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moh10ly

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About moh10ly

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  1. but this is a PoC, PoC customer won't really wait until the forum is up again . I have made some progress but i'm not sure if it would really satisfy the client. I have copied the tbook xml script example on the mass deployment page into settings.xml file which I downloaded from the phone (in XML file type) and replaced it with the portion where the tbook settings are located. <tbook> <item context="line1" type="none" index="0"> <name>Adrian</name> <number>42965</number> </item> <item context="active" type="colleagues" index="1"> <name>Roland</name> <number>16424</number> </item> ... <item context="line12" type="friends" index="99"> <name>Suzy</name> <number>78371</number> </item> </tbook> --------- I saved these into a file called settings.xml and loaded it onto the IIS local site, then ran this script on my Internet browser http://192.168.1.100/dummy.htm?settings=save&setting_server=http://192.168.1.192:8080/settings.xml&store_settings&=save This have copied the addressbook settings xml URL into the phone's tbook, but it just requires me to manually reboot the phone for settings to take effect. Is there anyway to restart the phone with the script on Internet explorer? Thanks
  2. So if the Ldap server is down or not accessible, the address book is not available? I don't think this is a good idea Could you please just give me a hint on how to start with this XML Directory deployment , I created an IIS webserver and set the phones settings to the following http://192.168.1.100/dummy.htm?settings=save&xcap_tbook_sync_interval=60&xcap_server_name=192.168.1.192&xcap_server_port=8080&store_settings&=save xcap_tbook_sync_interval!: 60 xcap_server_name!: 192.168.1.192 xcap_server_port!: 8080 xcap_directory_auid!: xml xcap_dir_doc_name!: directory.xml xcap_via_tls!: false My phone's IP address is 100, the IIS server is 192. but when I run packet trace I can see the http get request coming in this format which I'm not sure if IIS would handle it. the IIS server responses with file not found. but the xml directory file is there. http://server:port/direcory/users/sip:username@registrar/document This is the original link http://wiki.snom.com/Features/Mass_Deployment/Setting_Files/XML/Directory I would be grateful for your help
  3. What format do I need to import the address book into Snom One. and what settings do I need to do in order for phones to get address Book from Snom One. also in case this works, will the address book be saved into phones or would it only have access to it on Snom One. ? Thanks
  4. Hello everyone, This place might not be relevant to the question I'm asking but I have been in contact with Snom support and they seem busy. I have a critical request on the possibility of importing Cisco Call Manager address book into 200 Snom 700X phones in mass deployment . The number is not certain yet but it could be even more. what I have dig around about this until now is I found there is away to do it through XCAP server, I'm trying to deploy one using Kamailio on Linux but it's very difficult for me as i'm a newbie to Linux. The other method which I got from Snom support is this link but with no further comment on how this works http://wiki.snom.com/Features/Mass_Deployment/Setting_Files/XML/Directory I have setup the phone settings but i'm not sure how the phone will acquire the address book from Cisco call manager. Do I need to setup a server in order for phones to acquire the address book in XML directory format? I appreciate your help.
  5. the calls between two telephone internally do not disconnect but only received calls from the ITSP provider disconnects. do you recommend using TLS ? if not what else would the problem be. the client have two dif vlan networks, one for VoIP and other for Data! I have provided my client the options to use in Vlan. Thanks btw, I'm not sure which firmware he's using but I think on the phone he's using 8.7.3.8.
  6. I have a client who is testing Snom 710 phones, calls are disconnected whenever they try to call. what I can see in the logs are the alert "SIP: closing call -7 due to missing ack" I have advised my client to update their firmware since I saw this bug mentioned in the earlier firmware and has been fixed but my client is not even able to upgrade the firmware on the phone, it says (Software Update Downloading Firmware File not found error: 99) Anyone to help ? why is all this happening ? I'm attaching snapshots of the error on the phone firmware upgrade and the errors for calls. https://www.dropbox.com/s/qrnpjch0hqzf8ck/phone%20error.jpg 8/3/2013 10:01:39 [NOTICE] PHN: SDP: Skipped Codec G723 is not supported for 20 ms packets 8/3/2013 10:01:39 [NOTICE] PHN: SDP: Skipped Codec G723 is not supported for 20 ms packets 8/3/2013 10:01:41 [NOTICE] PHN: Resetting SRTP 8/3/2013 10:02:13 [NOTICE] PHN: SIP: final transport error: 1000053 -> udp:10.30.0.179:5060 8/3/2013 10:02:13 [ALERT ] PHN: SIP: closing call -7 due to missing ack 8/3/2013 10:02:15 [NOTICE] PHN: Fetching URL: snom://mb_exit 8/3/2013 10:06:53 [NOTICE] PHN: SDP: Skipped Codec G723 is not supported for 20 ms packets 8/3/2013 10:06:53 [NOTICE] PHN: SDP: Skipped Codec G723 is not supported for 20 ms packets 8/3/2013 10:06:54 [NOTICE] PHN: Resetting SRTP 8/3/2013 10:07:26 [NOTICE] PHN: SIP: final transport error: 1000057 -> udp:10.30.0.179:5060 8/3/2013 10:07:26 [ALERT ] PHN: SIP: closing call -8 due to missing ack 8/3/2013 10:07:28 [NOTICE] PHN: Fetching URL: snom://mb_exit 8/3/2013 10:09:00 [CRITIC] PHN: start_dst(1365303600) end_dst(1382846400) offset_dst(3600) offset_utc(7200) 8/3/2013 10:09:00 [CRITIC] PHN: start DST: 04/07/2013 03:00:00 (1365303600) 8/3/2013 10:09:00 [CRITIC] PHN: end DST: 10/27/2013 04:00:00 (1382846400) 8/3/2013 10:09:00 [CRITIC] PHN: SIP: Registered at registrar as 2440287@ (Expires: 1800 secs) 8/3/2013 10:09:00 [NOTICE] PHN: Settings stored! 8/3/2013 10:24:00 [CRITIC] PHN: SIP: Registered at registrar as 2440287@ (Expires: 1800 secs) 8/3/2013 10:24:00 [NOTICE] PHN: Settings stored!
  7. Hello friends, Is there anyway to find out when is forum.snom.com is gonna come back alive again?
  8. I actually have this setup already where Snom is a PSTN gateway for Lync. everything works fine in terms of simultaneous calls if I put the Lync user's DID in the redirection setup but I was just wondering if both would ring at same time.. the Lync user's DID and PBX extension.
  9. I would like to know if is it supported to call an extension and Lync DID of user simultaneously ? in more details, I have extension 40 which is a Lync user with DID 320 220 9663. I would like extension 41 to call extension 40 at both Desk and Lync at the same time. what I have tried so far is setting up Lync user's extension to 41 but that would only ring the desk phone. I tried redirection (Cell phone) and this works but I would rather use cell phone for cell phone not for desk purpose. any help is appreciated. Thanks.
  10. where do I place the sip:recording.server:5060? and is it possible to just use this scenario to invite the third party ? placing this in the phone settings ? <recording_mechanism perm="&">sip:ext@server_IP</recording_mechanism>
  11. I would like to know how to achieve the following: I would like to record calls between 2 SIP phones which are registered to SnomOnePBX but using a different Recording Server or product. Right now I have this scenario setup and ready to go except one more thing. In order to start recording, It requires a direct interaction from one of the two parties to dial third party extension. That third party which is the recording server needs to be invited manually in order for recording to start. What I would like to achieve is how to automatically invite this extension on any calls that take place between any sip phone that's registered to SnomOne PBX. or any SnomOne Phones without direct interaction from phones. Any input is much appreciated. Thanks
  12. Also it would be cool if we can control audio files and assign them to which part of the Auto attendant or the IVR from the web application instead of placing a file inside the audio folder and renaming it manually.
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