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
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
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
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
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
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
