How can we dial via pc->sip, like SjPhone? allows us to do? That would be fantastic if we could do that, for evample I would like to dial this url:

sip:9991425550@sip.lon.myPBX.com

-John

John, just type 9991425550@sip.lon.myPBX.com into the text field and hit Return.

How can I simulate a "switch hook flash" to enable me to transfer or conference calls?

That's not supported yet.

Is there any way to disallow certain codecs being negotiated?

Not yet.

For my provider (xs4all), the following DTMF is supported: - inband - G711 - RFC2833 and inband - G729 Comparing that with the DTMF info on this page, does that mean I can not use DTMF with Telephone?

You won't be able to send DTMF in Telephone with that provider, I suppose.

How can I get a SIP address ?

You should choose a SIP provider and register an account. Any SIP provider should work. Also take a look at our tested providers list http://code.google.com/p/telephone/wiki/ProvidersTested.

SIP address is usually a combination of user name and domain separated by @, e.g. user@domain.com. Just like email address.

Hi, any information on setting up an account with IPkall? It was posted on this site before, but I has been removed and I am not able to set it up. Thank you

First of all, we didn't have any information about IPKall before. IPKall does not look like a SIP provider. You can't get a SIP account there and use it in Telephone. But you can sign up with any other SIP provider, get the phone number at IPKall and forward it to the SIP address, which you received from the SIP provider. You can set up Telephone to use that SIP account.

Is there a way to enter a pause into a phone number string? Useful for when dialing a number that answers and then asks for an extension, or conference calls that need a meeting # etc. iPhone uses a , for a pause, for Address Book compatibility that would be great.

I have done some searches here on this site for information about PSTN or Public Switch Telephone Network. I may not fully understand the extent of the capabilities of SIP but I have tried to make some local calls to landlines and it fails because "PSTN not permitted" Could someone explain this please?

What SIP provider are you using? Does Telephone connect successfully? I mean, do you see Available state? If so, that could be your SIP provider which doesn't allow you to call PSTN (regular phones) unless you pay some money.

iptel.org is the SIP provider I signed up on. I will try one of this site's recommended providers.

It appears it will be a challenge to find free PSTN providers in USA.

I have a Lingo account but do not know what their "registry server" address is. Could not find anything on google. This is the SIP: "number@asdcsv.bw.iprimus.net".

Have you tried "asdcsv.bw.iprimus.net" as a Registry Server?

Is it correct that this application does not support Asterisk as a SIP-Server ? Because Asterisk only talks SIP on UDP 5060 and Telephone apparently tries to use TCP.

Telephone does support Asterisk. Telephone uses UDP to initiate SIP requests.

You can send me Telephone.log (after increasing the log level) to eofster@gmail.com.

This says it's a universal binary - but doesn't run on my power PC PowerBook? G4 - running it from the shell results in a Bus Error, leading me to believe this is an Intel only binary.

Telephone should work on G4. What Mac OS version are you using? Please note that only Leopard is supported.

is caller ID not supported? i use voipdiscount.com and have set it up to display my home number. however, when i dial from telephone.app, the number is not displayed.

It's not something you save on the server, it's a client option. I guess you should try to change SIP Address for that account to something like "<your verified callerid number>@<your sip server>".

Alexei, great job! Telephone is intuitive to use and well integrated! Feature Request: lists of missed, received and dialed calls with time stamps and duration. (Windows of missed calls could be automatically closed, then.) Thank you!

Thanks, Andre! I hope call history will be available with the 0.15 release. You can track changes for this issue here: http://code.google.com/p/telephone/issues/detail?id=58. Log in and star that issue to receive notifications, if you'd like.

Hi, in my case I need to put 9+actual_phone_number+PIN

Is that possible?

No, it's not.

I see I can add multiple sip accounts - but they have different stun servers. I can only see one setting for stun servers in network. how does it work to det-up multiple sip providers?

Thanks

You can choose STUN server from any of your providers or even a different one. It is used only to determine your external IP address when you're behind the NAT.

what's the SIP address and the registry server ? i just downloaded this app. and i don't have these infor. ?

It depends on your SIP provider. Here are some providers that our users have been tested with Telephone http://code.google.com/p/telephone/wiki/ProvidersTested.

Do you have plans for Snow Leopard? any areas in particular that you expect some issues? like CoreAudio? for example?

BTW, i really like Telephone. nice job!

I haven't try to run Telephone on Snow Leopard yet. I have plans to support it, of course.

Wonderful piece of software. Fantastic. Only one important thing missing for me, being able to transfer calls. Keep the good job and the simple gui.

Thank you.

Awesome, works great with Google Voice + Gizmo5. (Totally free!)

ubelievable software. love it. what i would like to have, is a menubar icon instead of dockicon

stoneh, you may want to star the corresponding issue to vote for it and to receive updates on it http://code.google.com/p/telephone/issues/detail?id=73.

Thanks, kawarmc! You may also want to star the issue for the "call transfer" feature http://code.google.com/p/telephone/issues/detail?id=31.

I just wanted to comfirm that Telephone and sip-provider terrasip works extremely well togheter. www.terrasip.com

Great application.

Hi guys! Thank you for a great application... If I want to submit a 'sip:' command, like the OP in this list, say of '9991425550@sip.lon.myPBX.com' but from FileMaker?, can you tell me how I configure a system to have this 'sip:' call passed to your Telephone.app rather than any other app on the user's machine?

As far as I know, Mac OS X does not have an "official" way to do it. But I've heard about RCDefaultApp which seems to allow one to set a particular program for the particular URL scheme, e.g. "sip:". And, of course, if you delete all other applications that handle "sip:" URL scheme, Mac OS X should pass such requests to Telephone.

Hello, I am trying to configure Telephone in mac with following SIP provider: nonoh.net and webcalldirect.com. I got this message.

Could not register with sip.nonoh.net. The error was: “408 Request Timeout”.

Can anyone suggest me what to do.

Thank you. Brundaban

First, try to disable ICE and DNS SRV.

@eofster: There is no built-in UI for selecting the default application for a given URL scheme but there is a public API. The Launch Services framework offers calls to check and register the application preferences. Telephone could call LSCopyDefaultHandlerForURLScheme to see if it is currently registered for the sip: scheme and if not it could ask the user if he or she wishes to change this. If the user says yes then Telephone can call LSSetDefaultHandlerForURLScheme.

Thanks for the hint, nickovs. I'll take a closer look at it.

would it be possible to set the panning, like Skype does? The microphone on my macbook pro is located on the left side. When I call with Skype, the audio only comes out of the right speaker.

I don't think it will be possible in the nearest future.

Do you have the same problem with Telephone?

Call transfer is a HUGE feature to add next... plus the ability to have a website opened when an incoming call is accepted. For example, let's say a call comes in from 8778779473 with caller name of PhoneWire?. The URL could then be http://www.phonewire.com/lookup.cgi?id=8778779473&name=PhoneWire or something similar that the user could edit where in the URL the variables are inserted, so contact management and SalesForce?.com integration with "screen pops" would be now an available feature just by opening the correct URL for the call!!!

@eofster About the sound: I think I didn't explain myself very good. In order to cancel the audio feedback, it's a good thing Skype only plays the audio stream on the right speaker. This kills all possible feedback and works like a charm. In this way, there is no need for a headset.

When trying to connect to a Panasonic PBX i get the error : 423 interval too brief Any idea about what i can try to configure ?

Yes, you can increase reregistration time interval, it's in the advanced account settings. You should set it according to your PBX config. If you don't know that time, you can try values like 600 or 1800.

Works great with 600, thank you !

Which settings are necessary for dialing SIP-addresses? Telephone fails on them with time out while another client dials sip:something on the same account.

Thanks for great application! Address Book integration rocks.

Just type "john@company.com" and hit Enter.

You can also just type "john" if you're both in the same "@company.com".

@eofster: Yes, I'm entering a "user@service" like address and then getting a time out. By the way, I have an IP-address as a domain. If I use a corresponding hostname, Telephone doesn't log in. DNS records are ok in both directions.

Do you have DNS SRV enabled? Try to disable it. If it doesn't solve your problem, start a new issue, write to the discussion group, or send me an email to eofster@gmail.com.

Just wanted to note that, it works great on Snow Leopard, besides the fact, and idk if this is just my case or.... but when I recieve a call from a PTSN > PC that the pc cannot hear what is being said from the phone, but can hear the other way around, and when the call is made from PTSN > PC it works just fine...

Also theres gotta be a way around gizmo's 3 min call out limit, hmm?

Nevermind, anwser was in the wiki, my apologies, I just had to insert the proxy address because of one way audio... Great app!

Awesome app! I've recommended it to all my buddies. Just want to confirm that it works very well with Acanac.

iTunes & iChat integration would be nice features to have. E.g. When you receive a call, iTunes pauses and iChat's status changes to something like "On the phone".

Actually, Telephone does pause iTunes for the call duration. There's a checkbox in preferences for that which should be checked by default.

And thanks for the idea with iChat!

Ok...i'm very new and dumb with this: How do you create an "account?" It asks for "full name" "Domain" "Username" and "Password". What would be the domain, etc?

You should choose a SIP provider. It will give you all those settings, just like with registering an email. Also, we already have some providers tested: http://code.google.com/p/telephone/wiki/ProvidersTested

Wonderful application. A pleasure to use and setup.

When dialing digits, like for an extension when using a SIP PBX, it says "Calling" and then I receive message "Address Not Found". I can receive calls with no problem though.

Well, that’s what your server is returning. Usually you must follow some specific dialing rules. For example, add some prefix or use international number format, i.e. +<country code><area code><phone number>. Often you can find those rules on the provider’s site.

Is there a function, presently or intended, to put 'Telephone' into the menu bar? Having an open window and icon on my dock is a bit of a pain.

Yes, there is a feature request: http://code.google.com/p/telephone/issues/detail?id=73

Is there a way to load a .csv into the phone and have it dial each number in order as soon as the line is hung up? This would be similar to the iDialUdrive app for iphones.

No, Telephone can’t do that.

Is Applescript support coming? Id like to script LaunchBar? to dial through Telephone.

I don’t think it will be available in the nearest future.

Great simple app! Thanks About the sip: scheme. There is a documented way to have an application register to open specific scheme. I've done it on an iPhone app and seen it on desktop apps. I can't find the link right now since my system is in the middle of snow lep. migration.

Yeah, we’ve got an issue for that: http://code.google.com/p/telephone/issues/detail?id=193

I am trying out Telephone with Sipgate and get the message that symmetric NAT is detected. I have also trialled iSoftPhone and have no problem with two-way conversations with my current router and firewall configurations. Why should Telephone have a problem and what I can I do about it given that I am sometimes behind a corporate firewall (where iSoftPhone works fine)? If I can sort this problem out then I will probably stick with Telephone.

Try disabling STUN in the network settings. Empty "STUN Server" field, close preferences window and hit Save.

Thanks for developing this great little app -- I'm sure it will blossom into a great tool at maturity! A SIP client is something Apple should have built-in!

Great App! Tell me about the Address Book plug-in, that would be great. MBP13 Snow Leopard.

Spoke too fast. I just figured out the plug-in. Simply click on the label for the phone number. Very nice.

great app, but can never hear my callers. I come in crystal clear on their end, but I hear nothing.

Any ideas?

Maybe your provider wants you to setup a STUN server in preferences. What provider are you using? Please write to your discussions group http://groups.google.com/group/telephone-app

I have an ipod headphone with the built-in mic and a macbook pro, but the app can't seem to detect the mic.

http://www.tuaw.com/2008/10/15/iphone-headphone-mic-works-with-new-laptops/

Does the mic work in other apps, iChat for example?

Hallo! Everything worked fine till today: "Could not start SIP user agent." What can I do? thanks for you help :-)

Please increase the log level and send Telephone.log to eofster@gmail.com. See this FAQ for the details.

Loving Telephone, but I occasionally have a heck of a time entering conference call numbers. It will often get repeated key presses...

Telephone ignores repeated key presses when entering DTMF. I think this collision could happen between the VoIP gateway and PSTN.

How do I dial a mobile phone from this? I am a novice user. Using Pennytel. I can dial pstn numbers fine. I tried with country code in front of the mobile eg. 61428 xxx xxx and with out...niether works

You dial them the way your SIP provider tells you to dial.

some how I can not install telephone on my mac ( intell with OSX 10.4.8)

i can not find any installtion requirements or other

Telephone runs on Leopard and Snow Leopard only.

Hi, very nice app design, simple and efficient. I have 2 issues. First is that I get into a loop when entering network preferences. I'll change them once, and on subsequent entries, I will be prompted to "save, don't save, cancel" my changes, even when I haven't made any.

Second issue is that I'm getting 1 way audio only, callers can her me, DTMF inputs, but I can't hear them. I had a similar issue w/Xlite, until I altered the "listen on IP" setting there, as I'm behind a VPN. IS the "TransportPublicHost?" key the equivalent setting here?

Thanks!

The first issue is fixed, but hasn’t been released yet. Just leave ‘Outbound Proxy Port’ filed blank is default 5060 is okay for you.

To fix the audio, I think you should enable NAT traversal on the server side or use STUN server in Telephone.

Don’t change ‘TransportPublicHost?’ in Telephone, unless you absolutely sure what you’re doing.

I would like to use "Telephone" for conference calls. I first dial the phone number of the conf call provider, when I get the message I then have to dial the conference number (for example 338877 or 338877#) and finally the host number (3322 or 3322#). But I can't do it with Telephone. It doesn't recognize the conf number neither the host number. How can I solve that ? Thanks

There are others who also reported problems with sending DTMF via free.fr provider. I haven’t investigated this yet.

Can I use G.722 Codec. My provider Sipgate.de supports it. How can I be sure, that Telephone uses this codec?

You can’t be sure unless there is some checkbox like ‘G.722 only’ from the server side. If will be possible to force any codec in Telephone when issue 37 is implemented (http://code.google.com/p/telephone/issues/detail?id=37).

There are any plans to release a new version with FLASH support? if yes, when? Thanks

What do you mean by flash, call transfer? I’m working on it right now.

Yes, that's it: call transfer. I press flash button and transfer to another phone number in my company. When it will be release? BTW your software is better than the others, because it's very light. Congratz.

Hello I'm from Brazil You have done your application is very good!!!

REVISED POST with working script!

mc.willprater above was asking for AppleScript? support to script LaunchBar? to dial through Telephone. In fact because Telephone handles tel: URLs*, you can do it without scripting support in Telephone itself. The following will work as a helper script for LB. It's pretty clunky -- I'm not much of a hacker -- but it does the job.

on run
	set the_number to text returned of (display dialog "Dial what number?" default answer "")
	handle_string(the_number)
end run

on handle_string(s)
	set s to CleanTheNumber(s)
	set s to "tel:" & s
	open location s
end handle_string

on CleanTheNumber(numToDial) -- remove punctuation from a string, leaving just the number
	set theDigits to {"0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "+"}
	set cleanedNumber to ""
	repeat with i from 1 to length of numToDial
		set j to (character i of numToDial)
		if j is in theDigits then set cleanedNumber to cleanedNumber & j
	end repeat
	return cleanedNumber
end CleanTheNumber

* http://www.rubicode.com/Software/RCDefaultApp/ to set which apps handle which URLs you can always use RCDefaultApp?

ps that was almost certainly based on a script I found to dial with something else a while back. Credit wherever it was due.

I am having the hardest time getting it to work with my Plantronics USB headset (on Snow Leopard). I can hear people, but the microphone does not seem to work. Can't get the Mic to work with ANY SIP client, but Skype and Gizmo5 work

I'm trying to use a provider that requires a specific User-Agent. Since you have great documentation on using the sources, I've changed the User-Agent there and recompiled to find out that the proxy always returns "404 not found" on REGISTER because of the "Route:" header (I've confirmed it, using a proxy to remove it and then send it to the provider). Is there some configuration I'm missing to avoid setting the "Route" header? I've been search the sources (both Telephone and pjsip) where to force it not to send that header, but can't find it. Any tip would be appreciated as this provider's Windows client doesn't work with Wine, so I really can't use anything at the moment..

Thank you very much

The best place to ask such questions is in our discussions group, because it is easier to follow conversations there (http://groups.google.com/group/telephone-app).

‘Route’ header is added when you’re using proxy — either ‘Connect using proxy’ in the account settings, or ‘Outbound Proxy’ in the network preferences. It forces Telephone to send all requests through that proxies. If you leave those fields blank, there won’t be ‘Route’ header. And if you’re dialing only phone numbers, Telephone will still send SIP requests to the server you specified in the ‘Domain’ field.

Sorry for posting here. The problem is I need to specify a proxy but I don't want it to use Route header. Maybe it's a protocol thing, because qutecom doesn't send Route header. I'll repost in groups for more details, sorry once again.

any update with new cool feautures coming anytime soon? btw - the app rocks!

Yes, call transfer is in the testing phase.

Uploading a torrent using "transmission" client with the specified speed limit of 5kb/s causes faulty speech transfer from my side, although I can hear the opponent without a glitch. The uplink bandwidth seems to be enough - up to 100kb/s, and when I quit transmission and reestablish connection - everything goes fine. It also seems skype doesn't behave that way - I'll try to continue experiments in the nearest future.

А мне на мой SIp позвонить могут? 3CX что-то не может никак с этим чудом работать на входящие, либо telephone не принимает входящие? никак не могу разобраться "куда думать"

stranniknavsegda, да, позвонить могут, Телефон никак не препятствует этому, если он в режиме “Доступен”. Если не приходят входящие звонки, надо смотреть в сторону сервера SIP регистраций. Если вы совсем не знаете откуда начать, можем собрать логи. Напишите мне на eofster@gmail.com, там проще вести диалог.

I use Telephone with Google Voice through Gizmo ( I know, you can't get an account now ). Then I use GV Connect to dial numbers, pick up with Telephone, done. Wow. So great.

Hi, Telephone is working fine but it seems that stun01.sipphone.com doesn't work, strange since it is a a well known stun server. Someone had error 171023 when dialing mobie number, how can that be, coz landline or shortcuts numbers where working ok What is 171023 error?

It’s pjsip error PJSIP_EMSGTOOLONG. Try to disable ICE in Telephone.

Hello, how can i press a number after i dialed? for example i call for a my banking and then i have to press my password number. How can i do that using telephone software? thanks.

Just press a number button on the keyboard.

VERY nice app! Is there a way to define where the notification window appears on an incoming call. (Scenario: i have multiple monitors and i would like to get to the notification window to answer the call even my 3rd monitor is switched off.) Thanks

Excellent application. THANK YOU.

katakefalos, are you talking about Growl notification (a translucent black rectangle in the upper-right corner)? There is a setting called ‘Screen’ in System Preferences > Growl > Display Options which, I believe, does what you want.

eofster: no i was talking about the actual pop up from telephone.app on an incoming call but it was very simple i just had to close out all pop ups and the last one i closed should be on the display where i want the next pop up to appear. Thank you for your response! A nice addition would be to have a transfer or conf button with sip functionality.

Any plan to support Message Waiting Indicator?

Is there a setting (or is there one on the horizon) to automatically enter country code. A lot of my contacts in my address book don't have the country code for phone number. Do I have to update the numbers of my entire address book, or is there a setting to automatically add that (like how Skype automatically adds it...)?

glavoie, from Telephone’s standpoint call waiting is always enabled. But if we’re talking about SIP provider’s call waiting option, I don’t think it is part of the SIP protocol itself.

hounsell.al, there is no such setting at the moment, and not in the horizon. Issue 98 might be related to this question.

Hello eofster,

I'm talking about this:

Used to tell the device/sofphone there are unlistened voicemail messages. Most (if not all) VoIP phones/ATA support this and many (most?) softphones also support this.

Like I said, I would really like to see this available with Telephone to know if I received messages.

my SIP provide requires a country code can you make it so there is always a prefix to the dialed number?

How can I make a conference call?

подскажите, а "телефон" работает с cisco call manager?

Я не совсем уверен, что точно обозначает Cisco Call Manager, но если это IP-АТС, которая умеет работать по протоколу SIP, то Телефон с ней работать должен.

Работает она с Cisco Call Manager. Как оказалось, надо делать ручные настройки. То есть допилить SIP адрес до формата XXXXX@192.168.1.120, а в общий настройках использовать логин (у меня, например, лдап авторизация), а не номер. Спасибо автору за простую и удобную программу.

While in a public library

Auto login failed:

Could not register with sipgate.co.uk

The error was: “408 Request Timeout”.

Help pls.

Does it work in other network locations? Public library probably blocks some UDP ports or UDP protocol itself.

0.15 при звонке с любого аккаунта ругается "Максимальное время обработки запроса истекло". Откатился на 0.4.13 - всё хорошо. В чём может быть проблема? SL 10.6.4/ 64-bit

отличный софт! я прям рад что есть такой! большое спасибо автору! :)

microjungle, http://code.google.com/p/telephone/issues/detail?id=324

Microphone of my logitech pro 9000 doesn't work, although that device is recognized by my system in system preferences (leopard 10.6.2). Any hint?

gabi.thoma, do you see it in Telephone's sound settings? Do you see it in System Preferences > Sound > Input?

the following script works well in skype

tell application "Skype" to get URL "callto://+1xxxxx"

Does anyone know what's the working script for Telephoen?

prramesh, tel:+12345678901 and sip:user@example.com should work.

no luck,

tell application "Telephone" to get URL "tel:+12345678901"

this one didn't work. (Telephone got an error: Can’t get URL "tel:+12345678901")

I can't tell about Apple Script, but "sip:" and "tel:" URI schemes are supported in Telephone. For example, it will make a call when you click such links in Safari or pass the URI as an argument to the "open" command in Terminal.

Dialectic (http://www.jonn8.com/dialectic/) will allow you to dial Telephone via AppleScript? as well as from tons of other locations (e.g., contextual menu support for selections in a web browser, from LaunchBar?, Butler or other launchers, etc.).

Замечательная программа. Спасибо. ИМХО, в программе не хватает просмотреть список последних набранных/принятых номеров с возможностью перезвонить. И если немножко отойти от минимализм-way, сделать своё дополнительное окошко контактов из "Адресной книги" в формате <Имя контакта - телефон>. Дело в том, что при большом количестве контактов, помнить точные имена каждого становиться сложно, к тому же автоматическое дополнение работает при условии точного совпадения начала имени. Две скромные кнопочки в окошке - не помешают минималистичности и простоте программы.

Hello I am new to both mac and telephone. I downloaded telephone and i tried to call using my Sip account I am hearing the receiver's voice but they are not hearing my voice. I tried to change my sound settings. But still I don't have any solution. Can any one suggest me any solution?

is there a way to let telephone use the systems default sound input and output, so that i can take a call with the integrated speakers and microphone of my macbook and plug in my usb headset later to get better audio quality without disconnecting the call? when i plug in my headset, snow leopard switches input and output to my headset, so that for instance itunes music switches to my headphones instantly.

sebastian.heyden, Telephone remembers sound input and output that have been selected in its own preferences.

Is there any way to select the DTMF type? In the voip client from freephoneline.ca there's an option to select either 1) In-band (Tone) or 2) Info (SIP).

A recent issue I had when trying to pickup message /w that other voip client was fixed by switching the DTMF type so it'd be nice to have the feature in Telephone also.

Telephone is great. I really like it and use it every day. However, I'd love to have a keyboard shortcut to answer an incoming call. How can this be done?

Hello,

great App. One question. Is there a way that the window of an incoming call disappear automatic, that I have not answered? I have to close every single window.

Hi, do you know that max number of registred account at the same time is only 8 ? After registering 9th account, application just fall with mac os x bug report and is not able to reopen again without editing plists in mac os x. Is there a way to increase the number of registred account ?

Thanks in advance. Eva

По функционалу оптимальная программа! Но, возникла проблема с multifon.ru (Услуга Мегафона). Входящие вызовы принимаются, с исходящими проблема, пишет что вызов запрещен, или вешает приложение (причем, какое то время после этого есть проблема с соединением c сервером в др приложениях) Пожалуйста, протестируйте с meltifon.ru

Hello, May I help you to translate this fantastic software in french ? Thanks

multifon.ru от мегафона, подключается, но не дает делать исходящие звонки. Пишет "Вызов запрещен". В чем может быть проблема?

Hi i'm from Germany and my Problem is that:

Since we got the new Fritz-Box Software Update the Callers Voice is very deep. What can i do ?

Greets from germany

Hello, thanks for making this nice proggie, I just realised that you're reading the numbers from the Address Book; however, Address Book entries that are marked as 'company' (10.6.x) don't have their respective name appearing in the drop-down list of search results in Telephone. Perhaps when you have time you can check and fix it? Normal contacts (name/surname) appear when typing numbers; but numbers belonging to companies, do not appear with their name! Thank you.

Is there a way to have all accounts to be Available on startup? I have 3 accounts enabled but only one starts as Available, the other 2 launch as Offline

bayxsonic, please write to our mailing list. Do other accounts become Available if you select that manually?

eofster, is this your mailing list? http://groups.google.com/group/telephone-app?pli=1 And, yes, the two accounts work correctly once I manually select Available.

Okay finally we found out that the problem is that we must deactivate the G.722 Codec and then it works.

How can i deactivate this codec in your Programm ?

Please Help Us!!!

Thanks

Здравствуйте! Не могу настроить услугу Мегафона Мультифон. Вот ссылка на их страницу с настройками: http://www.multifon.ru/publications/settings_sip/index.html Добавил аккаунт в программе X-Lite Beta - все работает. В чем может быть проблема?

This is the perfect softphone! Thank you for an excellent product!

On a german localized Mac OS, I have two entries for telephone in the system preferences pane of growl: One labeled Telefon(german for telephone), one labeled Telephone. Is this the intended behaviour?

06/01/2010 установил программу из Appstore. Так как не нашел способ связаться с автором, то пишу сюда. Прошу прощения :)

Для моих целей настроил три акаунта у одного и того же провайдера (sipnet.ru). И так, мои замечания:

1. При сворачивании всех трех окон в док получить доступ ко всем окнам я

могу только через HyperDoc?, или меню Findera. Отсутствует список окон в меню программы в Доке. 2. При поступлении звонка я не вижу на какой акаунт поступает звонок. И в заголовке окна и в информации о звонке отображается информация о входящем. 3. Хотелось бы, что бы окна разных акаунтов могли склеиваться что позволило бы более удобно их передвигать на пространстве экрана. 4. Команда на дозвон интуитивно не понятна (нажатие Enter). Не мешала бы банальная кнопка "Звонить".

vitaliy.anatskiy, получить доступ ко всем окнам можно через меню "Окно". Также работают грячие клавиши Command-1, Command-2 и т. д.

Using Telephone Version 0.15.2 w/ Mac OS 10.6.6. Advanced account preference setting set to Reregister every: 1800 seconds. Following is snip from Telephone.log file:

12:00:54.767 pjsua_acc.c Xxxxxx Xxxxxxx <sip:1777XXXXXXX@callcentric.com>: registration success, status=200 (Ok), will re-register in 62 seconds 12:01:51.834 pjsua_acc.c Xxxxxx Xxxxxxx <sip:1777XXXXXXX@callcentric.com>: registration success, status=200 (Ok), will re-register in 63 seconds 12:02:49.902 pjsua_acc.c Xxxxxx Xxxxxxx <sip:1777XXXXXXX@callcentric.com>: registration success, status=200 (Ok), will re-register in 64 seconds 12:03:48.970 pjsua_acc.c Xxxxxx Xxxxxxx <sip:1777XXXXXXX@callcentric.com>: registration success, status=200 (Ok), will re-register in 65 seconds 12:04:49.224 pjsua_acc.c Xxxxxx Xxxxxxx <sip:1777XXXXXXX@callcentric.com>: registration success, status=200 (Ok), will re-register in 60 seconds

Why is Telephone re-registering with the SIP server every 60 seconds instead of 1800 seconds?

dennis.janecek, that's because Callcentric itself tells Telephone to do so (returns it in the reply to the registration attempt).

2 finabmail, все работает с мультифоном p.s. хотя сам пробывал пару месяцев назад, не разрешало зонки, мож мегафон дурил...

I'm trying to get Telephone v0.15.2 on my Mac running MacOS X 10.6.6 working. Configuring my SIP provider appears to have been accomplished, only when trying to initiate a call a window pops up, stipulating that my bluetooth headset cannot be connected. Funny though I don't even have got one around!

Contro.Versell, that's a known issue that should be fixed in the next App Store release (Telephone 1.1). For now the only solution is to remove paring for this headset in Mac OS X Bluetooth preferences.

Two questions please: 1. Do you plan to include noise cancelation code at some point? 2. I see Viatalk is listed as one of the working providers. However, I appear to be only to initiate outgoing calls and not incoming calls. Do you have any suggestions? Thanks for your work.

Hi, do you have any plans to port this app to iOS devices? There are a number of sip clients for iPhone but none of them is as simple as this, and almost all look like the UI was made by a 6 year old with a crayon.

Cheers, Stephen

LOL on the UI comment of the previous poster. You are absolutely right!

Works perfectly with placetel.de as a SIP provider

Не могу подружить с мультифоном. Каждый раз говорит на найдено 404, хотя виндовый клиент подключается на ура. И в логе ничего не появляется (поставил LogLevel? 4) , только записи при старте программы

Лог такой:

REGISTER sip:multifon.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.1.83:56468;rport;branch=z9hG4bKPjU9hcfhKtc.G9BuJpTuc3PtfM-XiwOzvt? Route: <sip:sbc.megafon.ru;lr> Max-Forwards: 70 From: "Dmitry" <sip:7926246xxxxx@multifon.ru>;tag=Xk1AOqIG.E03wSSXT9poQGOr7PS.BKbm To: "Dmitry" <sip:792624xxxxx@multifon.ru> Call-ID: gmImYL9eZIkNW-IaMt0vf5hxuqHYSqQF CSeq: 59669 REGISTER User-Agent: Telephone 0.15.2 Contact: "Dmitry" <sip:79262466786@192.168.1.83:56468> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.83:56468;received=62.118.16.234;branch=z9hG4bKPjU9hcfhKtc.G9BuJpTuc3PtfM-XiwOzvt?;rport=63210 From: "Dmitry" <sip:79262xxxxx@multifon.ru>;tag=Xk1AOqIG.E03wSSXT9poQGOr7PS.BKbm To: "Dmitry" <sip:792624xxxxx@multifon.ru>;tag=aprqngfrt-mhr3ec1008qb1 Call-ID: gmImYL9eZIkNW-IaMt0vf5hxuqHYSqQF CSeq: 59669 REGISTER Reason: Q.850;cause=3;text="Call Terminated"

Excuse me! I am new to mac. Can you tell me how can I uninstall it? I got same preconfigured from my provider and unable to remove it now.

How can I configure a pre-dial number for all calls? I mean that TLPHN dials automaticaly two digits, for example: 05+number.

How can I configure a pre-dial number for all calls? I mean that TLPHN dials automaticaly two digits, for example: 05+number.

Hey, thanks for your useful work. I'm happy to have found a way to set up telephone with a sipgate account now that gizmo5's gone out of commission. I greatly prefer this app to sipgate's. I came to this page today in hopes of making a feature request - that is some options to make new call windows float above all - whether that be automatically, or if the growl notification is clicked - I can imagine trying out both and see how I leave it set.

I ask for this because I've found I've missed a few calls with inexpert timing, trying to get some headphones plugged in and then trying to search for the app and its window… in the mean time I'll get more practice perfecting my procedure. Thanks!

Всем привет! Сейчас я нахожусь в Китае и вот в чем проблема:

НИКАК не получается пользоваться SIPNET-аккаунтом в этой программе. Звоню, например, с Macbook Pro на моб.телефон. Звонок проходит, отвечаю и.. на "маке" не слышно собеседника, в то время как на моб.всё чётко и прекрасно. Перепробовал вроде все возможные комбинации с галочками, прокси и т.п. Может кто-нить подскажет, в чем может быть проблема?

Заранее, большое спасибо!

I'm using a pbx. Is there a way to let telephone compose some digit before the number stored in the address book? I don't' want to store 0 (that i use for having external calls) beside all my numbers :-)