Re: [asterisk-users] Swissvoice IP10s setup
Are you using trixbos ? If yes have a look at the forums on www.trixbox.org - Original Message - From: Paul A Brown To: asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 9:18 PM Subject: [asterisk-users] Swissvoice IP10s setup Hi Does anyone have a howto on how to set one of these up on Asterisk or Trix box please? I can make it SIP or MGCP so whatever you have ;-) I have found one page but it isn't really a howto setup Thanks in advance Paul -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Swissvoice IP10s setup
Hi Does anyone have a howto on how to set one of these up on Asterisk or Trix box please? I can make it SIP or MGCP so whatever you have ;-) I have found one page but it isn't really a howto setup Thanks in advance Paul___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice IP10S centralized phonebook
Anybody got any documentation/experience on the subject? I'm trying to get it working, but the documentation I have lacks any information on what should be installed on the server side. -- Igor Briški - [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Swissvoice IP10S centralized phonebook
This is the information I got from Swissvoice support, I didn't tried yet, but if it can helps. How to use an external phone book IP10S phone supports access to Cisco Phone Book but not all functionalities. The IP10 uses his own interface to access to the Phone Book. If you want to connect to your remote phone book, you have to do the following actions: First, copy the URL under Search by name in a Web browser, for example: http://192.168.1.5/cisco/directory/searchDirectory.php You are going to have a XML file display in the Web Browser, like this one: CiscoIPPhoneInput TitleDirectory Search/Title PromptEnter search criteria/Prompt URLhttp://10.3.100.190:8080/ciscodirectory?action=listpage=0/URL InputItem DisplayNameFirst Name/DisplayName QueryStringParamfirstname/QueryStringParam InputFlagsA/InputFlags /InputItem InputItem DisplayNameLast Name/DisplayName QueryStringParamlastname/QueryStringParam InputFlagsA/InputFlags /InputItem InputItem DisplayNameNumber/DisplayName QueryStringParamnumber/QueryStringParam InputFlagsT/InputFlags /InputItem /CiscoIPPhoneInput Copy the information from the URL line (URLhttp://10.3.100.190:8080/ciscodirectory?action=listpage=0/URL). The easiest way to set the path in your phone is by the Web interface (but it could also be done by Telnet). Connect to your phone web server. Login and password are normally: admin Select Configure common phonebook. In Select phone book to use chose the value: Remote Then click on submit. File the box below with the URL you get previously: The IP address and the port number of the Phonebook server can be manually entered or synchronised with the Call Agent In our case, IP address: 10.3.100.190, Port number: 8080,Path: /ciscodirectory?action=listpage=0 Then click on submit. If you return to your phone and select the common phone book, it is normally connected to the remote one now. You can search by a name or if you put nothing and press on OK, it will return you the entire content of the remote phone book. More information about Cisco Phonebook management To manage remote phone book, you need a server. It could be one you developed by yourself or one include in your proxy server (not all of them include this feature). It must follow the Cisco implementation; you can have more information here: http://www.cisco.com/univercd/cc/td/doc/product/voice/vpdd/cdd/3_1/index.htm Check for Cisco IP Phone Services Application Development Notes with Cisco CallManager 3.1. Igor Briski wrote: Anybody got any documentation/experience on the subject? I'm trying to get it working, but the documentation I have lacks any information on what should be installed on the server side. -- Igor Briški - [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Swissvoice IP10S opinions?
Any luck with these phones and their SIP firmware? I just changed one of my IP10s to use SIP and I can't get it to work with Asterisk (it doesn't register at all). Regards, Alejandro. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Saturday, October 30, 2004 6:09 AM To: JB Hewit; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Swissvoice IP10S opinions? Hi, On Sat, 2004-10-30 at 02:50, JB Hewit wrote: Hi, I'm looking at trying out an IP10S with Asterisk. I'll be recieving a single unit next week to try out and see what she can do. It seems to be comparable to a Snom190, but I don't seem to find much detail online about it with Asterisk. Is anyone out there using these phones? Any quirks, reviews, goodness, badness about them? Use the MGCP firmware, its a lot more mature than their new SIP version. I'm currently rolling out several hundreds of them and they make excellent 'standard issue' desk phones. Asterisk could get a bit more work in the Business package support, but thats all fancy. There was an issue with RTP, because the IP10 can only deal with 1 RTP stream at a time, which is why I asked Digium to implement the singlepath option :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Swissvoice IP10S opinions?
Hi, On Sat, 2004-10-30 at 02:50, JB Hewit wrote: Hi, I'm looking at trying out an IP10S with Asterisk. I'll be recieving a single unit next week to try out and see what she can do. It seems to be comparable to a Snom190, but I don't seem to find much detail online about it with Asterisk. Is anyone out there using these phones? Any quirks, reviews, goodness, badness about them? Use the MGCP firmware, its a lot more mature than their new SIP version. I'm currently rolling out several hundreds of them and they make excellent 'standard issue' desk phones. Asterisk could get a bit more work in the Business package support, but thats all fancy. There was an issue with RTP, because the IP10 can only deal with 1 RTP stream at a time, which is why I asked Digium to implement the singlepath option :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice IP10S opinions?
Hi, I'm looking at trying out an IP10S with Asterisk. I'll be recieving a single unit next week to try out and see what she can do. It seems to be comparable to a Snom190, but I don't seem to find much detail online about it with Asterisk. Is anyone out there using these phones? Any quirks, reviews, goodness, badness about them? -- Regards, JB Hewitt Business: http://www.stcpl.com.au Blog: http://blade.lansmash.com Best LAN ever: http://www.lansmash.com How to ask a ?: http://www.catb.org/~esr/faqs/smart-questions.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Swissvoice IP10S and RTP Port Operation
Hi Matthew, -Original Message- What is even better is that this coincides with the Terminating on result 502 from [EMAIL PROTECTED] error I get in * So, I am guessing these are related. Any help here would be greatly appreciated. I am so close to getting this phone working in MGCP mode. Did you post your phone config to the list ? (mirror pages or something) I don't really understand why it's so hard for you where my phones did basic functions (calling/being called) almost right out of the box. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice IP10S and RTP Port Operation
I had the telnet window to the phone open by chance and noticed this line twice when I tried to call the IP10: WARNING: may need to undo rtp port operation here The warning line appeared immediately when I picked up the handset. I have no idea what this means. I also tried calling the phone from a POTS phone and I got the same warning. What is even better is that this coincides with the Terminating on result 502 from [EMAIL PROTECTED] error I get in * So, I am guessing these are related. Any help here would be greatly appreciated. I am so close to getting this phone working in MGCP mode. Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Swissvoice ip10s
Hi, -Original Message- I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. SIP firmware is currently being tested, there are a few issues that need to be resolved. For your MGCP phone: configip10.cfg can be altered to add services: set features new 1 Transfer NOINFO NOCONF TRUE NOSEQ set features new 2 Operator NOINFO NOCONF FALSE extension of your secretary And then: set service_state IDLE NEW 2 set service_state ONE_ACTIVE_LINE NEW 1 This will add two services: In idle state: An operator button that speeddials your secretary (who can connect you through ;-) In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Swissvoice ip10s
Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] On Jun 24, 2004, at 12:33 AM, Florian Overkamp wrote: Hi, -Original Message- I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. SIP firmware is currently being tested, there are a few issues that need to be resolved. For your MGCP phone: configip10.cfg can be altered to add services: set features new 1 Transfer NOINFO NOCONF TRUE NOSEQ set features new 2 Operator NOINFO NOCONF FALSE extension of your secretary> And then: set service_state IDLE NEW 2 set service_state ONE_ACTIVE_LINE NEW 1 This will add two services: In idle state: An operator button that speeddials your secretary (who can connect you through ;-) In conversation: A Transfer button that hookflashes and gives a dialtone (There were some issues with that, and I have just now been asked by mark to verify if they have been resolved). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Swissvoice ip10s
Hi, -Original Message- Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Ouch! Can you check if it is still fetching any config files from your FTP-server at boot ? Might be your configs are corrupted somehow. If it is not even doing that, you might just have to ship it back to SwissVoice and have them fix it :-P Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Swissvoice ip10s
Yep I stayed and was able to get through to their ip-phone support in france. And with me only knowing english and the guy on the other end speaking broken english we kinda hashed out that it was a bad stick of flash ram in the phone. Communitech the USA provider for the phone is overnighting me a new one. AND emailing me the sip firmware for the mgcp phones. Thanks, Matt Hohman - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 9:24 AM Subject: RE: [Asterisk-Users] Swissvoice ip10s Hi, -Original Message- Thanks! well after doing some other .cfg file changes I hardlocked the phone durring startup! Any ideas? (pushing 1,4,7 on powerup isn't helping) Ouch! Can you check if it is still fetching any config files from your FTP-server at boot ? Might be your configs are corrupted somehow. If it is not even doing that, you might just have to ship it back to SwissVoice and have them fix it :-P Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Swissvoice ip10s
Ah nice, Let me know what SIP version you get, if it's any more recent than the one I have, I'd love to get a copy ;-) (There are some issues in my version that make the phone rather useless) Florian -Original Message- Yep I stayed and was able to get through to their ip-phone support in france. And with me only knowing english and the guy on the other end speaking broken english we kinda hashed out that it was a bad stick of flash ram in the phone. Communitech the USA provider for the phone is overnighting me a new one. AND emailing me the sip firmware for the mgcp phones. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice ip10s
I've noticed quite a few posts on the list about the swiss voice ip10s phone. We recently purchased a few of these phones and have had no luck getting the services button to work any ideas? are the example .cfg files for this phone? any idea when sip firmware is coming? Any help/info would be great. Thanks, Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]
[Asterisk-Users] swissvoice ip10s firmware?
Hi, Does anybody know the place to download the firmware for swissvoice ip10s I have several phones with application IP10 H3 v1.0.0 (Build 1) I'm looking for newer H.323 and also MGCP firmwares Are the SIP firmware available, according to web its targeted to Q1 2004, but we have week left in Q2 I sent several email to swissvoice support,, no answers Regards Juri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] swissvoice ip10s
hallo, i would like to ask if somebody have sip firmware for this ip phone from swissvoice. they announced sip firmware in April 2004 but so far i'm unable to contact product manager and get the sip firmware. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice IP10S not able to dial calls
Hi! I have set up a new SwissVoice phone and it can receive calls but I cannot make calls out from it. The setup is simple for now, 2 phones: SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999. Which verasion of Asterisk are you using? Please do check the known MGCP bugs, especially 881: http://bugs.digium.com/bug_view_page.php?bug_id=881 Also don't forget to provide info about your ip10 firmware version. Finally: Did you put the ip10 into the right context so that it actually has the required access rights to dial 7999? Cheers, Philipp I can call from the Cisco phone and it rings on the SwissVoice phone but when I dial from the SwissVoice phone I get a busy tone upon dialing the second digit. The log reads as follows: -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '9' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- MGCP handle_request(aaln/[EMAIL PROTECTED]) ast_channel already destroyed -- MGCP handle_request(aaln/[EMAIL PROTECTED]) set vmwi(-) Here are my configuration files: MGCP.conf === [10.1.24.112] context=local host=10.1.24.112 callerid = Brad Chilton 7726 callgroup=0,2-5 canreinvite=no pickupgroup=0,1 nat=no threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer ;callwaiting=yes ; this might be a cause of trouble for ip10s ;cancallforward=yes line = aaln/1 EXTENSIONS.conf === exten = 7726,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SwissVoice IP10S not able to dial calls
I have set up a new SwissVoice phone and it can receive calls but I cannot make calls out from it. The setup is simple for now, 2 phones: SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999. I can call from the Cisco phone and it rings on the SwissVoice phone but when I dial from the SwissVoice phone I get a busy tone upon dialing the second digit. The log reads as follows: -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '9' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- MGCP handle_request(aaln/[EMAIL PROTECTED]) ast_channel already destroyed -- MGCP handle_request(aaln/[EMAIL PROTECTED]) set vmwi(-) Here are my configuration files: MGCP.conf === [10.1.24.112] context=local host=10.1.24.112 callerid = Brad Chilton 7726 callgroup=0,2-5 canreinvite=no pickupgroup=0,1 nat=no threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer ;callwaiting=yes ; this might be a cause of trouble for ip10s ;cancallforward=yes line = aaln/1 EXTENSIONS.conf === exten = 7726,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] swissvoice ip10s
Hi! does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. Never tried, no clue. But I can tell you that newer ip10 firmware and latest head CVS (yesterday) don't play together at all - see bug 881. Appli version IP10 M v1.0.0 (Build3) Boot version IP10 Boot v0.3.6 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) Protocol MGCP 1.0 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SwissVoice IP10S can't take back call
Hello I'm testing swissvoice IP10S with asterisk 0.7.1 When i place a call and hold it, i can't take back this call. Asterisk seems to go into conference mode and i ve got a dialtone instead of getting back the call. Here is the asterisk output: Call placed -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '1' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '2' -- Executing Macro(MGCP/aaln/[EMAIL PROTECTED], apl1|mgcp/aaln/[EMAIL PROTECTED]) in new stack -- Executing Dial(MGCP/aaln/[EMAIL PROTECTED], mgcp/aaln/[EMAIL PROTECTED]|10|Ttr) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered MGCP/aaln/[EMAIL PROTECTED] -- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] Call hold -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- Started music on hold, class 'default', on MGCP/aaln/[EMAIL PROTECTED] -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down Call take back -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] -- MGCP Conferencing 0 and 1 on aaln/[EMAIL PROTECTED] -- Stopped music on hold on MGCP/aaln/[EMAIL PROTECTED] I have a look at chan_mgcp.c and it looks like asterisk thinks that there are 2 active calls and decide to conference instead of bringing up the only active call. Does anyone use this feature with mgcp? Does it work? Is this a problem with asterisk (it seems so)? or do i have to configure the phone to send something else than hf (hook flashing) to take back the call? what sequence sould it send? Thanks -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice ip10s MGCP questions and experiences
Hi there, here some questions and experiences after playing for one day with 3 Swissvoice ip10s and the latest * CVS: QUESTIONS: - what is the user option enter voice mail number good for? It doesn't appear to be of any practical use - does anyone have some Swissvoice info that I cannot find on their web site like the guide to MGCP XML (.svd), guide to configuration file format (.cfg), guide to phone configuration through TELNET? - do I need two lines (aaln/1 and aaln/2) or just one in my mgcp.conf (since the ip10 appears to support two lines)? - are there any specs available for the format of the operator/ user logo? Must this be b/w, or can it be greyscale (2 bit/ 4 bit/ 8 bit?). As I understand both .gif and .jpg are fine. Pixel size? - how can I obtain newer firmware that hopefully solves my crash problem? WORKING: - simple calls, phone book, hands-free - MWI: F4 key lights up when messages are available - pick-up: Assign the pick-up function together with *6 to F2. - assign FLASHING to F1 - DoNotDisturb (on F3) works as expected - blind transfer using # NOT or PARTIALLY WORKING: - I can only set the phone name via display -- free idle text? - with F4 configured as voicemail function: pressing that key dials # and I must modify extension.conf to point # to VoiceMailMain2? Is there no way to change the number dialed by the voicemail button? - MWI: the button lights up/extinguishes only when some kind of phone activity occurs; without any activity it might actually indicate wrong info (e.g. when user already check vm via web) - during startup the phone nicely display the waiting for call agent message; however when the call agent (call manager?) * disappears afterwards the phones do not display that waiting message... :-( - is there any way to make the service key do anything useful? Currently it has no function at all. - crash: let the two ip10 ring the third ip10 at the same time (without lifting the handset) - in the moment the 2nd phone is calling the third phone crashes and must be rebooted. I can reproduce this 100% with any of the three phones. :-( - the three call forward (CF) functions that can be assigned to the function keys F1-F4 don't appear to be useful. Instead of arrange an internal phone call forward this will immediately (!) dial the forward number entered at the very moment this function is activiated. - I can successfully use FLASHING (F1) for consultative transfer, however I get into bad trouble if the consulted extension does not like to take the call and hangs up -- there is apparently no way to get back to my original caller, although I see both lines in the display and can move my cursor up and down on them. This might also be an * problem. - web interface: save current phone settings as profile doesn't appear to be working, I see only 1 line displayed - despite the seemingly permitted nat=yes entry I so far found no way to get this phone working from behind NAT (I know that MGCP as such probably will never work behind NAT, but still... by the way there is a new RFC on the road to attack this issue...) NOTES: - always restart (not reload) Asterisk after modifying mgcp.conf! - mgcp.conf doesn't like the following keywords: disallow= allow= callwaitingcallerid= fromuser= - the display is fine and equipped with nice logos, icons etc. The only thing that it is missing is the backlight, but I guess I can live with that... - the menu is well organized and easily understood and navigated - I don't think it is acceptable to only include a single A5 sheet of paper and not have ANY other printed user documentation shipped with the phone. A web-based PDF is ok, but there should be more... Greetings, Philipp == hard/software info == Phone name Ext.987 Appli version IP10 M v0.3.0 (Build 6) Boot version IP10 Boot v0.3.4 DSP version Rel 9.1.0.4, Build p8 GG version R9.0.0 IPP (Build 5) IP address 192. 168. 4. 5 Protocol MGCP 1.0 == mgcp.conf == [192.168.13.5] context=default threewaycalling=yes transfer=yes callwaiting=yes host=192.168.13.5 nat=no canreinvite=no callgroup=0 pickupgroup=0,1 cancallforward=yes transfer=yes ;dtmf=inband callerid = John 987 mailbox=1234 line = aaln/1 line = aaln/2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
