Hubertus R.
-
Posts
9 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Hubertus R.
-
-
Hello all,
in my setup I have to offices served by one SnomOne PBX but two gateways. So in Location A I have my server and one PSTN gateway. In location B, connected via Tunnel, I have a few user and a small PSTN gateway. The calls from this small gateway are usually routed to the extensions also located in location B. Due to the behavior of SnomOne there should usually be no reinvite to the trunk. So the call for instance comes in at location B, goes to the PBX through the tunnel and is then returned through the tunnel to the extension. This causes 2 sip traffic streams through the tunnel. I found now http://forum.snomone.com/index.php?/topic/6206-what-do-these-parameters-specifically-control/ and the note regarding the possibility to enable reinvites for trunks. So for my understanding this should avoid the unnecessary load on the tunnel and connect the gateway and extension in location B directly for the audio stream.
Am I right and is there a way to enable the trunk reinvite option only for specific trunks?
Best
Hubertus
-
Thanks a lot, this solved my problem!!
-
Thanks for the quick response, her the new log:
[5] 2012/10/03 16:29:07: SIP Rx udp:192.168.151.250:5060: INVITE sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250> Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Max-Forwards: 70 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER Supported: 100rel, replaces, timer User-Agent: fec sip stack Allow-Events: refer, message-summary, dialog Content-Type: application/sdp Content-Length: 182 v=0 o=- 7872 1 IN IP4 192.168.151.250 s=SIP call c=IN IP4 192.168.151.250 t=0 0 m=audio 10024 RTP/AVP 8 18 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv [8] 2012/10/03 16:29:07: Allocating for call port 15, SIP call id EEB910E5393926108DC8190009190038 [5] 2012/10/03 16:29:07: SIP Tx udp:192.168.151.250:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2 Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 INVITE Content-Length: 0 [7] 2012/10/03 16:29:07: Set packet length to 20 [6] 2012/10/03 16:29:07: Call-leg 15: Sending RTP for EEB910E5393926108DC8190009190038 to 192.168.151.250:10024, codec not set yet [8] 2012/10/03 16:29:07: Incoming call: Request URI sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c, To is "25" <sip:25@192.168.151.250> [8] 2012/10/03 16:29:07: Trunk R1200-10@snom.mydomain has country code 49, area code 2581 [8] 2012/10/03 16:29:07: call port 15: state code from 0 to 404 [7] 2012/10/03 16:29:07: Set packet length to 20 [5] 2012/10/03 16:29:07: SIP Tx udp:192.168.151.250:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2 Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.150.16:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Content-Length: 0 [5] 2012/10/03 16:29:07: SIP Rx udp:192.168.151.250:5060: ACK sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2 Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 ACK Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Content-Length: 0 [8] 2012/10/03 16:29:07: Clearing call port 15, SIP call id EEB910E5393926108DC8190009190038 [8] 2012/10/03 16:29:07: Hangup: Call 15 not found
I replaced my phone number by 0123123456. Everything else is untouched. I already tried to figure out using the manual, but so far I'm not lucky. Our other gateways didn't make any of those problems (berofix).
Thanks in advance
Hubertus
-
Hello,
I'm having troubles in getting a Bintec R1200 to run as a gateway for incoming calls. I successfully set up the connection using it as a SIP Registration. Also the Call Identification from P-Asserted-Identify works fine. Only the routing to the correct extension doesn't work. So far I can only set a fixed routing using the "Send call to extension" field.
Maybe someone can give me an advise, what I have to change to get this running. Here are the SIP messages, that might help. The called ISDN-MSN was the 25, which should be routed to the internal extension 25.
[5] 2012/10/03 14:33:12: SIP Rx udp:192.168.151.250:5060: INVITE sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250> Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Max-Forwards: 70 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER Supported: 100rel, replaces, timer User-Agent: fec sip stack Allow-Events: refer, message-summary, dialog Content-Type: application/sdp Content-Length: 183 v=0 o=- 23506 1 IN IP4 192.168.151.250 s=SIP call c=IN IP4 192.168.151.250 t=0 0 m=audio 10010 RTP/AVP 8 18 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv [5] 2012/10/03 14:33:12: SIP Tx udp:192.168.151.250:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250>;tag=7a36527553 Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 INVITE Content-Length: 0
As can be seen, the system cannot find the target extension.
[5] 2012/10/03 14:33:12: SIP Tx udp:192.168.151.250:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250>;tag=7a36527553 Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.150.16:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [5] 2012/10/03 14:33:12: SIP Rx udp:192.168.151.250:5060: ACK sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250>;tag=7a36527553 Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 ACK Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Content-Length: 0
In case I set the "Send call to extension" field manually to 25, the successful invite to sip extension 25 looks like this:
[5] 2012/10/03 14:30:42: SIP Tx tcp:192.168.222.26:58226: INVITE sip:25@192.168.101.135:58226;transport=tcp;line=ylef8f1x SIP/2.0 Via: SIP/2.0/TCP 192.168.150.16:5060;branch=z9hG4bK-3ffa59e0b2a97dfcc3306e4b81299598;rport From: "0123123456" <sip:0123123456@snom.mydomain;user=phone>;tag=49668 To: "Sip User 25" <sip:25@snom.mydomain> Call-ID: abdee4b8@pbx CSeq: 3565 INVITE Max-Forwards: 70 Contact: <sip:25@192.168.150.16:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 256 v=0 o=- 50613 50613 IN IP4 192.168.150.16 s=- c=IN IP4 192.168.150.16 t=0 0 m=audio 51504 RTP/AVP 8 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/10/03 14:30:42: SIP Rx tcp:192.168.222.26:58226: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.150.16:5060;branch=z9hG4bK-3ffa59e0b2a97dfcc3306e4b81299598;rport=5060 From: "0123123456" <sip:0123123456@snom.mydomain;user=phone>;tag=49668 To: "Sip User 25" <sip:25@snom.mydomain>;tag=zeczbc5izk Call-ID: abdee4b8@pbx CSeq: 3565 INVITE Contact: <sip:25@192.168.101.135:58226;transport=tcp;line=ylef8f1x>;reg-id=1 Content-Length: 0
So as long as I just want to have one MSN being sent to one extension, this is ok. But I have of course multiple ones I would like to use, without configuring a trunk for each.
Best
HR
-
Thanks for the reply, again
I changed this setting to UDP as it was recommended by beronet for their hardware:
http://wiki.beronet.com/index.php/Berofix_with_Snom_One
As far as I understand, TLS it not fully supported by their hardware. I now asked the support for their explanation and will post the answer here. But I changed the transport layer already to TCP for the whole system yesterday evening and after a reboot, it worked as intended.
So far, thank you very much for you help and I would like to say, that I'm very happy with Snom One. As we only have snom phones, I really think it was worth to change from Asterisk to benefit for the way better interactions out of the box. As soon as we cross the limit of the users, I will not hesitate to pay for the yellow Edition.
Regards
Hubertus
-
Today I figured out a way to solve the problem temporarily:
I forced the snom phones to use TCP by adding ";transport=tcp" to the outbound proxy address. Unfortunately this only works until the next time the phones settings are updated by the provisioning.
Is there a way to force TCP for certain extensions? Otherwise also TCP for all would be acceptable for me. The Transport layer for the SnomOne under PnP->snom in the web interface is currently set to UDP.
Thanks for any suggestions
Hubertus
-
The VPN connection was not originally installed just for telephony, but actually to connect main office and branch and use the same ressources (AD, XenApp, Shares, ...). As the infrastructure is already available, I wanted to use it also for telephony. I don't see a reason to tunnel most of my traffic through the VPN but let the phones connect "around" it - also in terms of safety.
Regarding TLS, do you mean this can also improve my problem still using the VPN?
PnP did not work in the branch office behind the tunnel. So I entered the settings URI manually and set the HTTP Client settings for each of the phones.
If I got you wrong somewhere, let me know...
-
Hello all,
I'm running successfully a SnomONE v4.5.0.1075 on a Windows 2008R2. In the local network I have some Snom 360,370 and 870 connected and they all can successfully call outbound and internal as well as receive calls. Now I also got some clients in a second branch that is connected through OpenVPN to the main network. The manual provisioning of those Snom 370s was successful and I can call those phones and establish the call successfully. Only when trying to call from those devices somewhere else, it fails.
Both when trying to call one of the extensions in the main network and when trying to call outbound (through berofix Gateway) from this branch, the calls won't establish. But when dropping the call then, the call is established. But as then initiating extension already aborted, the end of the line is dead. I can exclude routing problems for sure.
Before SnomOne we used an Asterisk and due to the VPN and thereby reduced MTU., I had to enable the packet fragmentation. http://wiki.snom.com/FAQ/How_to_solve_problems_with_outgoing_calls_in_firmware_version_8%3F
Afterwards this setup worked. But now with SnomOne both enabling this feature as well as short SIP headers, I can't make any outgoing call successfully.
Anybody has an idea how to solve this issue? I can of course provide detailed log information, but please give me a hint what you need. I don't want to spam this forum with logs.
Calling from a PC with XLite4 in this network successfully establishes a call. Also using one of the extensions for the Snom phones....
Best
Hubertus
Trunk reinvite for specific trunks
in Trunk Setup
Posted
Thanks for the instant response. As I'm coming from the Asterisk world, I was in fact hoping for this canreinvite-option. But on the other hand your explanation does have a valid point. So far we are not using recording nor the possility for phones to call in. Our business is just not big enough. So I will for now rely on the option to use the global flag and worst case accept the double load on my tunnel.
Thanks a lot