reco

hi there, a client is asking for a meeting point. what are good options for pbxnsip/snomone? polycom or snom? thanx

yes on the phone. would be nice to automatically provision a 2nd profile as a backup.

please also a mac osx darwin pbxnsip binary please

nice thanx

hi there, wants the best way to trouble shoot dropped calls? i have people complaining about dropped calls. i see the following in my logs [5] 20111102110243: BYE Response: Terminate 57013b3cc783-98y2n65yo8ry and this Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-de8a3ad89b13788879a54f5777f70b62;rport=5061 From: <sip:8004928468@domain.com;user=phone>;tag=0425310071 To: "Coleen Mac Queen" <sip:21@domain.com>;tag=pg3pm9np15 Call-ID: 57013b3cc783-98y2n65yo8ry CSeq: 2314 BYE Contact: <sip:21@10.0.24.111:4102;transport=tls;line=ljhsjt2p>;reg-id=1 User-Agent: snom870/8.4.32 RTP-RxStat: Total_Rx_Pkts=53264,Rx_Pkts=53264,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=53291,Tx_Pkts=53293,Remote_Tx_Pkts=53232 Content-Length: 0 how can i denitrify where the issue is? thanx

hi there, we had our pbxnsip on two different public ip addresses. some time one provider does some maintenance work and the line is down. is there a way to provision two sip profiles and use the 2nd as a backup in case the server of the first profile is not available? thanx

which vm environment could you recommend? VMware, virtuzzio, virtual box, ....? how is the license managed in a VM environement? still mac address? thanx

thanx for the issue.turned out this was not the issue.

hi there, my trunk provider voxbeam is asking me to remove to user=phone from the to line. change To: <sip:1212xxxxxxx@sbc.voxbeam.com;user=phone>;tag=VBSBC.1948.2045 into To: <sip:1212xxxxxxx@sbc.voxbeam.com>;tag=VBSBC.1948.2045 how can i do that? reco

is there a way to update a setting on all domain? also to update all extensions? x

this could be the issue. i deleted all the co lines from all trunks and domains. that solved the issue for now. will try to bring them back. if this is the case what should i do with button 1? just no config? x

i am actually tempted to track the concept of co-lines totally and use park orbits instead. they are trunk independent. what do you think? would be cool if there is a way to get rid of the feature codes in the dialed list and the announcement that a call was parked in the orbit # x

i set the pattern on the loopback to: ^(\+?[0-9]{10,20})

i do have 4.2.1.4025 (Darwin) with snom 870. works fine even i have different codes for Call Park and Call Park Receive. one annoying thing though is the *85xx in the dialed numbers list. is there a way to exclude feature codes form that list?

i badly need this any update?

i am monitoring my dial plans. seems rules which should match are skipped cause of co lines? can somebody explain me the reason for this? i would expect the pbx to send the call to trunk: voxbeam_js [8] 20111010171436: To is <sip:12129960700@johnsheeley.com;user=phone>, user 0, domain 5 [8] 20111010171436: From user 20 [8] 20111010171436: Call state for call object 455: idle [7] 20111010171436: set_codecs: for 70b7263c8244-u3zz195hutuu codecs "", codec_preference count 7 [9] 20111010171436: Dialplan: Evaluating !^311!sip:12126399675@\r;user=phone!i against 2129960700@johnsheeley.com [9] 20111010171436: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 2129960700@johnsheeley.com [7] 20111010171436: Skipping pattern match because CO-line is not available for trunk voxbeam_js my dial plan: 51;icall-domestic-sheeley;;^311;12126399675 101;voxbeam_js;;*; 400;AmericanVOIP domestic JS;;^([0-9]{10})@.*;"sip:1\1@\r;user=phone" 401;AmericanVOIP domestic JS;;^1([0-9]{10})@.*;"sip:1\1@\r;user=phone" 403;AmericanVOIP international JS;;^011([0-9]*)@.*;"sip:011\1@\r;user=phone" 500;icall-domestic-sheeley;;^([0-9]{10})@.*;"sip:1\1@\r;user=phone" 501;icall-domestic-sheeley;;^1([0-9]{10})@.*;"sip:1\1@\r;user=phone" 502;icall-international-sheeley;;^011([0-9]*)@.*;"sip:011\1@\r;user=phone" trunk `voxbeam_js` has no co lines any idea whats going on? thanx

i think i found the issue. with version 3 i used to add a `Try Loopback` with pattern `*` Replacement `` (empty) in the beginning of a dial plan to enable inter domain calling followed by a trunk. dial plan csv: 50;*;;*; 200;voxbeam_nex9;;*; with the loopback i have the issue. once i remove it seems to work fine: working dial plan: 200;voxbeam_nex9;;*; reco

is it possible that this was caused by a loopback: *

hi there, nope i just replaced my domain with domain.com i have country code set to: 1 area code to : 212 phone number: 212 333 5555 i have multiple domains so not localhost configured yes absolutely. also i do have an extension 12 in other domains. account has a domain default dial plan yes looks like. any suggestions?

hi there, on some calls i am getting 404 not found and i cannot figure out why. any idea? thanx SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.0.24.138:4993;branch=z9hG4bK-69lcg8nopbbd;rport=4993 From: "First Last" <sip:12@domain.com>;tag=koerzr4mij To: <sip:2223335555@domain.com;user=phone>;tag=0a70de7b9f Call-ID: 1d74263c345d-dg32cs7hcuwn CSeq: 1 INVITE Content-Length: 0 [8] 20111010131442: Incoming call: Request URI sip:2223335555@domain.com;user=phone, To is <sip:2223335555@domain.com;user=phone> [8] 20111010131442: Set the To domain based on From user 12@domain.com [9] 20111010131442: SIP Tx tls:10.0.24.138:4993: SIP/2.0 404 Not Found Via: SIP/2.0/TLS 10.0.24.138:4993;branch=z9hG4bK-69lcg8nopbbd;rport=4993 From: "First Last" <sip:12@domain.com>;tag=koerzr4mij To: <sip:2223335555@domain.com;user=phone>;tag=0a70de7b9f Call-ID: 1d74263c345d-dg32cs7hcuwn CSeq: 1 INVITE Contact: <sip:12@10.0.24.2:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/4.2.1.4025 Content-Length: 0

hi there, i am looking into adding fully redundancy to my pbx. dual wan interfaces. is somebody doing this successfully? i am having problems that the pbx always tries to use the default network interface. thanx

was this fixed? if yes in which release?

hi there my hardware just died. i need to replace it asap. where can i get a license to run pbxnsip with a new mac address? the trial is limited to 10 extensions ;( please emai me at reco@nex9.com thanx

is there a way to include some information which shows the user form which trunk the call came from? background info: my client has a private line which rings only one extension. he wants so see on the incoming call that it is that private line. i am using snom 870 and pbxnsip 3.4 thanx

i hear you, i did setup my virtual keys to monitor all extensions. when i have a call on hold its really hard to get to the virtual key menu. is there a better way to do it? my client has a receptionist which is transferring all the calls she/he would need to know if that person is on a call or available. reco