all thebuttons they provide is to program them to register as

Re: [Asterisk-Users] Asterisk Call Recording and SIP canreinvite

Hi! This and Time Bandit's comment makes sense. I didn't realize that these options in the Dial string will force Asterisk to stay in the media path even if canreinvite=yes. You might even have another option: DTMF via SIP INFO Quote from asterisk-devel two days ago:

Re: [Asterisk-Users] app_queue on 1.2 ?

Hi Florian, have you check that this is not connected to bug 5810? Just a guess. Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] SNOM Shared line DevState

Hi Steef! Does anyone have some ideas on how to setup a shared line on several SNOM phones in a reliable manner? The description of what exactly you are trying to accomplish is a bit scarce, which makes good suggestions a bit difficult... ;-) Calls enter on number 123. They do not have to

Re: [Asterisk-Users] SNOM Shared line DevState

Hi! Does anyone have some ideas on how to setup a shared line on several SNOM phones in a reliable manner? Here's an option I forgot: Put the incoming caller into a MeetMe room and then ring whatever internal phones you'd like to ring. Use app_devstate to play with the lights as

Re: [Asterisk-Users] dial-out and variable inheritance problems

Hi! my test dial.out file: Channel: Local/[EMAIL PROTECTED] MaxRetries: 0 Context: mytest-in Extension: 1 Priority: 1 Set: __MYVAR1=hello As you can see, the MYVAR1 variable did not inherit, which breaks my dial-out application. This way it worked well for a long time, however an

Re: [Asterisk-Users] dial-out and variable inheritance problems

Hi again, found the matching bug report including fix SVN/CVS, just for the record: http://bugs.digium.com/view.php?id=5917 Philipp my test dial.out file: Channel: Local/[EMAIL PROTECTED] MaxRetries: 0 Context: mytest-in Extension: 1 Priority: 1 Set: __MYVAR1=hello As you can see, the

Re: [Asterisk-Users] AGI Problem

Hi! The first result is ok (-1) but not the second and the third. Why do I get different results for the same command? Hm... u might want to try this: # Check print EXEC ChanIsAvail IAX/24\n; $result = STDIN; print VERBOSE \$result\ 0\n; $result = STDIN; # Check print EXEC ChanIsAvail

Re: [Asterisk-Users] Re: SNOM and 1.0.9

Hi! I still cannot get this to work on 1.0.9. exten = 451,hint,sip/451 * Try hint,SIP/451 instead of hint,sip/451. The bugtracker has an open ticket on case-sensitivity of the hint priority. * Make sure that in the advanced settings your SNOM is set to not filter packets from registrar

Re: [Asterisk-Users] SNOM and 1.0.9

Hi! on 1.0.9 the lights work. In this way: person is on the phone: light is on Person is not on the phone: light is off since 1.2 the lights will blink when the phone is running and above states work the same. Side note: Asterisk v1.2.0 comes with a new sip.conf setting:

[Asterisk-Users] SIP softphone with subscription/hint support?

Hi there, for testing purposes I am searching for a freely available softphone that supports SIP subscriptions and display the status of a few of these via e.g. a simulated LED. I know about * EyeBeam (not free) * SNOM softphone (needs Win XP and has old firmware) Are there other softphones

[Asterisk-Users] hint for MGCP (devicestate): bug 5515

Hi there, for all those that'd like to also see MGCP devices to work with the 'hint' priority I'd like to notify you of a patch in Mantis by gkloepfer that happily awaits your testing (in Asterisk 1.2.x): http://bugs.digium.com/view.php?id=5515 With this patch applied phones like snom or eye

[Asterisk-Users] Dial() and j option: What is correct?

Hi there, as you probably know Asterisk 1.2 comes with a new Dial() behaviour on busy. However I find conflicting documentation - which one is correct? j - Jump to priority n+101 if all of the requested channels were busy. j - this option prevetns jumping to an extension n+101 So will I

[Asterisk-Users] Solved - Re: 1.2 won't compile: res_config_odbc.c

For the archive: Upgrading from unixODBC 2.0.7-3 to 2.2.0-5 solved the problem for me. rpm -e libodbc++-devel-0.2.2pre4-12 rpm -e unixODBC-devel-2.0.7-3 rpm -U unixODBC-2.2.0-5.i386.rpm so far I didn't succeed in getting 1.2 compiled on a RH72 System (with gcc 3.0.4). I'd appreciate any

[Asterisk-Users] 1.2 won't compile: res_config_odbc.c

Hi there, so far I didn't succeed in getting 1.2 compiled on a RH72 System (with gcc 3.0.4). I'd appreciate any tips... ;- Cheers, Philipp gcc -shared -Xlinker -x -o res_features.so res_features.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing- declarations -g3

Re: [Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

really like to consolidate everything towards SIP; with MGCP e.g. call pickup for a SIP device doesn't work, and also call waiting isn't possible. -- * Philipp von Klitzing* * D-52062 Aachen, Friesenstr.3, GERMANY * * Tel/Fax: +49-241-40.133.40 * * [EMAIL

Re: [Asterisk-Users] Application return values

Hi! The answer is simple: You can't. The Asterisk documentation should stop including this information (for the user level) to prevent this kind of confusion. Cheers, Philipp ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] CAPI - displaying individual MSN

Hi! msn=8304490 incomingmsn=8304490 Each user has a different numer, e.g. 83044910, 83044911, 83044912 and so on. This number should appear on the display of the called party, but how do I configure this? With the above configuration the display always shows 8304490. I've tried to

[Asterisk-Users] CVS won't compile: res_odbc error

Hi there, admittedly this is a slightly aged RH system. ;- Yet what I'd like to know is if this compile problem is clearly due to the gcc version, or if there is a way to solve the issue...? Cheers, Philipp [EMAIL PROTECTED] src]# gcc -v Reading specs from

Re: [Asterisk-Users] updating display of a hardphone based on agentslogging in

Hi! On Mon, Oct 03, 2005 at 05:41:38PM +0200, Mark Elkins wrote: I'm also using SNOM320/360 phones. Ideally - set up one button to toggle the Agent Status (in/out == On/Off) ??? Kinda make sense if app_devstate (or similar) made it into mainstrean Asterisk - so line indication lamps could be

[Asterisk-Users] Can't compile zaptel (CVS Head) on Debian

Hi there, with both CVS HEAD and 1.2beta1 I don't succeed with either make or make linux26. I checked more than once to make sure the required packages are in place - any suggestions? Philipp /lib/modules/2.6.8-2-686/build make -C /lib/modules/2.6.8-2-686/build SUBDIRS=/usr/src/zaptel modules

[Asterisk-Users] sip SUBSCRIPTION bug in 1.0.9

Hi! Asterisk 1.0.9 (maybe also earlier versions?) contains a bug the effectively disables subscriptions for phones with multiple regististrations in place. When processing a nonce response as a result to an 407 authentication request Asterisk with SIP DEBUG reports Found peer YYY even though

Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards

Hi! We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. Have you tried a different sound card and/or a USB handset (which includes an external sound card)? And what exactly do you mean with terrible sound?

Re: [Asterisk-Users] updating display of a hardphone based on agents logging in

Hi! I haven't done this myself (yet), but look at app_devstate that comes with the bristuff patches to toggle one of the SNOM LEDs as needed. http://www.voip-info.org/tiki- index.php?page=Asterisk+cmd+BristuffDevstate Other - more clumsy - ideas would include a) making Asterisk call itself

Re: [Asterisk-Users] SoftPhones: Bad, or just bad QoS?

Hi! Hi again, folks. I've been getting feedback from this list and elsewhere that softphones are generally not considered good enough for hardcore business use. Can someone point me to where I can find more detail on this debate? - you comp needs to have its speakers turned on in order

Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

Hi there, can't resist to add a few thoughts as well... Hi, folks. I am planning on implementing Asterisk in 2006, and need to budget for it now, so I need to know what I'll need to get. My company has about 50 users, and is currently languishing on a very old Comdial PBX. Wiley is

Re: [Asterisk-Users] Grandstream firmware 1.0.6.2 - T.38!

Hi! I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ 2 quick notes, a quick test seem to indicate iLBC is broken (didn't try any troubleshooting). The release notes have a couple of comments on iLBC, you

RE: [Asterisk-Users] 8+ line receptionist only setup

Hi! Well only the receptionist and higher level authorities will have the cisco 7960. For the rest I am probably thinking of a Sipura or Snom phones to keep costs down. I would suggest that you take a good look at the SNOM 360 instead of Cisco. Cheers, Philipp

Re: [Asterisk-Users] Router with QoS recommendations

Hi! As I have a Cisco PIX 515, with NO QoS functionality, and I™m looking for a router that does outgoing QoS to put in front of my PIX. Problem is that I™m using my 768/8096Kbit ADSL for both data and VoIP, and as soon as data is being sent to the internet the sound quality drops to

Re: [Asterisk-Users] Snom and Multiple calls

Hi! On the snom (I've tested this on the 220 and 360), the phone will immediately reject any new INVITE that arrives with 486 BUSY HERE if there's already a call on the phone opening That is very interesting - can you present a review of the Snom 360 hardware, even if it is a short one?

Re: [Asterisk-Users] Hold Pickup

Hi! seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial prefix12345 to pick up a call holding on ext. 12345. It looks like the closest to what I want (without changing Asterisk) would be

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