nathans

Hi, Our telco provider does not provide naming into on the incoming caller ID. As such the daily call logs look like this: 16:14 I unknown-caller-name (5553908711) 70 5553908711 03:24 Is there a way to eliminate the "unknown-caller-name" string from the log and leave only the phone number? We are running a snomone+ 2011-4.2.1.4025 with a sangoma E1 card Thanks Nathan

As somehow who had 3 snomone+ installs, I have to agree to some of these points. We choose the snomone+ box mainly because we were looking for an easy plug & play setup. That is the same reason we went for 100% snom phones on these installs. Nevertheless we had some issues in all of them making the plug & play just a dream and ended up having to spend more time than expected on configurations. Granted we had some very specific technical challenges and snom support as well as sangoma support did an awesome job in getting us up and running. Having said that, I can say the 3 systems (once working) are working flawlessly.

This one is trance as I'm pretty sure it was working and somehow we did something and it is not longer working. Basically we are trying to use the default star code for call forwarding *77 or call pick up and we get a recording for "service is not available" On the snomone trace below extension 9999 is trying to forward a call to extension 45. 7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024: INVITE sip:*7745@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone> Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 X-Serialnumber: 000413412E6B P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom870/8.4.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:187.174.100.120>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 528 v=0 o=root 100582088 100582088 IN IP4 75.149.181.121 s=call c=IN IP4 75.149.181.121 t=0 0 m=audio 53466 RTP/AVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XzhHQKdn2K8ILVlcwvTbc6XTO0iWl7pmUhQ4ijEM a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport=1024 From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 INVITE Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport=1024 From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 INVITE User-Agent: snom-PBX/2011-4.2.1.4025 WWW-Authenticate: Digest realm="192.168.30.2",nonce="059ba8b53b1b0982fd128db9b5ef8e58",domain="sip:*7745@192.168.30.2:5060;user=phone",algorithm=MD5 Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024: ACK sip:*7745@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024: INVITE sip:*7745@187.174.100.120;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9 To: <sip:*7745@187.174.100.120;user=phone> Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:9999@192.168.88.107:3072;line=4r9hwkd9>;reg-id=1 X-Serialnumber: 000413412E6B P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom870/8.4.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:187.174.100.120>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="9999",realm="192.168.30.2",nonce="059ba8b53b1b0982fd128db9b5ef8e58",uri="sip:*7745@192.168.30.2:5060;user=phone",response="7d4d1b35a0321805252b0311df97c9e3",algorithm=MD5 Content-Type: application/sdp Content-Length: 528 v=0 o=root 100582088 100582088 IN IP4 192.168.88.107 s=call c=IN IP4 192.168.88.107 t=0 0 m=audio 53466 RTP/AVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XzhHQKdn2K8ILVlcwvTbc6XTO0iWl7pmUhQ4ijEM a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport=1024;received=75.149.181.121 From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9 To: <sip:*7745@187.174.100.120;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 INVITE Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport=1024;received=75.149.181.121 From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9 To: <sip:*7745@187.174.100.120;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 INVITE Contact: <sip:9999@192.168.30.2:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Type: application/sdp Content-Length: 396 v=0 o=- 405392631 405392631 IN IP4 192.168.30.2 s=- c=IN IP4 192.168.30.2 t=0 0 m=audio 53968 RTP/AVP 0 8 9 18 99 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2012/01/09 17:42:38: SIP Rx udp:75.149.181.121:1024: ACK sip:9999@192.168.30.2:5060 SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-lh9yrb0744rt;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 Content-Length: 0 [7] 2012/01/09 17:42:40: SIP Rx udp:75.149.181.121:1024: BYE sip:9999@192.168.30.2:5060 SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-78xiabcq4825;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 User-Agent: snom870/8.4.33 RTP-RxStat: Total_Rx_Pkts=106,Rx_Pkts=106,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=165,Tx_Pkts=121,Remote_Tx_Pkts=0 Content-Length: 0 [7] 2012/01/09 17:42:40: SIP Tx udp:75.149.181.121:1024: SIP/2.0 200 Ok Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-78xiabcq4825;rport=1024 From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 3 BYE Contact: <sip:9999@192.168.30.2:5060> User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0

Understood. But I would think that before doing a trace or anything I should be seeing the multicast address being populated in the multicast paging group in each phone, no? If i manually enter the address, paging works. My problem is configuring 85+ phones manually...

attachment for paging account

By paging group, you a Paging account? I have one with a multicast address on it as shown on the screenshot attached. They phone do not have anything showing on the multicast .

Hi, I'm probably missing something, but how do you autoprovision a whole group of phones to assign them a multicast addresses to be part of paging group? I know it can be done one-by-one via the phone's web interface and I know how to create a paging extension telling it which extension should it be part of. Thanks. Nathan

Hello, We have a customer who wants to have a "send to fax" option/code when on a call. Their sales people are on the phone with a client and when they close a deal the client will fax them a confirmation. They are used to doing this on the same call. This is, they close the deal and the client asks them for a Fax Tone on the same call. They used to hit a code (*55) and transfer the call to the local fax machine that was hooked up to one of the FXS of their PBX. Any ideas on what would be the best setup/solution to achieve this same functionality with a snomone box? We only have FXO ports, no FXS. Thanks Nathan

Thanks. #1 and #2 seem to be doing the trick. As for recognizing the inbound caller ID, it seems like either the Telco or the Sangoma are not recognizing the number of the analog trunk that is ringing: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5066;rport=5066;branch=z9hG4bKd9257140-1dd1-11b2-837e-fc130d81ef11 From: "unknown-caller-name" <sip:unknown-ani@192.168.31.2:5066>;tag=pxip-callid-1319747569-345281-720721662-1123ds-8f6ac0b-b574aab4 To: "111" <sip:111@localhost:5060>;tag=49dec7da98 Call-ID: d924f0f8-1dd1-11b2-903a-9ee89d225bf7@snomONE CSeq: 9561457 INVITE Content-Length: 0 [7] 2011/10/27 16:32:54: Set packet length to 20 [6] 2011/10/27 16:32:54: Sending RTP for d924f0f8-1dd1-11b2-903a-9ee89d225bf7@snomONE to 192.168.31.2:14612, codec not set yet [5] 2011/10/27 16:32:54: Received incoming call without trunk information and user has not been found [7] 2011/10/27 16:32:54: Set packet length to 20 [7] 2011/10/27 16:32:54: SIP Tx udp:127.0.0.1:5066: My questions are: 1) Is there any settings on the Sangoma FXO config or the snomone+ to tweak to recognize the inbound trunk 2) If not, what is the best way of configuring the 16 FXO lines so that I can know which one is ringing and routed accordingly? Any ideas? Thanks! Nathan

HI, Setup: snomone+ 16FXO sangoma NBE cards. On the "Trunk" definition for the Sangoma 16FXO tab I have set it up as a SIP Gateway, Inbound and outbound, NBE, Enabled and 12 ports. (see attachment) Question: 1) On the "Number of ports" I'm limited to 12 while I have 16. How could I enable the whole 16 FXOs? 2) on the CO Lines: I just added 12 CO lines because of the above, but I'm guessing it should be the 16 as in the number of FXO channels, right? 3) How do I make inbound Caller ID work? I'm able to make outbound call no problem, but inbound I have not been able to get the caller ID to route calls. Any ideas? Thanks! Nathan

We have a snomone+ that is rapidly approaching full disk utilization. Yesterday was at 11% free disk space. Today at 7%. We tried SSH to the box using the superadmin but had 2 problems: 1) what to delete 2) no privileges as superadmin to delete any files! What is the correct cleaning mechanism and how to tell what is the disk hog. Thanks for the help Nathan

Hi, We have a customer in Mexico who just ordered a new digital E1 PRI. The only problem is that Telmex (the local telco) hands out these with a G703 interface consisting of 2 coaxial cables out of a Watson sdhl modem. Anybody knows or has experience on how to interface and connect this to a regular sangoma t1/e1 card? We have a balun that goes from the g703 coaxial 75ohms to a rj45 120ohms. Is that enough? what about the rj45 cable then that goes from the balun to the sangoma? should it be a regular ethernet? e1 cable?? Any help and ideas are most welcomed. Thanks! Nathan

Hi, I'm using a SnomONE+ appliance in a setup where the router is the DHCP server for the LAN. I have disabled the internal snomONE+ DHCP server by hitting "stop server" in the Servers->DHCP menu but every time I restart the appliance the DHCP is enabled. What is the right way in snomONE to disabled it always?

Hi Laurent, I have the same issue on two different snomOne+. The system hangs in some sort of shutdown but never restarts. Maybe snom can look at this. Nathan

Hi, Can some one post detailed instructions on what is the best method to upgrade a snmoONE Plus appliance? We are running 2011-4.2.0.3981 (Linux) and want to upgrade to the latest 4.2.1.4025 version. We tried to wen gui upgrade page but everything is broken or not doing anything in there. It tells us we are running version 1.0.1 (?) Do we need to do a manual upgrade or auto upgrade .sh script? what is the correct link to the latest FW? Is it 32 or 64bit? Thanks for the help. Nathan

Something I noticed today and I do not know if it has anything to do with the issues is that for my domain name I'm not using a FQDN. I'm just using a domain I made up like snomne.owl.com. I can see that the 300's are getting the following settings on their identities: Registrar: snomone.owl.com Outbound proxy: sip:192.168.1.2:5061;transport=tls 192.168.1.2 is indeed the snomONE+ internal (and only IP) but snomone.owl.com does not resolve to it (or anything for that matter) I noticed this as on one M9 we have I could not make it register until I changed the registrar from the name to the numeric IP. I guess my questions is, does it matter? and if it does, how/where I change the setting so the phones get the numeric IP instead of the domain name?

Nothing I can see. Can we get an expert to tell us?

To me they make sense, but here they go in case I'm missing something: snom_3xx_fs.xml: <?xml version="1.0" encoding="utf-8"?> <firmware-settings> <firmware perm="RW">http://provisioning.snom.com/download/fw/snom300-8.4.31-SIP-f.bin</firmware> </firmware-settings> Which is the new FW that is not being downloaded by the phone. I can ping and reach the snom.com site no problem for the phone network. no restrictions at all. snom_3xx_fw.xml: <?xml version="1.0" encoding="utf-8"?> <phone-settings> <firmware_status perm="RW">http://192.168.1.2:80/prov/snom_3xx_fs.xml?model=snom300</firmware_status> </phone-settings> 192.168.1.2 is the IP of the snomONE+ located in the same network & subnet as the snom 300. Any ideas? THANKS!!!

Yep. the generated folder has a folder per extension with 8 files in it: /usr/local/snomONE/generated/snomone.owl.com/4003/snom_3xx_fs.xml /usr/local/snomONE/generated/snomone.owl.com/4003/snom_3xx_fw.xml /usr/local/snomONE/generated/snomone.owl.com/4003/snom_3xx_phone.xml /usr/local/snomONE/generated/snomone.owl.com/4003/snom_300_fkeys.xml /usr/local/snomONE/generated/snomone.owl.com/4003/snom_gui_lang.xml /usr/local/snomONE/generated/snomone.owl.com/4003/snom_web_lang.xml /usr/local/snomONE/generated/snomone.owl.com/4003/snom300-0004133B4C48.htm /usr/local/snomONE/generated/snomone.owl.com/4003/snom300.htm Both the phones and the snmoONE are connected directly to a witch and using the same network 192.168.1.0/24 with no router or firewall between them. I'm also only using one of the NICs on the snomONE (the LAN one) The other one is not connected at all. If I manually register the phone it works, but if I restart it it never upgrades the FW to the one in the provisioning URL.

How do I elevate the permissions?

On the snomONE+ in the "servers' tab there is a 'View PBX passwords" feature which is very handy when you auto provision phones and are trying to debug as you can see in plan text the randomly generated SIP passwords. The only problem is that in mine with about 50 extensions, they are in some strange order and not sorted by extension (or anything esle I can tell). They start to skip lines and/columns and you end up with not matching extensions to passwords and not all them show without any button for next page. Anyone else having this? Any ideas? Thanks Nathan

Hi, We have upgraded to the latest version and that has indeed taken away the issue of the scrolling directory appearing every time we dialed a number. But the issue with the PnP on the snom 300 still remains: We configured all of the domain settings, extensions, password, etc and have the SIP subscribe option ON on the snomONE+. We assign a MAC adress to an extension. And we plug the 300 directly to the same switch as the snomONE plus. It powers up with an IP but id does not upgrade FW (it comes with 7.x) and it does not register at all either. Any ideas? Thanks Nathan

This means I can also authenticate via web to the phone using admin/pin combination?

Thanks to both. The admin and PIN pnp passwords I know as I did set them up for the domain. My problem as that I still thought that when you http to any phone provisioned by the system if you used the admin/domainpassword combination at the prompt you would get to the admin menus. If you used your user/userpassword at the prompt you will get the limited web interface. Is this not true? I thought the PIN was only used with the keypad to access settings via the phone itself. I'm a missing something? Thanks!

Hi, This is probably something very simple but is driving me crazy: I have a bunch of snom 300 & 821s connected and auto provisioned to a snomONE+ running the latest version of FW on all. The phones register OK and everything is working except when I try to web login to the phones: After being prompted from the admin login/password I get like a limited version of the WEB gui. I do not see the identities tab and when I go to advanced settings I can see all of the Network/Behavior/AUdio/Sip/QoS update tabs but the only thing under it is a label and a box for "Administrator Login" and a save button. (attached screen shot) I have not changed any of the default XMLs or done anything other than auto provision the phones using PnP. How do I get the full remote admin web gui? What am I missing?? Thanks! Nathan